ETSI Mapeo SIP ISUP
ETSI Mapeo SIP ISUP
ETSI Mapeo SIP ISUP
0 (2011-09)
Technical Specification
The present document has been developed within the 3rd Generation Partnership Project (3GPP TM) and may be further elaborated for the purposes of 3GPP.
The present document has not been subject to any approval process by the 3GPP Organisational Partners and shall not be implemented.
This Specification is provided for future development work within 3GPP only. The Organisational Partners accept no liability for any use of this Specification.
Specifications and reports for implementation of the 3GPP TM system should be obtained via the 3GPP Organisational Partners' Publications Offices.
Release 8 2 3GPP TS 29.163 V8.16.0 (2011-09)
Keywords
GSM, UMTS, LTE, BICC, ISUP, circuit mode, IP,
network
3GPP
Postal address
Internet
http://www.3gpp.org
Copyright Notification
© 2011, 3GPP Organizational Partners (ARIB, ATIS, CCSA, ETSI, TTA, TTC).
All rights reserved.
UMTS™ is a Trade Mark of ETSI registered for the benefit of its members
3GPP™ is a Trade Mark of ETSI registered for the benefit of its Members and of the 3GPP Organizational Partners
LTE™ is a Trade Mark of ETSI currently being registered for the benefit of its Members and of the 3GPP Organizational Partners
GSM® and the GSM logo are registered and owned by the GSM Association
3GPP
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Contents
Foreword........................................................................................................................................................... 27
1 Scope ...................................................................................................................................................... 28
2 References .............................................................................................................................................. 28
3 Definitions and abbreviations................................................................................................................. 33
3.1 Definitions ....................................................................................................................................................... 33
3.2 Abbreviations ................................................................................................................................................... 34
4 General ................................................................................................................................................... 35
4.1 General interworking overview ....................................................................................................................... 35
5 Network characteristics .......................................................................................................................... 36
5.1 Key characteristics of ISUP/BICC based CS networks ................................................................................... 36
5.2 Key characteristics of IM CN subsystem ......................................................................................................... 36
6 Interworking with CS networks ............................................................................................................. 36
6.1 Interworking reference model .......................................................................................................................... 36
6.1.1 Interworking reference points and interfaces ............................................................................................. 37
6.1.2 Interworking functional entities ................................................................................................................. 37
6.1.2.1 Signalling Gateway Function (SGW) ................................................................................................... 37
6.1.2.2 Media Gateway Control Function (MGCF) .......................................................................................... 37
6.1.2.3 IP Multimedia - Media Gateway Function (IM-MGW) ....................................................................... 37
6.2 Control plane interworking model ................................................................................................................... 37
6.3 User plane interworking model ........................................................................................................................ 37
7 Control plane interworking .................................................................................................................... 38
7.1 General............................................................................................................................................................. 38
7.2 Interworking between CS networks supporting ISUP and the IM CN subsystem ........................................... 39
7.2.1 Services performed by network entities in the control plane ..................................................................... 39
7.2.1.1 Services performed by the SS7 signalling function .............................................................................. 39
7.2.1.2 Services of the SGW............................................................................................................................. 39
7.2.1.3 Services of the MGCF .......................................................................................................................... 39
7.2.1.4 Services of the SIP signalling function ................................................................................................. 39
7.2.2 Signalling interactions between network entities in the control plane ....................................................... 40
7.2.2.1 Signalling between the SS7 signalling function and MGCF ................................................................ 40
7.2.2.1.1 Signalling from MGCF to SS7 signalling function ......................................................................... 40
7.2.2.1.2 Signalling from SS7 signalling function to MGCF ......................................................................... 40
7.2.2.1.3 Services offered by SCTP and M3UA ............................................................................................ 40
7.2.2.1.3.1 Services offered by SCTP ......................................................................................................... 40
7.2.2.1.3.2 Services offered by M3UA........................................................................................................ 40
7.2.2.2 Signalling between the MGCF and SIP signalling function ................................................................. 40
7.2.3 SIP-ISUP protocol interworking ................................................................................................................ 40
7.2.3.1 Incoming call interworking from SIP to ISUP at I-MGCF ................................................................... 41
7.2.3.1.1 Sending of IAM .............................................................................................................................. 41
7.2.3.1.2 Coding of the IAM .......................................................................................................................... 42
7.2.3.1.2.0 General ...................................................................................................................................... 42
7.2.3.1.2.1 Called party number .................................................................................................................. 42
7.2.3.1.2.2 Nature of connection indicators................................................................................................. 43
7.2.3.1.2.3 Forward call indicators .............................................................................................................. 43
7.2.3.1.2.4 Calling party's category ............................................................................................................. 44
7.2.3.1.2.4A Originating Line Information .................................................................................................... 44
7.2.3.1.2.5 Transmission medium requirement ........................................................................................... 45
7.2.3.1.2.5a Transmission medium requirement prime and USI prime (optional) ........................................ 47
7.2.3.1.2.6 Calling party number ................................................................................................................. 47
7.2.3.1.2.7 Generic number ......................................................................................................................... 51
7.2.3.1.2.8 User service information ........................................................................................................... 51
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Annex A (informative): Summary of differences items between 3GPP TS 29.163 and ITU-T
Q.1912.5 ........................................................................................................ 220
A.1 List of differences .................................................................................................................................... 220
Annex B (normative): Codec Negotiation between a BICC CS network and the IM CN
subsystem ...................................................................................................... 221
B.1 Introduction .......................................................................................................................................... 221
B.2 Control plane interworking .................................................................................................................. 221
B.2.1 Incoming call interworking from SIP to BICC at I-MGCF ........................................................................... 221
B.2.1.1 Sending of IAM ........................................................................................................................................ 221
B.2.1.2 Sending of SDP answer ............................................................................................................................ 221
B.2.2 Outgoing call interworking from BICC to SIP at O-MGCF .......................................................................... 222
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Foreword
This Technical Specification has been produced by the 3 rd Generation Partnership Project (3GPP).
The contents of the present document are subject to continuing work within the TSG and may change following formal
TSG approval. Should the TSG modify the contents of the present document, it will be re-released by the TSG with an
identifying change of release date and an increase in version number as follows:
Version x.y.z
where:
y the second digit is incremented for all changes of substance, i.e. technical enhancements, corrections,
updates, etc.
z the third digit is incremented when editorial only changes have been incorporated in the document.
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1 Scope
The present document specifies the principles of interworking between the 3GPP IM CN subsystem and BICC/ISUP
based legacy CS networks, in order to support IM basic voice, data and multimedia calls.
The present document addresses the areas of control and user plane interworking between the IM CN subsystem and CS
networks through the network functions, which include the MGCF and IM-MGW. For the specification of control plane
interworking, areas such as the interworking between SIP and BICC or ISUP are detailed in terms of the processes and
protocol mappings required for the support of both IM originated and terminated voice and multimedia calls.
Other areas addressed encompass the transport protocol and signalling issues for negotiation and mapping of bearer
capabilities and QoS information.
The present document specifies the interworking between 3GPP profile of SIP (as detailed according to 3GPP TS
24.229 [9]) and BICC or ISUP, as specified in ITU-T Recommendations Q.1902.1 to Q.1902.6 [30] and ITU-T Q761 to
Q764 [4] respectively.
The present document also specifies the interworking between circuit switched multimedia telephony service, as
described in 3GPP TS 26.110 [78] 3GPP TS 26.111 [79], and ITU-T Recommendation H.324 [81] and packet switched
multimedia services, as described in 3GPP TS 26.235 [80] and 3GPP TS 26.236 [32], in particular and the interworking
between the 3GPP profile of SIP and the inband control protocols for multimedia communication as specified in ITU-T
Recommendations H.245 [82] and H.324 Annex K [81].
The present document addresses two interworking scenarios with respect to the properties of the CS network:
2 References
The following documents contain provisions which, through reference in this text, constitute provisions of the present
document.
References are either specific (identified by date of publication, edition number, version number, etc.) or
non-specific.
For a non-specific reference, the latest version applies. In the case of a reference to a 3GPP document (including
a GSM document), a non-specific reference implicitly refers to the latest version of that document in the same
Release as the present document.
[1] ITU-T Recommendation G.711 (11/88): "Pulse Code Modulation (PCM) of voice frequencies".
[2] ITU-T Recommendation H.248.1 (05/02): "Gateway control protocol: Version 2".
[3] ITU-T Recommendation Q.701 (03/93), Q.702 (11/88), Q.703 (07/96), Q.704 (07/96), Q.705
(03/93), Q.706 (03/93), Q.707 (11/88), Q.708 (03/99), Q.709 (03/93): "Functional description of
the message transfer part (MTP) of Signalling System No. 7".
[4] ITU-T Recommendations Q.761to Q.764 (12/99): "Specifications of Signalling System No.7
ISDN User Part (ISUP)".
[5] Void.
[7] Void.
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[8] 3GPP TS 24.228: "Signalling flows for the IP multimedia call control based on SIP and SDP".
[9] 3GPP TS 24.229: "IP Multimedia Call Control Protocol based on SIP and SDP".
[11] 3GPP TS 22.228: "Service requirements for the IP Multimedia Core Network Subsystem".
[13] Void.
[14] 3GPP TS 29.205: "Application of Q.1900 series to Bearer Independent CS Network architecture;
Stage 3".
[15] 3GPP TS 29.332: "Media Gateway Control Function (MGCF) – IM-Media Gateway (IM-MGW)
interface, Stage 3".
[20] 3GPP TS 29.202: "Signalling System No. 7 (SS7) signalling transport in core network; Stage 3".
[21] IETF RFC 2474: "Definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6
Headers".
[23] IETF RFC 3267: "Real-Time Transport Protocol (RTP) payload format and file storage format for
the Adaptive Multi-Rate (AMR) Adaptive Multi-Rate Wideband (AMR-WB) audio codecs".
[25] 3GPP TS 29.414: "Core network Nb data transport and transport signalling".
[28] Void.
[29] ITU-T Recommendation Q.2150.1: "Signalling transport converter on MTP3 and MTP3b".
[30] ITU-T Recommendations Q.1902.1 to Q.1902.6 (07/01): "Bearer Independent Call Control".
[31] Void.
[32] 3GPP TS 26.236: "Packet switched conversational multimedia applications; Transport protocols".
[33] Void.
[34] Void.
[35] ITU-T Recommendation Q.765.5: "Signalling system No. 7 – Application transport mechanism:
Bearer Independent Call Control (BICC)".
[36] IETF RFC 3264: "An Offer/Answer Model with the Session Description Protocol (SDP)".
[37] IETF RFC 3312: "Integration of Resource Management and Session Initiation Protocol (SIP)".
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[38] ITU-T Recommendation Q.850 (05/1998) including Amendment 1 (07/2001): "Usage of cause and
location in the Digital Subscriber Signalling System No. 1 and the Signalling System No. 7 ISDN
User Part".
[40] IETF RFC 3323: "A Privacy Mechanism for the Session Initiation Protocol (SIP)".
[41] IETF RFC 3325: "Private Extensions to the Session Initiation Protocol (SIP) for Asserted Identity
within Trusted Networks".
[42] ITU-T Recommendation Q.730 (12/99), Q.731.1 (07/96), Q.731.3 (03/93), Q.731.4 (03/93),
Q.731.5 (03/93), Q.731.6 (03/93), Q.731.7 (06/97), Q.731.8 (02/92), Q.732.5-2 (12/99), Q.732.7
(07/96), Q.733.1 (02/92), Q.733.2 (03/93), Q.733.3 (06/97), Q.733.4 (03/93), Q.733.5 (12/99),
Q.734.1 (03/93), Q.734.2 (07/96), Q.735.1 (03/93), Q.735.3 (03/93), Q.735.6 (07/93), Q.736.1
(10/95), Q.736.3 (10/95), Q.737.1 (06/97): "ISDN user part supplementary services".
[43] ITU-T Recommendation I.363.5 (1996): "B-ISDN ATM Adaptation Layer specification: Type 5
AAL".
[44] ITU-T Recommendation Q.2110 (1994): "B-ISDN ATM adaptation layer - Service Specific
Connection Oriented Protocol (SSCOP)".
[45] ITU-T Recommendation Q.2140 (1995): "B-ISDN ATM adaptation layer - Service specific
coordination function for signalling at the network node interface (SSCF AT NNI)".
[46] ITU-T Recommendation Q.2210 (1996): "Message transfer part level 3 functions and messages
using the services of ITU-T Recommendation Q.2140".
[49] ITU-T Recommendation Q.1912.5 (03/04): "Interworking between Session Initiation Protocol
(SIP) and Bearer Independent Call Control Protocol or ISDN User Part".
[50] 3GPP TS 26.102: "Adaptive Multi-Rate (AMR) speech codec; Interface to Iu and Uu".
[51] IETF RFC 3550: "RTP: A Transport Protocol for Real-Time Applications".
[52] IETF RFC 3551: "RTP Profile for Audio and Video Conferences with Minimal Control".
[53] IETF RFC 3555: "MIME Type Registration of RTP Payload Formats".
[57] 3GPP TS 26.103: "Speech Codec List for GSM and UMTS".
[58] 3GPP TS 28.062: "Inband Tandem Free Operation (TFO) of speech codecs".
[59] IETF RFC 3556: "Session Description Protocol (SDP) Bandwidth Modifiers for RTP Control
Protocol (RTCP) bandwidth".
[60] 3GPP TS 24.604: "Communication Diversion (CDIV) using IP Multimedia (IM) Core Network
(CN) subsystem; Protocol specification".
[61] 3GPP TS 24.605 "Conference (CONF) using IP Multimedia (IM) Core Network (CN) subsystem;
Protocol specification".
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[62] 3GPP TS 24.606: "Message Waiting Indication (MWI) using IP Multimedia (IM) Core Network
(CN) subsystem; Protocol specification".
[63] 3GPP TS 24.607: "Originating Identification Presentation (OIP) and Originating Identification
Restriction (OIR) using IP Multimedia (IM) Core Network (CN) subsystem; Protocol
specification".
[64] 3GPP TS 24.608 "Terminating Identification Presentation (TIP) and Terminating Identification
Restriction (TIR) using IP Multimedia (IM) Core Network (CN) subsystem; Protocol
specification".
[65] 3GPP TS 24.610: "Communication HOLD (HOLD) using IP Multimedia (IM) Core Network
(CN) subsystem; Protocol specification".
[66] Void.
[67] 3GPP TS 24.611: "Anonymous Communication Rejection (ACR) and Communication Barring
(CB) using IP Multimedia (IM) Core Network (CN) subsystem; Protocol specification".
[68] Void.
[69] IETF RFC 4040: "RTP Payload Format for a 64 kbit/s Transparent Call".
[70] ETSI EN 300 356-1 (V4.2.1): "Integrated Services Digital Network (ISDN); Signalling System
No.7 (SS7); ISDN User Part (ISUP) version 4 for the international interface; Part 1: Basic services
[ITU-T Recommendations Q.761 to Q.764 (1999) modified]".
[71] ETSI EN 300 356-21 (V4.2.1): "Integrated Services Digital Network (ISDN); Signalling System
No.7 (SS7); ISDN User Part (ISUP) version 4 for the international interface; Part 21: Anonymous
Call Rejection (ACR) supplementary service".
[72] ITU-T Recommendation T.38 (06/98): "Procedures for real-time Group 3 facsimile
communication over IP networks".
[73] IETF RFC 3362: "Real-time Facsimile (T.38) - image/t38 MIME Sub-type Registration".
[75] IETF RFC 3515: "The Session Initiation Protocol (SIP) REFER method".
[76] Void.
[77] IETF RFC 5079: "Rejecting Anonymous Requests in the Session Initiation Protocol (SIP)".
[78] 3GPP TS 26.110: "Codec for circuit switched multimedia telephony service; General description".
[79] 3GPP TS 26.111: "Codec for Circuit switched Multimedia Telephony Service; Modifications to
H.324".
[80] 3GPP TS 26.235: "Packet switched conversational multimedia applications; Default codecs".
[81] ITU-T Recommendation H.324 (04/09): "Terminal for low bitrate multimedia communication".
[83] ITU-T Recommendation H.261 (03/93): "Video codec for audiovisual services at p x 64 kbit/s".
[84] ITU-T Recommendation H.263 (01/05): "Video coding for low bitrate communication".
[85] Void.
[86] Void.
[87] Void.
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[88] 3GPP TS 24.173: "IMS Multimedia Telephony Communication Service and Supplementary
Services, stage 3".
[89] IETF RFC 5009: "Private Header (P-Header) Extension to the Session Initiation Protocol (SIP) for
Authorization of Early Media".
[90] IETF RFC 2663: "IP Network Address Translator (NAT) Terminology and Considerations".
[91] IETF RFC 4244: "An extension to the Session Initiation Protocol (SIP) for Request History
Information".
[92] ITU-T Recommendation Q.769.1 (12/99): "Signalling system No. 7 – ISDN user part
enhancements for the support of number portability".
[93] IETF RFC 4694: "Number portability parameters for the "tel" URI".
[94] Void.
[95] Void.
[96] ETSI EN 300 403-1 (V1.3.2): "Integrated Services Digital Network (ISDN); Digital Subscriber
Signalling System No. one (DSS1) protocol; Signalling network layer for circuit-mode basic call
control; Part 1: Protocol specification [ITU-T Recommendation Q.931 (1993), modified]".
[97] IETF RFC 3966: "The tel URI for Telephone Numbers".
[98] Void.
[99] IETF draft-johnston-sipping-cc-uui-03 "Transporting User to User Information for Call Centers
using SIP".
Editors Note: The reference to draft draft-johnston- sipping-cc-uui is needs to be replaced by a reference to an IETF
sip working group document providing the final definition of the SIP UUI header, when such a document
becomes available. Those drafts can not be formally referenced until they become RFCs.
[100] IETF RFC 4575: "A Session Initiation Protocol (SIP) Event Package for Conference State".
[101] 3GPP TS 24.654: "Closed User Group (CUG) using IP Multimedia (IM) Core Network (CN)
subsystem, Protocol Specification".
[103] 3GPP TS 23.014: "Technical Specification Group Core Network; Support of Dual Tone Multi-
Frequency (DTMF) signalling".
[104] 3GPP TS 26.114: "IP Multimedia Subsystem (IMS); Multimedia Telephony; Media handling and
interaction".
[105] IETF RFC 4733: "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals".
[106] IETF draft-ietf-bliss-call-completion-08: "Call Completion for Session Initiation Protocol (SIP)".
Editor's note: The above document cannot be formally referenced until it is published as an RFC.
[107] IETF draft-salud-alert-info-urns-00: "Alert-Info URNs for the Session Initiation Protocol (SIP)".
Editor's note: The above document cannot be formally referenced until it is published as an RFC.
[108] IETF RFC 4715: "The Integrated Services Digital Network (ISDN) Subaddress Encoding Type for
tel URI".
[109] IETF RFC 4585: "Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based
Feedback (RTP/AVPF)".
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[110] IETF RFC 5104: "Codec Control Messages in the RTP Audio-Visual Profile with Feedback
(AVPF)".
[111] 3GPP TS 24.615: "Communication Waiting (CW) using IP Multimedia (IM) Core Network (CN)
subsystem, Protocol Specification".
[113] IETF RFC 4458: "Session Initiation Protocol (SIP) URIs for Applications such as Voicemail and
Interactive Voice Response (IVR)".
[114] IETF RFC 5993: "RTP Payload format for Global System for Mobile Communications Half Rate
(GSM-HR)".
Editor's note: The above document cannot be formally referenced until it is published as an RFC.
[116] IETF RFC 3326: "The Reason Header Field for the Session Initiation Protocol (SIP)".
[118] 3GPP TS 23.172: "Technical realization of Circuit Switched (CS) multimedia service UDI/RDI
fallback and service modification; Stage 2".
3.1 Definitions
For the purposes of the present document, the terms and definitions given in 3GPP TR 21.905 [6], ITU-T
Recommendation E.164 [48] and the following apply:
SS7 signalling function: function in the CS network, which has the capabilities to transport the SS7 MTP-User parts
ISUP and BICC+STCmtp
SIP signalling function: function in the IM CN subsystem, which has the capabilities to transport SIP
Incoming or Outgoing: used in the present document to indicate the direction of a call (not signalling information)
with respect to a reference point.
Incoming MGCF (I-MGCF): entity that terminates incoming SIP calls from the IMS side and originates outgoing calls
towards the CS side using the BICC or ISUP protocols.
Outgoing Interworking Unit (O-MGCF): entity that terminates incoming BICC or ISUP calls from the CS side and
originates outgoing calls towards the IMS using SIP.
Root Termination: refers to Media Gateway as an entity in itself, rather than a Termination within it. A special
TerminationID, "Root" is reserved for this purpose. See ITU-T Recommendation H.248.1. [2]
Signalling Transport Converter (STC): function that converts the services provided by a particular Signalling
Transport to the services required by the Generic Signalling Transport Service.
STCmtp: Signalling Transport Converter on MTP. See ITU-T Recommendation Q.2150.1 [29].
BICC+STCmtp: this terminology means that BICC signalling always needs to be used on top of STCmtp sublayer.
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3.2 Abbreviations
For the purposes of the present document, the abbreviations as defined in 3GPP TR 21.905 [6] and the following apply:
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4 General
For the ability to support the delivery of basic voice calls between the IM CN subsystem and CS networks, basic
protocol interworking between SIP (as specified in 3GPP TS 24.229 [9]) and BICC or ISUP (as specified in ITU-T
Recommendations Q.1902.1-6 [30] and ITU-T Recommendations Q761 to Q764 [4] respectively) has to occur at a
control plane level, in order that call setup, call maintenance and call release procedures can be supported. The MGCF
shall provide this protocol mapping functionality within the IM CN subsystem.
User plane interworking between the IM CN subsystem and CS network bearers (e.g. 64k TDM, ATM/AAL2 circuit or
IP bearer) are supported by the functions within the IM-MGW. The IM-MGW resides in the IM CN subsystem and
shall provide the bearer channel interconnection. The MGCF shall provide the call control to bearer setup association.
The IM CN subsystem shall interwork, at the control and user plane, with BICC and ISUP based legacy CS networks.
The support of supplementary services shall be as defined in 3GPP TS 22.228 [11]. The MGCF and IM-MGW shall
support the interworking of the IM CN subsystem to an external ISUP based CS network. They may also support
interworking to a BICC based CS network where no 3GPP specific extension is applied. The MGCF and the IM-MGW
may also support interworking to a BICC based CS network where 3GPP specific extensions in accordance with 3GPP
TS 29.205 [14] are applied.
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5 Network characteristics
The interface towards a CS-PLMN may either be one of the interfaces mentioned in the paragraph above or a signalling
interface based on BICC with 3GPP specific extensions, as specified for the 3GPP Nc interface in 3GPP TS 29.205
[14], and the IM-MGW may support the 3GPP Nb interface, as specified in 3GPP TS 29.414 [25] and 3GPP TS 29.415
[26]. If the 3GPP Nc interface is applied as signalling interface, the 3GPP Nb interface is used as user plane interface
and the Nb UP Framing protocol is applied.
BGCF Mj
BICC/ISUP over
SCTP/IP
CSCF MGCF SGW
Mg
BICC/ISUP
Mn BICC over SCTP/IP
over MTP
CS channels
Mb IM- e.g. PCM
(Note 3) MGW CS network
User Plane
Control Plane
NOTE 1: The logical split of the signalling and bearer path between the CS network and the IM CN subsystem is as
shown, however the signalling and bearer may be logically directly connected to the IM-MGW.
NOTE 2: The SGW may be implemented as a stand-alone entity or it may be located in another entity either in the
CS network or the IM-MGW. The implementation options are not discussed in the present document.
NOTE 3: The IM-MGW may be connected via the Mb to various network entities, such as a UE (via a GTP Tunnel to
a GGSN), an MRFP, or an application server.
NOTE 4: A SGW function is not required for certain signalling transports, where M3UA+SCTP+IP is used in CS
network and IM-MGCF.
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Protocol for Mg reference point: The single call control protocol applied across the Mg reference point (i.e. between
CSCF and MGCF) will be based on the 3GPP profile of SIP as defined in accordance with 3GPP TS 24.229 [9].
Protocol for Mn reference point: The Mn reference point describes the interfaces between the MGCF and IM-MGW,
and has the properties as detailed in 3GPP TS 29.332 [15].
Protocol for Mj reference point: The single call control protocol applied across the Mj reference point (i.e. between
BGCF and MGCF) will be based on the 3GPP profile of SIP as defined in accordance with 3GPP TS 24.229 [9].
Protocol for Mb reference point: The Mb reference point is defined in accordance with 3GPP TS 23.002 [10] and is
IPv6 based.
The functionality defined within MGCF shall be defined in accordance with 3GPP TS 23.002 [10].
External CS networks use BICC or ISUP to originate and terminate voice calls to and from the IM CN subsystem.
Therefore, in order to provide the required interworking to enable inter network session control, the control plane
protocols shall be interworked within the IM CN subsystem. This function is performed within the MGCF (see clause
6.1.2).
External legacy CS networks use circuit switched bearer channels like TDM circuits (e.g. 64 kbits PCM), ATM/AAL2
circuit or IP bearers to carry encoded voice frames, to and from the IM CN subsystem.
Other CN networks use ATM/AAL 1 or AAL 2 or IP as a backbone, with different framing protocols.
3GPP
Release 8 26 3GPP TS 29.163 V8.16.0 (2011-09)
Therefore, in order to provide the required interworking to enable media data exchange, the user plane protocols shall
be translated within the IM CN subsystem. This function is performed within the IM-MGW (see clause 6.1.2).
The transport of SS7 signalling protocol messages of any protocol layer that is identified by MTP level 3, in SS7 terms,
as a user part (MTP3-user) shall be accomplished in accordance with the protocol architecture defined in the following
clauses. For the present document these protocol layers include, but are not limited to, Bearer Independent Call Control
(BICC)+STCmtp and ISDN User Part (ISUP).
7.1 General
The following sub-clauses define the signalling interworking between the Bearer Independent Call Control (BICC) or
ISDN User Part (ISUP) protocols and Session Initiation Protocol (SIP) with its associated Session Description Protocol
(SDP) at a MGCF. The MGCF shall act as a Type An exchange (ITU-T Recommendation Q.764 [4]) for the purposes
of ISUP and BICC Compatibility procedures. The services that can be supported through the use of the signalling
interworking are limited to the services that are supported by BICC or ISUP and SIP based network domains.
BICC is the call control protocol used between Nodes in a network that incorporates separate call and bearer control.
The BICC/ISUP capabilities or signalling information defined for national use is outside the scope of the present
document. It does not imply interworking for national-specific capabilities is not feasible.
The capabilities of SIP and SDP that are interworked with BICC or ISUP are defined in 3GPP TS 24.229 [9]
Services that are common in SIP and BICC or ISUP network domains will seamlessly interwork by using the function
of the MGCF. The MGCF will originate and/or terminate services or capabilities that do not interwork seamlessly
across domains according to the relevant protocol recommendation or specification.
Table 1 lists the services seamlessly interworked and therefore within the scope of the present document.
Table 1: Interworking Capabilities between BICC/ISUP and SIP profile for 3GPP
Service
Speech/3.1 kHz audio
CS data Calls (optional)
En bloc address signalling
Overlap address signalling from the CS side towards the IMS
Out of band transport of DTMF tones and information. (BICC only)
Inband transport of DTMF tones and information. (BICC and ISUP)
Direct-Dialling-In (DDI)
Multiple Subscriber Number (MSN)
Calling Line Identification Presentation (CLIP)
Calling Line Identification Restriction (CLIR)
Connected line presentation (COLP)
Connected line restriction (COLR)
Carrier routeing
3GPP
Release 8 27 3GPP TS 29.163 V8.16.0 (2011-09)
In order to support the seamless operation of the MTP3-User part information between networks incorporating SS7 and
IP (either IPv4, see RFC 791 [16], or IPv6, see RFC 2460 [39]), the SGW shall support the services of MTP as well as
the services of the M3UA (see 3GPP TS 29.202 [20]) and SCTP (see RFC 2960 [18]).
The MGCF shall provide the interaction, through the use of its interworking function, between the SS7 MTP3-User part
information, e.g. ISUP, and SIP. The MGCF interworking function shall also provide the translation between the SS7
MTP3-User part information and SIP, where the interworking of SIP to ISUP and BICC+STCmtp are detailed below.
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Release 8 28 3GPP TS 29.163 V8.16.0 (2011-09)
SCTP offers the ability to reliably transfer the SCTP User applications, e.g. M3UA, between the SCTP User application
peers. The initialization procedure used for an association between two SCTP end-to-end peers, and the initialization to
the SCTP User applications shall be performed as detailed in RCF 2960 [18].
When an association between two SCTP peers has been established, the use of M3UA shall provide the transport
service in accordance with MTP (see ITU-T Recommendations Q.701 to Q.709 [3]) to the MTP3-User, e.g. ISUP.
The naming and addressing concepts between the MGCF and SIP signalling function shall be detailed in accordance
with 3GPP TS 23.228 [12]. The issues of general IP address management are discussed in 3GPP TS 23.221 [47].
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Release 8 29 3GPP TS 29.163 V8.16.0 (2011-09)
An I-MGCF shall support both incoming INVITE requests containing SIP preconditions and 100rel extensions in the
SIP Supported or Require header fields, and INVITE requests not containing these extensions, unless the Note below
applies.
NOTE: If the I-MGCF is deployed in an IMS network that by local configuration serves no user requiring
preconditions, the MGCF may not support incoming requests requiring preconditions.
The I-MGCF shall interwork forked INVITE requests with different request URIs.
If the SIP precondition extension is not included in the Supported or Require header field, the I-MGCF shall send an
IAM immediately after the reception of the INVITE, as shown in figure 3. The I-MGCF shall set the continuity
indicators to "Continuity check not required".
If a Continuity Check procedure is supported in the ISUP network and SIP precondition extension are included in the
SIP Supported or Require header field, the I-MGCF shall send the IAM immediately after the reception of the INVITE,
as shown in figure 3. If the received SDP indicates that precondition is fulfilled the I-MGCF shall set the continuity
indicators to "continuity check is not required". If the received SDP indicates that precondition is not fulfilled the I-
MGCF shall set the continuity indicators to "continuity check performed on a previous circuit". The procedure in figure
3 applies when the value of the continuity indicator is either set to "continuity check required", "continuity check
performed on a previous circuit" or "continuity check not required". If the continuity indicator is set to "continuity
check required" the corresponding procedures at the Mn interface described in clause 9.2.2.3 also apply.
I-MGCF
INVITE
IAM
Figure 3: Receipt of an INVITE request (continuity procedure supported in the ISUP network)
If Continuity Check procedure is not supported in the ISUP network, and the SDP in the received INVITE request
contains preconditions not met, the I-MGCF shall delay sending the IAM until the SIP preconditions are met and set the
continuity indicators in the resulting IAM to "Continuity check not required".
I-MGCF
INVITE
SDP indicating
IAM
pre-conditions
met
Figure 4: Receipt of an INVITE request (continuity procedure not supported in the ISUP network)
3GPP
Release 8 30 3GPP TS 29.163 V8.16.0 (2011-09)
The I-MGCF shall reject an INVITE request for a session only containing unsupported media types by sending a status
code 488 "Not Acceptable Here". If several media streams are contained in a single INVITE request, and if the I-MGCF
does not support multimedia interworking according to Annex E, then the I-MGCF shall select one of the supported
media streams, reserve the codec(s) for that media stream, and reject the other media streams and unselected codecs in
the SDP answer, as detailed in IETF RFC 3264 [36]. If supported audio media stream(s) and supported non-audio
media stream(s) are contained in a single INVITE request, an audio stream should be selected.
The I-MGCF shall include a To tag in the first backward non-100 provisional response, in order to establish an early
dialog as described in IETF RFC 3261 [19].
If an MGCF discovers an emergency call it shall, depending on national requirements, map that to appropriate
indication in ISUP.
According to IETF RFC 3261 [19] and IETF RFC 3264 [36], if an INVITE message is received without an SDP offer,
then the I-MGCF sends an SDP offer in the first reliable non-failure message.
7.2.3.1.2.0 General
The following ISDN user part parameters description can be found in ITU-T Recommendation Q.763 [4].
The E.164 address encoded in the Request-URI shall be mapped to the called party number parameter of the IAM
message.
INVITE IAM
Request-URI Called Party Number
E.164 address Address Signal:
(format +CC Analyse the information contained in received E.164 address.
NDC SN) If CC is country code of the network in which the next hop terminates, then remove "+CC"
(e.g. as User info and use the remaining digits to fill the Address signals.
portion of a SIP If CC is not the country code of the network in which the next hop terminates, then remove
URI with "+" and use the remaining digits to fill the Address signals.
user=phone, or (NOTE 2)
as tel URI) Odd/even indicator: set as required
Nature of address indicator:
Analyse the information contained in received E.164 address.
If CC is country code of the network in which the next hop terminates, then set Nature of
Address indicator to "National (significant) number".
If CC is not the country code of the network in which the next hop terminates, then set
Nature of Address indicator to "International number".
(NOTE 1)
Internal Network Number Indicator:
1 routing to internal network number not allowed
Numbering plan Indicator:
001 ISDN (Telephony) numbering plan (Rec. E.164)
national operator Address Signal:
option for service use received non E.164 number to fill the Address signals with national
numbers: significant number.
Non E.164 Odd/even indicator: set as required
numbers Nature of address indicator:
(as a local- (NOTE 3)
"National (significant) number". (NOTE 1)
number with a Internal Network Number Indicator:
phone-context in 1 routing to internal network number not allowed
the User Info Numbering plan Indicator:
portion in a SIP 001 ISDN (Telephony) numbering plan (Rec. E.164)
URI with Address Signal:
user=phone, or (NOTE 3)
Use received non E.164 number to fill the Address signals.
3GPP
Release 8 31 3GPP TS 29.163 V8.16.0 (2011-09)
00 continuity check not required, if the continuity check procedure is not supported in the succeeding
network (figure 4).
01 continuity check required, if a continuity check shall be carried out on the succeeding circuit.
(figure 3)
10 continuity check performed on a previous circuit otherwise, if the continuity check procedure is
supported in the succeeding network, but shall not be carried out on the succeeding circuit otherwise.
(figure 3)
1 outgoing echo control device included, for speech calls, e.g., TMR is "3.1KHz audio".
0 outgoing echo control device not included, for known data calls, e.g., TMR "64 kBit/s unrestricted"
or HLC "Facsimile Group 2/3".
1 interworking encountered
As a network operator option, the value D = 0 "No interworking encountered" is used if the TMR = 64 kBit/s
unrestricted is used.
NOTE: This avoids sending of a progress indicator with progress information 0 0 0 0 0 0 1 "Call is not end-to-end
ISDN; further call progress information may be available in-band", so the call will not be released for that
reason by an ISDN terminal.
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Release 8 32 3GPP TS 29.163 V8.16.0 (2011-09)
As a network operator option, the value F = 1 "ISDN user part/BICC used all the way" is used if the TMR = 64 kBit/s
unrestricted is used.
NOTE: This avoids sending of a progress indicator with progress information 0 0 0 0 0 0 1 "Call is not end-to-end
ISDN; further call progress information may be available in-band", so the call will not be released for that
reason by an ISDN terminal.
if any used supplementary service requires ISUP or BICC all the way, depending on operator policy:
Otherwise:
As a network operator option, the value I = 1 "originating access ISDN" is used if the TMR = 64 kBit/s unrestricted is
used.
NOTE: This avoids sending of a progress indicator with progress information 0 0 0 0 1 1 "Originating access is
non-ISDN", so the call will not be released for that reason by an ISDN terminal.
00 no indication
If the PSTN XML is supported as a network option, the Forward Call indicators derived as shown in Table 02a shall
take precedence.
Table 02a: Mapping of PSTN XML elements to Forward call indicators parameter
INVITE IAM
PSTN XML Forward call indicators parameter
PSTN XML with Progress indicator bit D Interworking Indicator
with Progress Description value 6 0 "no interworking encountered (No. 7 signalling all the way)"
(Meaning: originating access ISDN) bit F ISDN User Part indicator
1 "ISDN User Part used all the way"
bit I ISDN access indicator
1 "originating access ISDN"
NOTE: Progress Indicator with Progress Description value "6" shall not be included in an ATP within the IAM.
The ISUP Originating Line Information parameter is defined by ANSI Standard ATIS-1000113 [117], Chapter 3.
See Annex H for the normative interworking of the OLI parameter as a network option.
3GPP
Release 8 33 3GPP TS 29.163 V8.16.0 (2011-09)
The I-MGCF may either transcode the selected codec(s) to the codec on the PSTN side or it may attempt to interwork
the media without transcoding. If the I-MGCF transcodes, it shall select the TMR parameter to "3.1 kHz audio". If the I-
MGCF does not transcode, it should map the TMR, USI and Access Transport parameters from the selected codec
according to Table 2a. However, if the I-MGCF supports the PSTN XML body as a network option, and if a PSTN
XML body is received in the INVITE request and the I-MGCF selects media encoded in any of the formats in Table 2a
(G.711, Clearmode or t38) among the offered media, the I-MGCF shall derive these parameters from the PSTN XML
body instead, as detailed in Table 2b. The I-MGCF should only apply the mapping in Table 2b if the TMR and USI
values derived from the selected codec according to Table 2a are equivalent with the values within the first Bearer
Capability element in the PSTN XML, and otherwise the I-MGCF should apply the mapping according to Table 2a.
If no SDP is received from the remote peer (as described in clause 7.2.3.1.1), then the TMR parameter should be set to
"3.1 kHz audio". Transcoding shall be applied as required.
3GPP
Release 8 34 3GPP TS 29.163 V8.16.0 (2011-09)
m= line b= line (NOTE 4) a= line TMR parameter USI parameter (optional) (NOTE 1) HLC IE in the
ATP parameter
(optional)
<media> <transport> <fmt-list> <modifier>:<bandwidth- rtpmap:<dynamic-PT> TMR codes Information User Information High Layer
value> <encoding name> Transfer Layer 1 Protocol Characteristics
<clock rate>[<encoding Capability Indicator Identification
(NOTE 5) parameters>]
audio RTP/AVP 0 N/A or AS: up to (64 kbit/s N/A "3.1KHz audio" (NOTE 3)
+ RTP/UDP/IP overhead)
audio RTP/AVP Dynamic PT N/A or AS: up to (64 kbit/s rtpmap:<dynamic-PT> "3.1KHz audio" (NOTE 3)
+ RTP/UDP/IP overhead) PCMU/8000
audio RTP/AVP 8 N/A or AS: up to (64 kbit/s+ N/A "3.1KHz audio" (NOTE 3)
RTP/UDP/IP overhead)
audio RTP/AVP Dynamic PT N/A or AS: up to (64 kbit/s rtpmap:<dynamic-PT> "3.1KHz audio" (NOTE 3)
+ RTP/UDP/IP overhead) PCMA/8000
audio RTP/AVP Dynamic PT AS: (64 kbit/s+ rtpmap:<dynamic-PT> "64 kbit/s unrestricted" "Unrestricted
RTP/UDP/IP overhead) CLEARMODE/8000 or digital
(NOTE 2) "64 kbit/s preferred" information"
(NOTE 7) (NOTE 6)
image Udptl [73] t38 [73] N/A or AS: up to (64 kbit/s Based on ITU-T T.38 "3.1 KHz audio" "3.1 KHz audio" "Facsímile
+ UDP/IP overhead) [72] Group 2/3"
image tcp t38 [73] N/A or AS: up to (64 kbit/s Based on ITU-T T.38 "3.1 KHz audio" "3.1 KHz audio" "Facsímile
+ TCP/IP overhead) [72] Group 2/3"
NOTE 1: In this table the codec G.711 is used only as an example. Other codecs are possible.
NOTE 2: CLEARMODE is specified in RFC4040 [69].
NOTE 3: HLC is normally absent in this case. It is possible for HLC to be present with the value "Telephony", although 6.3.1/Q.939 indicates that this would normally be
accompanied by a value of "Speech" for the Information Transfer Capability element.
NOTE 4: The MGCF should return a b:AS bandwidth modifier with a bandwidth of 64kbit/s + RTP/UDP/IP overhead in the SDP answer to request that the peer does not send
with a higher bandwidth. If the received b=line indicates a bandwidth greater than 64kbit/s + RTP/UDP/IP overhead, the MGCF should also accept the incoming call.
NOTE 5: <bandwidth value> for <modifier> of AS is in units of kbit/s.
NOTE 6: In the case where the Clearmode codec appears together with speech codecs in the same m-line, the value "Unrestricted digital inf. w/tones/ann" is applicable but is
mapped into the USI prime parameter (see sub-clause 7.2.3.1.2.5a).
NOTE 7: The value "64 k/bits preferred" should only be used if the Clearmode codec appears together with speech codecs in the same m-line and two PSTN XML Bearer
Capability elements appear in the initial INVITE request as described in the sub-clause 7.2.3.1.2.5a.
3GPP
Release 8 35 3GPP TS 29.163 V8.16.0 (2011-09)
INVITE IAM
PSTN XML Value ISUP Parameter Content
HighLayerCompatibility Access Transport High layer compatibility (NOTE 1)
Parameter
LowLayerCompatibility Low layer compatibility
BearerCapability (NOTE 2) User Service
Information
HighLayerCompatibility User Tele Service High layer compatibility
BearerCapability(Information Speech TMR Speech
TransferCapability) (NOTE
2)
3.1 kHz audio 3.1 kHz audio
Unrestricted digital 64 kbit/s unrestricted
information
Unrestricted digital 64 kbit/s unrestricted
information with
tones/announcements
NOTE 1: If two high layer compatibility information elements are received, they shall be transferred in the same
order as received in the PSTN XML body in the INVITE message.
NOTE 2: If there are two BCs present, see subclause 7.2.3.1.2.5a.
NOTE 3: The above mapping assumes that there is only a single BearerCapability present.
The Fallback mechanism described in the present Clause shall only apply if two PSTN XML Bearer Capability
elements appear within the INVITE Request and the MGCF supports the PSTN XML body as a network option.
When the INVITE request includes multiple audio codecs with codec appearing in Table 2a then the I-MGCF shall:
- If the first stated codec in the INVITE is the equivalent as stated within the second Bearer Capability element in
the PSTN XML then the I-MGCF shall map the second XML Bearer Capability element into the USI prime and
shall set the TMR to "64 kBit/s preferred".
- If the second stated codec in the INVITE is the equivalent as stated within the first Bearer Capability element in
the PSTN XML then the I-MGCF shall map the first XML Bearer Capability element into the USI and shall map
the TMR prime from the first PSTN XML BearerCapability (InformationTransferCapability).
- If the compared codec stated within the INVITE is not equivalent as stated within XML Bearer Capability
element then the XML Bearer Capability element shall be discarded.
If the Fallback mechanism is not supported by the succeeding network the procedures as described within clause
7.2.3.1.2.5 shall apply, using the first Bearer Capability element in the PSTN XML.
In cases where the fallback appears within the terminating entity and sends back a codec to which is fallen back then the
I-MGCF shall only apply the cut-though to the chosen codec.
In cases where the fallback appears within the terminating entity and sends back a SDP Answer that is equivalent with
the TMR prime codec (fallback codec) then the I-MGCF shall only apply the cut-though to the fallback codec.
In cases where preconditions are used the I-MGCF has to wait for the SDP answer where the preconditions are met and
fallback codec is sent back.
The SIP "Privacy" header is defined within IETF RFC 3323 [40]. The SIP "P-Asserted-Identity" header is defined in
IETF RFC 3325 [41].
3GPP
Release 8 36 3GPP TS 29.163 V8.16.0 (2011-09)
Has a "P- Has a "From" Calling Party Calling Party Number Generic Generic
Asserted- header field Number parameter Number Number
Identity" (NOTE 3) parameter APRI (additional parameter
header field containing a URI Address signals calling party APRI
(NOTE 2, that encodes an number)
NOTE 5, E.164 address address
NOTE 6) been received signals
been (NOTE 6)?
received?
No No Network option to Network option to set Parameter not Not
either include a APRI to "presentation included applicable
network provided restricted" or
E.164 number "presentation allowed"
(See table 4) or (NOTE 4)
omit the Address (See table 5)
signals. As a network option the
(NOTE 4) APRI "presentation
restricted by network"
(NOTE 7) can be used
instead of the APRI
"presentation restricted"
No Yes Network Option to Network option to set Network APRI =
either include a APRI to "presentation Option to "presentation
network provided restricted" or either omit the restricted" or
E.164 number "presentation allowed" parameter (if "presentation
(See table 4) or (NOTE 4) CgPN has allowed"
omit the Address (See table 5) been omitted) depending on
signals. As a network option the or derive from SIP Privacy
(NOTE 4) APRI "presentation the "From" header.
restricted by network" header (See table 6)
(NOTE 7) can be used (NOTE 1)
instead of the APRI (See table 6)
"presentation restricted"
Yes No Derive from APRI = "presentation Not included Not
P-Asserted- restricted" or applicable
Identity "presentation allowed"
(See table 5) depending on SIP
Privacy header.
(See table 5)
Yes Yes Derived from APRI = "presentation Network APRI =
P-Asserted- restricted" or Option to "presentation
Identity "presentation allowed" either omit the restricted" or
(See table 5) depending on SIP parameter or "presentation
Privacy header. derive from the allowed"
(See table 5) "From" header depending on
(NOTE 1) SIP Privacy
(See table 6) header.
(see table 6)
3GPP
Release 8 37 3GPP TS 29.163 V8.16.0 (2011-09)
Has a "P- Has a "From" Calling Party Calling Party Number Generic Generic
Asserted- header field Number parameter Number Number
Identity" (NOTE 3) parameter APRI (additional parameter
header field containing a URI Address signals calling party APRI
(NOTE 2, that encodes an number)
NOTE 5, E.164 address address
NOTE 6) been received signals
been (NOTE 6)?
received?
NOTE 1: This mapping effectively gives the equivalent of Special Arrangement to all SIP UAC with access to the
I-MGCF.
NOTE 2: It is possible that the P-Asserted-Identity header field includes both a tel URI and a sip or sips URI. In this
case, either the tel URI or the SIP URI with user="phone" and a specific host portion, as selected by
operator policy, may be used.
NOTE 3: The "From" header may contain an "Anonymous User Identity". An "Anonymous User Identity" includes
information that does not point to the calling party. IETF RFC 3261 recommends that the display-name
component contain "Anonymous". That the Anonymous User Identity will take the form defined in 3GPP TS
23.003 [74]. The Anonymous User Identity indicates that the calling party desired anonymity. The From
header may also contain an Unavailable User Identity as defined in 3GPP TS 23.003 [74], that indicates
that the calling party is unknown.
NOTE 4: A national option exists to set the APRI to "Address not available".
NOTE 5: 3GPP TS 24.229 guarantees that the received number is an E.164 number formatted as an international
number, with a "+" sign as prefix.
NOTE 6: The E.164 numbers considered within the present document are composed by a Country Code (CC),
followed by a National Destination Code (NDC), followed by a Subscriber Number (SN). On the IMS side,
the numbers are international public telecommunication numbers ("CC"+"NDC"+"SN") and are prefixed by a
"+" sign. On the CS side, it is a network option to omit the CC.
NOTE 7: This is an ETSI specific value described within ETSI EN 300 356-1 [70].
Table 4: Setting of the network-provided BICC/ISUP calling party number parameter with a CLI
(network option)
3GPP
Release 8 38 3GPP TS 29.163 V8.16.0 (2011-09)
Table 5: Mapping of P-Asserted-Identity and privacy headers to the ISUP/BICC calling party number
parameter
3GPP
Release 8 39 3GPP TS 29.163 V8.16.0 (2011-09)
Table 6: Mapping of SIP from header field to BICC/ISUP generic number (additional calling party
number) parameter (network option)
The I-MGCF shall perform the following interworking procedure if the Hop Counter procedure is supported in the CS
network.
At the I-MGCF the Max-Forwards SIP header shall be used to derive the Hop Counter parameter if applicable. Due to
the different default values (that are based on network demands/provisions) of the SIP Max-Forwards header and the
Hop Counter, a factor shall be used to adapt the Max Forwards to the Hop Counter at the I-MGCF. For example, the
following guidelines could be applied:
1) Max-Forwards for a given message should be monotone decreasing with each successive visit to a SIP entity,
regardless of intervening interworking, and similarly for Hop Counter.
2) The initial and successively mapped values of Max-Forwards should be large enough to accommodate the
maximum number of hops that may be expected of a validly routed call.
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Release 8 40 3GPP TS 29.163 V8.16.0 (2011-09)
The Principle of adoption could be implemented on a basis of the network provision, trust domain rules and bilateral
agreement.
If the I-MGCF supports the PSTN XML body as a network option and an INVITE containing a PSTN XML body is
received, an available "ProgressIndicator" element in the PSTN XML body shall be mapped into a Progress Indicator in
the Access Transport Parameter of the sent IAM as shown in table 7.2.3.1.2.10.1.
INVITE IAM
PSTN XML Access Transport Parameter
Progress indicator Progress indicator (NOTE)
NOTE: A ProgressIndicator with Progress Description value 6 shall not be included into the ISUP ATP, and is
mapped instead to Forward call indicators parameter according to table 02a.
7.2.3.1.2A.1 Coding of the IAM when a Number Portability Routing Number is available
ITU-T Q.769.1 [92] describes three possible addressing methods for signalling of the Called Party E.164 address and
Number Portability Routing Number (ITU-T Q.769.1 [92] uses the terms directory number and network routing number
respectively). The choice of these methods is based on network operator and national requirements.
The following sub-clauses describe how the IAM is populated, based on these methods, when a Number Portability
Routing Number is available in the Request URI in the form of a Tel URI "rn=" parameter.
When the optional Number Portability Routing Number is available and supported, these procedures take precedence
over procedures for coding of the Called Party Number described in clause 7.2.3.1.2.1.
If the Number Portability Database Dip Indicator is present within the Request-URI the procedures described in clause
7.2.3.1.2A.2 apply. When a Number Portability Routing Number is not available, the Called Party Number parameter is
populated as described in clause 7.2.3.1.2.1.
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Release 8 41 3GPP TS 29.163 V8.16.0 (2011-09)
Table 7a.0a: Coding of the called party number and called directory number with Number Portability
Separate Directory Number Addressing Method
INVITE IAM
Request-URI Called Party Number Called Directory Number
Called Party E.164 address Address Signal: Address Signal:
(format +CC NDC SN) Analyse the information contained in received Remove "+CC" from the Called
(e.g. as User info in SIP URI Number Portability Routing Number. If the Party E.164 address and use the
with user=phone, or as tel Number Portability Routing number contains remaining digits to fill the Address
URL) plus Number Portability an E.164 address, then remove "+CC" and use signals.
Routing Number (format +CC the remaining digits to fill the Address signal.
NDC SN) (e.g. as Tel URI rn= Otherwise, use the digits of the Number
parameter) plus Number Portability Routing number to fill the Address
Portability Database Dip signal.
Indicator as defined in IETF (NOTE)
RFC 4694 [93] (e.g. as Tel URI Odd/even indicator: set as
Odd/even indicator: set as required
npdi parameter) required
Nature of address indicator: Nature of address indicator:
set Nature of Address indicator to "Network Set Nature of Address indicator to
routing number in national (significant) number "National (significant) number".
format" "National (significant) number" and
"Network routing number in network specific
number format" may alternately be chosen as
described in ITU-T Q.769.1 [92]
Internal Network Number
Internal Network Number Indicator:
Indicator:
1 routing to internal network number not
1 routing to internal network
allowed
number not allowed
Numbering plan Indicator: Numbering plan Indicator:
001 ISDN (Telephony) numbering plan (Rec. 001 ISDN (Telephony) numbering
E.164) plan (Rec. E.164)
NOTE: If PSTN XML and ISUP Sending Terminated (ST) signal are supported as a network option, then the PSTN
XML sendingCompleteIndication, if present, is mapped to the sending terminated digit (hexadecimal digit F)
in the address signals field of the Called Party Number parameter.
3GPP
Release 8 42 3GPP TS 29.163 V8.16.0 (2011-09)
Table 7a.0b: Coding of the called party number with Number Portability Concatenated Addressing
Method
INVITE IAM
Request-URI Called Party Number
Called Party E.164 address Address Signal:
(format +CC NDC SN) Analyse the information contained in received Number Portability
(e.g. as User info in SIP URI with user=phone, Routing Number. If the Number Portability Routing number
or as tel URL) plus Number Portability Routing contains an E.164 address, then remove "+CC" and use the
Number (format +CC NDC SN) (e.g. as Tel URI remaining digits to fill the Address signal. Otherwise, use the digits
rn= parameter) plus Number Portability of the Number Portability Routing number to fill the Address
Database Dip Indicator as defined in IETF RFC signal.
4694 [93] (e.g. as Tel URI npdi parameter)
Remove the "+CC" from the Called Party E.164 address and
append the remaining digits to the Address signal.
Odd/even indicator: set as required
Nature of address indicator:
set Nature of Address indicator to "Network routing number
concatenated with called directory number" or "National
(significant) number" as described in ITU-T Q.769.1 [92]
Internal Network Number Indicator:
1 routing to internal network number not allowed
Numbering plan Indicator:
001 ISDN (Telephony) numbering plan (Rec. E.164)
Table 7a.0c: Coding of the network routing number and called party number with Number Portability
Separate Network Routing Number Addressing Method
INVITE IAM
Request-URI Network Routing Number Called Party Number
Called Party E.164 address Address Signal: Address Signal:
(format +CC NDC SN) Analyse the information contained in received Remove "+CC" from the Called Party
(e.g. as User info in SIP Number Portability Routing Number. If the E.164 address and use the remaining
URI with user=phone, or as Number Portability Routing number contains digits to fill the Address signals.
tel URL) plus Number an E.164 address, then remove "+CC" and use (NOTE)
Portability Routing Number the remaining digits to fill the Address signal.
(format +CC NDC SN) (e.g. Otherwise, use the digits of the Number
as Tel URI rn= parameter) Portability Routing number to fill the Address
plus Number Portability signal.
Database Dip Indicator as Odd/even indicator: set as required Odd/even indicator: set as required
defined in IETF RFC 4694 Nature of address indicator: Nature of address indicator:
[93] (e.g. as Tel URI npdi Set Nature of Address indicator to "Network Set Nature of Address indicator to
parameter) routing number in national (significant) number "National (significant) number".
format".
"Network routing number in network specific
number format" may alternately be chosen as
described in ITU-T Q.769.1 [92]
Internal Network Number Indicator:
1 routing to internal network number
Numbering plan Indicator:
not allowed
001 ISDN (Telephony) numbering plan (Rec.
Numbering plan Indicator:
E.164)
001 ISDN (Telephony) numbering
plan (Rec. E.164)
NOTE: If PSTN XML and ISUP Sending Terminated (ST) signal are supported as a network option, then the PSTN
XML sendingCompleteIndication, if present, is mapped to the sending terminated digit (hexadecimal digit F)
in the address signals field of the Called Party Number parameter.
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Network Operator or National policy may allow forward transfer of number portability status information, as described
in ITU-T Q.769.1 [92]. In this case, the following coding applies.
INVITE IAM
Request-URI Number Portability Forward Information
Called Party E.164 address If the Number Portability Database Dip Indicator is present,
(format +CC NDC SN) and there is no Number Portability Routing Number, set to
(e.g. as User info in SIP URI with user=phone, or as "number portability query done for called number, non-ported
tel URL) plus Number Portability Routing Number called subscriber".
(format +CC NDC SN) (e.g. as Tel URI rn=
parameter) plus Number Portability Database Dip If the Number Portability Database Dip Indicator is present,
Indicator as defined in IETF RFC 4694 [93] (e.g. as and a Number Portability Routing Number is present, set to
Tel URI npdi parameter) "number portability query done for called number, ported
called subscriber".
7.2.3.1.2B.1 Coding of the IAM when a Carrier Identification Code (CIC) is present
The procedures followed in clause 7.2.3.1.2.1 apply with the following addition.
Based on network configuration, if the tel-URI parameter in a tel-URI or the userinfo part of a SIP URI with
user=phone in the Request-URI of an initial INVITE request, contains a "cic=" parameter, as defined in IETF RFC
4694 [93], the I-MGCF may extract the carrier identification code from the "cic=" field for routeing the call. If the
outgoing IAM message contains the Transit Network Selection (TNS) parameter, as defined in ITU-T Q.763 [4], based
on network configuration the TNS may be populated using the carrier identification code from the "cic=" field. The
format of the "cic" parameter (e.g. global-cic and local-cic) shall be compliant to IETF RFC 4694 [93].
7.2.3.1.2B.2 Void
I-MGCF
SDP indicating
preconditions met and/or
active media
COT
If the IAM has already been sent, the Continuity message shall be sent indicating "continuity check successful", when
all of the following conditions have been met:
- The requested remote preconditions (if any) in the IMS network have been met.
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- The media stream previously set to inactive mode is set to active (a s specified in IETF RFC 4566 [56]).
- A possible outstanding continuity check procedure is successfully performed on the outgoing circuit.
7.2.3.1.3A.1 General
The procedures in the present clause are only applicable it the I-MGCF supports the network option of overlap
signalling, using either the in-dialog method or the multiple INVITEs method. Within one IMS only a single of those
methods shall be used.
After the ISUP IAM message has been sent the I-MGCF can receive additional digits. The additional digits may either,
as a network option, be received in in-dialog SIP INFO requests (in-dialog method) as specified in subclause
7.2.3.1.3A.2 or in additional SIP INVITE requests (multiple INVITEs method) as specified in 7.2.3.1.3A.3.
I-MGCF
SIP message
carrying
additional digits
SAM
If interworking of overlap signalling using the in-dialog method is supported by the I-MGCF, the ISUP IAM message
has already been sent, but no ISUP ACM or REL message has been received, and a SIP INFO request carrying
additional digits is received, then the I-MGCF shall generate a SAM and pass it to outgoing BICC/ISUP procedures.
The SAM shall contain in its Subsequent Number parameter only the additional digits received in this SIP INFO
request.
If interworking of overlap signalling using the multiple INVITEs method is supported by the I-MGCF, the ISUP IAM
message has already been sent, but no ISUP ACM or REL message has been received, and the I-MGCF receives an
INVITE with the same Call-ID and From tag as a previous INVITE which was associated with a BICC/ISUP call/bearer
control instance currently existing on the BICC/ISUP side, then:
a) If the number of digits in the Request-URI is greater than the number of digits already received for the
communication, the I-MGCF shall generate a SAM and pass it to outgoing BICC/ISUP procedures. The SAM
shall contain in its Subsequent Number parameter only the additional digits received in this Request-URI
compared with the digits already received for the communication. The I-MGCF shall reply to any earlier
INVITE with a 484 Address Incomplete response if this has not already been done.
b) If the number of digits in the Request-URI is equal or less than the number of digits already received for the
communication, then the I-MGCF shall immediately send a 484 Address Incomplete response for this INVITE.
In this case, no SAM shall be sent to BICC/ISUP procedures.
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I-MGCF
1. INVITE (1)
2. IAM
3. INVITE (2)
4. SAM
5. 484 (1)
6. INVITE (3)
7. SAM
8. 484 (2)
9.ACM
10. 180 (3)
Figure 5b: Receipt of subsequent INVITE for multiple INVITEs method overlap signalling
On sending of a 484 Address Incomplete message for an INVITE transaction, the I-MGCF considers any offer/answer
exchange initiated by the INVITE to be terminated. The new INVITE initiates a new offer/answer exchange. However,
if resources have already been reserved and they can be reused within the new offer/answer exchange, the precondition
signaling shall reflect the current status of the affected preconditions.
7.2.3.1.4.0 General
The I-MGCF shall send the SIP 180 Ringing when receiving any of the following messages:
I-MGCF
ACM
180 Ringing (Subscriber Free)
P-Early-Media (1)
Ring tone
NOTE: Including the P-Early-Media Header is a network option for a speech call.
Figure 6: The receipt of ACM
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I-MGCF
Ring tone
NOTE: Including the P-Early-Media Header is a network option for a speech call.
Figure 7: Receipt of CPG (Alerting)
For a speech call, if the I-MGCF supports the P-Early-Media header as a network option, and if the INVITE request
includes the P-Early-Media header, the I-MGCF shall include in the SIP 180 Ringing response a P-Early-Media header
authorizing early media, except when
- the I-MGCF has already sent a reliable provisional response (see IETF RFC 3262 [54]) including a P-Early-
Media header, as defined in IETF RFC 5009 [89], and
NOTE: If the I-MGCF signals the P-Early-Media header authorizing early media, then the IMS can expect tones
or announcements to the calling party to flow from the CS network via an MGW controlled by the I-
MGCF. In particular, once the I-MGCF sends the 180 Ringing response, ringback is expected in media
from the CS network.
As a network option, an I-MGCF may generate a Call-Info header field, or an Alert-Info header field according to rules
and procedures of IETF RFC 3261 [19] to provide media instead of the in-band media received from the PSTN.
The procedures in the present subclause apply only if the I-MGCF supports the PSTN XML body as a network option.
The I-MGCF shall map the Access Transport Format received in the CPG or ACM into PSTN XML elements as shown
in Table 7a.0f and include this PSTN XML body in the 180 Ringing.
The I-MGCF shall map a possibly available "ProgressIndicator" element in the ATP parameter within the ACM or CPG
into a Progress Indicator in the PSTN XML body of the 180 Ringing . In addition, the I-MGCF shall map the Backward
call indicators parameter and the Optional backward call indicators parameter (if present) within the ACM or CPG into
other ProgressIndicator(s) in the PSTN XML body of the 180 Ringing as shown in table 7a.0g.
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If the I-MGCF supports the PSTN XML body as a network option and the I-MGCF performed fallback as described in
subclause 7.2.3.1.2.5a, in support of a succeeding network not supporting fallback, then the I-MGCF shall create a
PSTN XML body derived as follows, and include it in the 180 Ringing:
- a BearerCapability element is copied from the first BearerCapability element received in the PSTN XML in the
INVITE;
- if two HighLayerCompatibility elements were present in the PSTN XML body in the INVITE, then a
HighLayerCompatibility element is copied from the first HighLayerCompatibility element received in the PSTN
XML in the INVITE; and
- a ProgressIndicator element with "Progress Description" value 5 ("Interworking has occurred and has resulted in
a telecommunication service change"),"Coding Standard" value "00 (ITU-T standardized coding)", and default
value "0011 (Transit Network)" for the "Location" parameter.
If the Transmission Medium Used parameter (TMU) is present and the BC is not available in the ATP in the Address
Complete Message (ACM) or Call Progress Message (CPG), and if no progress indicator No. 1 (call is not end-to-end
ISDN) or No. 2 (destination address is non-ISDN) has been received in the CPG or ACM, the TMU value (Speech or
3.1 kHz audio) shall be mapped to the PSTN XML BearerCapability.
If the Transmission Medium Used parameter (TMU) is present and the BC is available in the ATP in the Address
Complete Message (ACM) or Call Progress Message (CPG), and if no progress indicator No. 1 (call is not end-to-end
ISDN) or No. 2 (destination address is non-ISDN) has been received in the CPG or ACM, the received BC value shall
be included in the PSTN XML BearerCapability.
NOTE: The TMU is only signalled backwards in the event of there being both a TMR and TMR Prime previously
signalled in the IAM and fallback occurring to the bearer capability identified in TMR Prime. If fallback
does not occur, then the absence of the TMU parameter implies that the call is using the bearer capability
associated with the TMR parameter (i.e. "UDITA") – in which case the selected BearerCapability is
included in the ATP parameter and thus the mapping to the PSTN XML BearerCapability is performed as
in table 7a.0f.
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For a speech call upon receipt of one of the following messages, if the I-MGCF supports the P-Early-Media header as a
network option, and if the I-MGCF has received the P-Early-Media header in the INVITE request, and has not already
sent a provisional response including a P-Early-Media header with parameters indicating authorization of early media,
then the I-MGCF shall send the 183 Session Progress response with a P-Early-Media header authorizing early media:
- ACM with the value of the called party’s status indicator "no indication" and one of the options described in
table 7.2.3.1.4A.1. If the I-MGCF supports the PSTN XML body as a network option, the I-MGCF shall map
parameters within the ACM into the PSTN XML body within the 183 as indicated in table 7.2.3.1.4A.1.Based on
local configuration, the I-MGCF may also send a 183 Session Progress response with a P-Early-Media header
authorizing early media if it receives an ACM with other parameters than described in table 7.2.3.1.4A.1.
I-MGCF
ACM
183 Progress
(No indication)
P-Early-Media
appropriate announcement
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Table 7.2.3.1.4A.1: ACM Parameters that trigger the 183 Session Progress response
NOTE: As a network option the I-MGCF can also map ACM into 183 in other cases than those described in table
7.2.3.1.4A.1.
2. Event indicator is set to "Progress" and one of the options described in table 7.2.3.1.4A.2.
I-MGCF
CPG
183 Progress (in-band information available)
P-Early-Media
appropriate announcment
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Table 7.2.3.1.4A.2: CPG Parameters that trigger the 183 Session Progress response
PSTN XML with Progress Indicator with "Progress Optional backward call indicators parameter
Description" value No. 8
("In-band information or appropriate pattern is now In-band information indicator
available") (NOTE 3) 1 "In-band info or an appropriate pattern is now available"
PSTN XML with ProgressIndicator with "Progress Event indicator
Description" value No. 1 000 0010 (progress)
(Call is not end-to-end ISDN: further progress
information may be available in-band") (NOTE 3) Backward call indicators parameter
NOTE: As a network option the I-MGCF can also map CPG into 183 in other cases than those described in table
7.2.3.1.4A.1.
If the I-MGCF supports the PSTN XML body as a network option, the I-MGCF shall map the Access Transport
Parameterreceived in the CPG or ACM into PSTN XML elements as shown in Table 7a.0f and include this XML body
in the 183 Session Progress. The I-MGCF shall include both a ProgressIndicator possibly received in the Access
Transport Parameter and ProgressIndicators derived according to table 7.2.3.1.4A.1 or table 7.2.3.1.4A.2 in the PSTN
XML.
As a network option, an I-MGCF may generate a Call-Info header field, or an Alert-Info header field according to rules
and procedures of IETF RFC 3261 [19] to provide media instead of the in-band media received from the PSTN.
- ACM with call diversion information not indicating that presentation is not allowed and optional backward call
indicators indicate that in-band information is available.
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I-MGCF
181 Call is being
forwarded ACM( call diver-
P-Early-Media (1) sion information)
Audio
NOTE 1: Including the P-Early-Media Header is a network option for a speech call.
Figure 7c: The receipt of ACM (call diversion information)
- CPG with call diversion information not indicating that presentation is not allowed and optional backward call
indicators indicate that in-band information is available.
I-MGCF
181 Call is being
forwarded
P-Early-Media (1) CPG(Call diversion
information)
Audio
NOTE 1: Including the P-Early-Media Header is a network option for a speech call.
Figure 7d: Receipt of CPG (Call diversion information)
For a speech call, if the I-MGCF supports the P-Early-Media header as a network option, and if the INVITE request
includes the P-Early-Media header, the I-MGCF shall include in the SIP 181 Call is being forwarded response a
P-Early-Media header authorizing early media, except when
- the I-MGCF has already sent a reliable provisional response including a P-Early-Media header, as defined in
IETF RFC 5009 [89], or a 180 Ringing response; and
NOTE: If the I-MGCF signals the P-Early-Media header authorizing early media, then the IMS can expect tones
or announcements to the calling party to flow from the CS network via an MGW controlled by the
I-MGCF.
7.2.3.1.4C Sending of 183 Session Progress for overlap signalling using the in-dialog
method
If the I-MGCF supports the network option of overlap signalling using the in-dialog method, and the SIP INVITE
request contained an indication that the 100rel extension is supported or required, the I-MGCF shall send a reliable 183
(Session Progress) response immediately after the reception of an INVITE request.
NOTE: If the INVITE request does not contain an indication that the 100rel extension is supported or required, it
is assumed that overlap is not used, and that no further digits will be received.
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I-MGCF
ANM
200 OK (INVITE)
I-MGCF
If the I-MGCF supports the PSTN XML body as a network option, the I-MGCF shall map the Access Transport
Parameterreceived in the ANM or CON into PSTN XML elements as shown in Table 7.2.3.1.5.1 and include this PSTN
XML body in the 200 OK (INVITE).
On receipt of an ANM/CON message containing the ATP including the Bearer Capability set to "unrestricted digital
information with tones/announcement" without TMU parameter the 200 OK message shall contain the PSTN XML
Bearer Capability "unrestricted digital information with tones/announcement".
If the I-MGCF supports the PSTN XML body as a network option, the I-MGCF shall map an available BCI element in
the ANM or CON into a Progress Indicator in the PSTN XML body as shown in table 7.2.3.1.5.2; the I-MGCF shall
include both a ProgressIndicator possibly received in the Access Transport Parameter and ProgressIndicators derived
according to table 7.2.3.1.5.2 in the PSTN XML.
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Table 7.2.3.1.5.2: Sending criteria of the XML with Progress indicator No (Value of PI)
I-MGCF
BYE
REL
I-MGCF
CANCEL
REL
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Indicators parameter is shown in Table 8a. Table 8 shows the coding of the Cause Value in the REL if it is not available
from the Reason header field. In both cases, the Location Field shall be set to "network beyond interworking point".
Component of SIP
Component value BICC/ISUP Parameter field Value
Reason header field
Protocol "Q.850" Cause Indicators parameter –
protocol-cause "cause = XX" (NOTE 1) Cause Value "XX" (NOTE 1)
– – Location "network beyond
interworking point"
NOTE 1: "XX" is the Cause Value as defined in ITU-T Recommendation Q.850 [38].
Editor’s Note: The mapping of reason headers towards the ISDN may be misused due to possible user creation of
the reason header since there is no screening in IMS.
If the I-MGCF supports this PSTN XML body as a network option and the I-MGCF interworks media encoded in any
of the formats in Table 2a (G.711, Clearmode or t38) without transcoding, and if a PSTN XML body is received in the
BYE or CANCEL request, the I-MGCF shall derive the Access Transport Parameter in the REL message from the
PSTN XML body as shown in Table 8b.
NOTE: According to SIP procedures, in the case that the REL message is received and a final response (e.g. 200
OK (INVITE)) has already been sent (but no ACK request has been received) on the incoming side of the
I-MGCF then the I-MGCF does not send a 487 Request terminated response and instead waits until the
ACK request is received before sending a BYE message.
If the REL message is received and the final response (i.e. 200 OK (INVITE)) has not already been sent, the I-MGCF
shall send a Status-Code 4xx (Client Error), 5xx (Server Error) response or 6xx (Global Failure) response. The Status
code to be sent is determined by examining the Cause value received in the REL message. Table 9 specifies the
mapping of the cause values, as defined in ITU-T Recommendation Q.850 [38], to SIP response status codes. Cause
values not appearing in the table shall have the same mapping as the appropriate class defaults according to ITU-T
Recommendation Q.850 [38].
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A Reason header field containing the received (Q.850) Cause Value of the REL shall be added to the SIP final response
or BYE request sent as a result of this subclause. The mapping of the Cause Indicators parameter to the Reason header
is shown in Table 9a. IETF draft-jesske-dispatch-reason-in-responses [115] describes the use of the Reason header field
in responses. The Reason header field itself is described in IETF RFC 3326 [116].
Table 9a: Mapping of Cause Indicators parameter into SIP Reason header fields
As a network option, an I-MGCF may generate an Error-Info header field according to rules and procedures of
IETF RFC 3261 [19] to provide media instead of the in-band media received from the PSTN.
If the I-MGCF supports the PSTN XML body as a network option, the I-MGCF shall map the Access Transport
Parameter received in the REL into PSTN XML elements as shown in Table 9aa and include this PSTN XML body in
the SIP final response or BYE.
NOTE: For the RSC message, the circuit identified by the CIC is affected.
For the GRS message, the affected circuits are identified by the CIC and the Range subfield of the Range
and Status parameter.
For the CGB message, the affected circuits are identified by the CIC and the Range and Status parameter.
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If an initial address message has been sent for the affected circuit and at least one backward message relating to that call
has been received then:
- If the final response (i.e. 200 OK (INVITE)) has already been sent, the I-MGCF shall send a BYE message.
- If the final response (i.e. 200 OK (INVITE)) has not already been sent, the I-MGCF shall send a SIP response
with Status-Code 480 Temporarily Unavailable.
A Reason header field containing the (Q.850) Cause Value of the REL message generated by the ISUP procedures shall
be added to the SIP message (BYE or 480 response) to be sent by the SIP side of the I-MGCF.
IM CN Subsystem CS Network
MGCF
REFER
To: B
P-Asserted ID: A
Refer to: C
Method: INVITE
403 Forbidden
Upon receipt of a REFER request at the MGCF, the default behaviour of the I-MGCF is to reject the REFER request
with a 403 Forbidden response.
NOTE: The I-MGCF may also decide for example to execute the REFER request as specified in IETF RFC 3515
[75] as an operator option, but such handling is outside of the scope of the present document.
A Reason header field containing the (Q.850) Cause Value of the REL message sent by the I-MGCF shall be added to
the SIP Message (BYE request or final response) sent by the SIP side of the I-MGCF.
Editor's Note: It is FFS whether to indicate the cause value for internal error in the network to the user.
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7.2.3.2.1.1 General
O-MGCF
IAM
INVITE
Upon reception of an IAM message, the O-MGCF shall send a SIP INVITE request, as further detailed in the sub-
clauses below.
An O-MGCF shall support both the SIP preconditions and 100 rel extensions and indicate the support of the SIP
preconditions and 100rel extensions in the INVITE request, unless the Note below applies.
NOTE: If the O-MGCF is deployed in an IMS network that by local configuration serves no user requiring
preconditions, it may send the INVITE request without indicating support of preconditions.
If the Continuity Check indicator in the Nature of Connection Indicators parameter in the incoming IAM is set to
indicate either "continuity check required on this circuit" or "continuity check performed on previous circuit", the O-
MGCF should defer sending the INVITE request until receiving a COT message.
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NOTE: If the Continuity Check indicator in the Nature of Connection Indicators parameter in the incoming IAM
is set to indicate either "continuity check required on this circuit" or "continuity check performed on
previous circuit" and the O-MGCF sends the INVITE request before receiving a COT message, the
following considerations apply: If the receiving terminal is not supporting the SIP precondition and the
SIP UPDATE method, clipping may occur. Furthermore, if the MGCF sets the SDP "inactive" attribute in
the initial INVITE request and the receiving terminal is not supporting the SIP precondition, the
interworking procedures within the present specification do not describe all necessary signalling
interactions required to set up a call, in particular with respect to the sending of the re-INVITE that may
also cause additional delay in the call setup. In addition, the interworking of the ringing indication might
not be possible if the peer sends the ringing indication only as response to a re-INVITE.
O-MGCF
IAM
COT
(NOTE)
INVITE
NOTE: Waiting for the COT is recommended, if the Continuity Check indicator in the Nature of Connection
Indicators parameter in the incoming IAM is set to indicate either "continuity check required on this circuit"
or "continuity check performed on previous circuit"
Figure 12: Receipt of an IAM (Waiting for the COT message)
If no calling party number is received in the incoming IAM message, as a network option, the O-MGCF may send an
INR message to request the calling party number and not send the INVITE request until receiving an INF message with
calling party number. If no calling party number is received in the INF message, O-MGCF may reject or continue the
call based on local configuration.
O-MGCF
IAM
INR
INF
INVITE
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O-MGCF
IAM
SAM
INVITE
After initiating the normal incoming BICC/ISUP call establishment procedures, determining the end of address
signalling and selecting to route the call to the IMS domain, the O-MGCF shall send the initial INVITE.
The end of address signalling shall be determined by the earlier of the following criteria:
b) by receipt of the maximum number of digits used in the national numbering plan; or
c) by analysis of the called party number to indicate that a sufficient number of digits has been received to route the
call to the called party; or
d) by observing that timer Ti/w1 has expired after the receipt of the latest address message and the minimum
number of digits required for routing the call have been received.
If the end of the address signalling is determined in accordance with criteria a) b) or c), the timer Ti/w2 is started when
INVITE is sent. Also, if the PSTN XML body is supported as a network option, the O-MGCF shall insert the PSTN
XML sendingCompleteIndication.
The Fallback mechanism described in the present Clause shall only apply if the MGCF supports the PSTN XML body
as a network option.
When the IAM includes a TMR and a TMR prime parameter and a USI and USI Prime parameter then the O-MGCF
shall:
- Map the USI Prime into the second Bearer Capability stated in the XML Bearer Capability element and
- For the TMR and USI Prime the mapping shall apply according to the procedures described within clause
7.2.3.2.2.2 for the first offered codec.
- Map the USI into the first Bearer Capability stated in the XML Bearer Capability element and
- For the TMR prime and USI the mapping shall apply according to the procedures described within clause
7.2.3.2.2.2 for the second offered codec.
NOTE: If the Fallback mechanism is not supported the ISUP Fallback procedures apply.
In cases where the fallback appears within the terminating entity and sends back a SDP Answer that is equivalent with
the TMR prime codec (fallback codec) then the O-MGCF shall only apply the cut-though to the fallback codec.
In cases where preconditions are used the O-MGCF has to wait for the SDP answer where the preconditions are met and
fallback codec is sent back.
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7.2.3.2.1a.1 General
As a network option, the O-MGCF is not required to determine the end of address signalling. In this case the O-MGCF
shall send an initial INVITE when a preconfigured number of digits has been reached.
When the MGCF receives ISUP SAM messages the additional digits received in the SAMs may either be sent using the
in-dialog overlap method as specified in subclause 7.2.3.2.1a.2 or using the multiple INVITEs overlap method as
specified in 7.2.3.2.1a.3. It depends on the network configuration which of these methods is applied. However, within
one IMS only a single method shall be used.
O-MGCF
IAM
INVITE
404/484
SAM
INVITE
18x
SAM
INFO
200 (INFO)
If the O-MGCF sends an initial SIP INVITE request before the end of address signalling is determined, the O-MGCF
shall:
- use the SIP precondition extension within the SIP INVITE request;
- be prepared to handle incoming SIP 18x provisional responses, establishing early dialogs; and
- be prepared to handle incoming SIP 404 or 484 error responses as detailed in Clause 7.2.3.2.12.1.
NOTE: A SIP INVITE request with incomplete address information can be rejected with a SIP 404 or 484 error
response.
On receipt of a SAM from the BICC/ISUP side, unless the O-MGCF has received a SIP 180 (Ringing) response for the
call, the following O-MGCF procedures apply:
- If no response has been received for the previous INVITE request of the same call, the O-MGCF shall wait for
the response and then apply the procedures in the next bullets to transfer the digits received in the SAM. If
additional SAMs are received while the O-MGCF is waiting for the response for the previous SIP INVITE
request, the digits within shall be combined with the digits of the previous SAMs.
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- If an early dialog has not been established, and a SIP 404 or 484 error response has been received for the last
previous SIP INVITE request for the same call, the O-MGCF shall send a SIP INVITE request complying to the
following:
- The SIP INVITE request shall use the SIP preconditions extension.
- The SIP INVITE request shall include all digits received so far for this call in the Request-URI.
- The SIP INVITE request shall include the same Call-ID and From tag as the previous SIP INVITE request
for the call.
- If an early dialog has been established, and a response has been received for any previously sent SIP INFO
request, the O-MGCF shall send an in-dialog SIP INFO request complying the following:
- The SIP INFO request shall only include the digits received since the previous SIP request with digits was
sent (see Note).
- If no response has been received for the previous SIP INFO request, the O-MGCF shall wait for the response and
then apply the procedures in the previous bullet to transfer digits received in the SAM. If additional SAMs are
received while the O-MGCF is waiting for the response for the previous SIP INFO request, the digits within
shall be combined with the digits of the previous SAMs.
- Restart Ti/w2.
If timer Ti/w2 has expired, or the O-MGCF has received a SIP 180 (Ringing) response for the call, the O-MGCF shall
ignore subsequent SAMs received.
NOTE: The encoding of the digits within the SIP INFO request is described in Clause 7.2.3.2.20.2.
Editor's note: It needs to be verified whether the timer procedures associated with the in-dialog method are correct.
7.2.3.2.1a.3 Additional digits sent using the multiple INVITEs overlap method
O-MGCF
IAM
INVITE
404/484
SAM
INVITE
404/484
SAM INVITE
If the O-MGCF sends an SIP INVITE request before the end of address signalling is determined, the O-MGCF shall:
- use the SIP precondition extension within the SIP INVITE request;
- be prepared to handle incoming SIP 404 or 484 error responses as detailed in Clause 7.2.3.2.12.1.
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NOTE: An SIP INVITE request with incomplete address information will be rejected with a SIP 404 or 484 error
response.
On receipt of a SAM from the BICC/ISUP side, unless the O-MGCF has received a SIP 180 (Ringing) response for the
call, the O-MGCF shall:
- The SIP INVITE request shall use the SIP preconditions extension.
- The SIP INVITE request shall include all digits received so far for this call in the Request-URI.
- The SIP INVITE request shall include the same Call-ID and From tag as the previous SIP INVITE request
for the call.
- restart Ti/w2;
If timer Ti/w2 has expired, or the O-MGCF has received a SIP 180 (Ringing) response for the call, the O-MGCF shall
ignore subsequent SAMs received.
7.2.3.2.2.0 Overview
Table 10aa provides a summary of how the header fields within the outgoing INVITE message are populated.
IAM INVITE
Called Party Number Request-URI
To
Calling Party Number P-Asserted-Identity
Privacy
From
Generic Number ("additional calling party number") From
Hop Counter Max-Forwards
TMR/USI Message Body (application/SDP)
The called party number parameter of the IAM message is used to derive Request URI of the INVITE Request. The
Request URI is a tel URI or SIP URI with "user=phone" and shall contain:
- an E.164 International public telecommunication number prefixed by a "+" sign (e.g. tel: +4911231234567), or
- a non E.164 number (national operator option for service numbers), expressed as a local number as per IETF
RFC 3966 [97].
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IAM INVITE
Called Party Number/ Nature of address Request-URI and To header field
indicator
National (significant) number Insert "+CC" before the Address signals (NOTE 1)
International number Insert "+" before the Address signals
Network-specific number or according to local policies should either be:
reserved for national use - a global number (+CC), if the called party number may be converted
into an E.164 address
If the O-MGCF indicates support of the SIP preconditions in the initial INVITE request and local preconditions have
not been met, the SDP media description shall contain precondition information as per 3GPP TS 24.229 [9]. Depending
on the coding of the continuity indicators different precondition information (IETF RFC 3312 [37]) is included. If the
continuity indicator indicates "continuity performed on a previous circuit" or "continuity required on this circuit", and
the INVITE is sent before receiving a COT message (which is not recommended according to subclause 7.2.3.2.1.1),
then the O-MGCF shall indicate that the preconditions are not met. Otherwise the O-MGCF shall indicate whether the
preconditions are met, dependent on the status of the local resource reservation. If the local preconditions are not met
the O-MGCF should set the media stream to inactive mode (by including an attribute "a=inactive"). If the local
configuration indicates that O-MGCF is deployed in the IMS network that serves users supporting SIP precondition
mechanism, the attribute "a=inactive" may be omitted when the initial SDP offer indicates local preconditions are not
met. If the initial SDP offer indicates local preconditions are fulfilled, the O-MGCF shall not set the media stream to
inactive mode.
If the O-MGCF determines that a speech call is incoming, the O-MGCF shall include the AMR codec transported
according to IETF RFC 3267 [23] with the options listed in clause 5.1.1 of 3GPP TS 26.236 [32] in the SDP offer,
unless the Note below applies. Within the SDP offer, the O-MGCF should also provide SDP RR and RS bandwidth
modifiers specified in IETF RFC 3556 [59] to disable RTCP, as detailed in Clause 7.4 of 3GPP TS 26.236 [32]. The O-
MGCF may include other codecs according to operator policy.
NOTE: If the O-MGCF is deployed in an IMS network that by local configuration serves no user equipment that
implements the AMR codec, then the AMR codec may be excluded from the SDP offer.
To avoid transcoding or to support non-speech services, the O-MGCF may add media derived from the incoming ISUP
information according to Table 10b. The support of the media listed in Table 10b is optional. If the O-MGCF supports
the PSTN XML body as a network option and adds media derived from the incoming ISUP information according to
Table 10b, the O-MGCF shall also map the media related ISUP information into the XML body as shown in Table
7.2.3.2.2.7.1.
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Table 10b: Coding of SDP media description lines from TMR/USI: ISUP to SIP
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Has a Calling Party Calling Has a Generic Generic P-Asserted-Identity From header field Privacy header field
Number parameter with Party Number (ACgPN) Number header field
complete E.164 Number with a complete APRI
number, and with APRI E.164 number and
Screening Indicator = with Screening
UPVP or NP (NOTE 1) Indicator = UPNV
been received? been received?
N - N - Header field not included SIP or SIPS URI with addr-spec Header field not included
of Unavailable User Identity
(NOTE 2) (NOTE 6)
N - Y (NOTE 3) "presentation Header field not included addr-spec derived from Generic Header field not included
allowed" Number (ACgPN) address
signals if available
or network provided value
(NOTE 6)
N - Y (NOTE 3) "presentation Header field not included SIP or SIPS URI with addr-spec Header field not included
restricted" of Unavailable User Identity
(NOTE 2) (NOTE 6)
Y "presentation N - Derived from Calling Party Tel URI or SIP URI derived from Privacy header is not
allowed" Number parameter Calling Party Number parameter included or if included, "id"
address signals address signals (See table 15) is not included (See table
(See table 14) 16)
Y "presentation Y "presentation Derived from Calling Party Derived from Generic Number Privacy header is not
allowed" allowed" Number parameter (ACgPN) address signals included or if included, "id"
address signals (See table 13) is not included
(See table 14) (NOTE 6) (See table 16 )
Y "presentation Y "presentation Derived from Calling Party Tel URI or SIP URI derived from Privacy header is not
allowed" restricted" Number parameter Calling Party Number parameter included or if included, "id"
address signals address signals (See table 15) is not included
(See table 14) (NOTE 9) (See table 16 )
Y "presentation N - Derived from Calling Party SIP or SIPS URI with addr-spec priv-value =: "id". (See
restricted" Number parameter of Anonymous URI (NOTE 7) table 16)
address signals (NOTE 6)
(See table 14)
Y "presentation Y "presentation Derived from Calling Party Derived from Generic Number priv-value =: "id".
restricted" allowed" Number parameter (ACgPN) address signals
address signals (See table 13)
(See table 14) (NOTE 6)
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Has a Calling Party Calling Has a Generic Generic P-Asserted-Identity From header field Privacy header field
Number parameter with Party Number (ACgPN) Number header field
complete E.164 Number with a complete APRI
number, and with APRI E.164 number and
Screening Indicator = with Screening
UPVP or NP (NOTE 1) Indicator = UPNV
been received? been received?
Y "presentation Y "presentation Derived from Calling Party SIP or SIPS URI with addr-spec priv-value =: "id" (NOTE 8)
restricted" restricted" Number parameter of Anonymous URI (NOTE 7)
address signals (NOTE 6) (NOTE 8)
(See table 14)
Y "presentation N - Header field not included. addr-spec is set to Privacy header is not
restricted by "unavailable@hostportion" included or if included, "id"
network" (NOTE 5) is not included (See table
(NOTE 4) 16)
Y "presentation Y "presentation Header field not included. Derived from Generic Number Privacy header is not
restricted by allowed" (ACgPN) address signals included or if included, "id"
network" (See table 13) is not included
(NOTE 6) (See table 16 )
Y "presentation Y "presentation Header field not included. addr-spec is set to Privacy header is not
restricted by restricted" "unavailable@hostportion" included or if included, "id"
network" (NOTE 5) is not included
(See table 16 )
NOTE 1: A Network Provided CLI in the CgPN parameter may occur on a call to IMS. Therefore in order to allow the "display" of this Network Provided CLI at a SIP UAS it shall
be mapped into the SIP From header. It is also considered suitable to map into the P-Asserted-Identity header since in this context it is a fully authenticated CLI
related exclusively to the calling line, and therefore as valid as a User Provided Verified and Passed CLI for this purpose.
NOTE 2: The "From" header may contain an "Unavailable User Identity". An "Unavailable User Identity" includes information that does not point to the calling party and
indicates that the caller's identity is unknown. The encoding of the "Unavailable User Identity" shall be as defined in 3GPP TS 23.003 [74].
NOTE 3: This combination of CgPN and ACgPN is an error case but is shown here to ensure consistent mapping across different implementations.
NOTE 4: This is an ETSI specific value described within ETSI EN 300 356-1 [70].
NOTE 5: The setting of the hostportion is according to operator policy.
NOTE 6: In accordance with IETF RFC 3261 [19] procedures, a tag shall be added to the "From" header.
NOTE 7: The "From" header may contain an "Anonymous User Identity". An "Anonymous User Identity" includes information that does not point to the calling party and
indicates that the caller has withheld their identity. The encoding of the "Anonymous User Identity" shall be as defined in 3GPP TS 23.003 [74].
NOTE 8: As a network option, the "From" header may be derived from the Generic Number parameter address signals (see table 13) and in the Privacy header the priv-value
set to „id‟+ „header‟ + „user‟. This option is only recommended to use within a trusted domain where an entity such a TAS is configured to be inserted into the call path
that is able to change the "From" Header content to an anonymous user identity (NOTE 7).
NOTE 9: As a network option, the "From" header may be derived from the Calling Party Number parameter address signals (see table 15). In this case privacy header is not
included.
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Table 13: Mapping of generic number (additional calling party number) to SIP from header fields
Table 14: Mapping of calling party number parameter to SIP P-Asserted-Identity header fields
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Table 15: Mapping of BICC/ISUP Calling Party Number parameter to SIP From header fields
Table 16: Mapping of BICC/ISUP APRIs into SIP privacy header fields
See Annex C for normative interworking of a Calling party's category to a "cpc" URI parameter within P-Asserted-
Identity header field.
See Annex H for normative interworking of the "oli" URI parameter as a network option.
If the Hop Counter procedure is supported in the CS network, the O-MGCF shall use the Hop Counter parameter to
derive the Max-Forwards SIP header. Due to the different default values (that are based on network
demands/provisions) of the SIP Max-Forwards header and the Hop Counter, an adaptation mechanism shall be used to
adopt the Hop Counter to the Max Forwards at the O-MGCF. For example, the following guidelines could be applied.
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a) Max-Forwards for a given message should be monotone decreasing with each successive visit to a SIP entity,
regardless of intervening interworking, and similarly for Hop Counter.
b) The initial and successively mapped values of Max-Forwards should be large enough to accommodate the
maximum number of hops that may be expected of a validly routed call.
The factor used to map from Hop Counter to Max-Forwards for a given call will depend on call origin, and will be
provisioned at the O-MGCF based on network topology, trust domain rules, and bilateral agreement.
The Principle of adaptation could be implemented on a basis of the network provision, trust domain rules and bilateral
agreement.
For speech and video calls, the O-MGCF shall insert an IMS Communication Service Identifier, indicating the IMS
Multimedia Telephony Communication Service.
The IMS Communication Service Identifier for the IMS Multimedia Telephony Communication Service is defined in
3GPP TS 24.173 [88].
For a speech call, if the O-MGCF supports the P-Early-Media header as a network option, then it shall include the
header in each outgoing INVITE request.
If the O-MGCF supports the PSTN XML body as a network option, the O-MGCF shall map ISUP information into the
XML body as shown in Table 7.2.3.2.2.7.1 and 7.2.3.2.2.8.1.
IAM INVITE
ISUP Parameter Content PSTN XML
Access Transport Parameter High layer compatibility HighLayerCompatibility (NOTE 1,
NOTE 2, NOTE 3)
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If the O-MGCF supports the PSTN XML body as a network option, the Forward call indicators parameter and an
available "ProgressIndicator" element in the IAM shall be mapped into a Progress Indicator in the PSTN XML body of
the INVITE as shown in table 7.2.3.2.2.8.1.
IAM INVITE
Forward call indicators parameter Access transport PSTN XML body with
parameter Progress indicator with
ISDN User Part indicator ISDN access indicator "Progress Description"
value No. (Value of PI)
0
("ISDN User Part Value non-significant Value non-significant No. 1 (NOTE 1)
not used all the way")
1 0
("ISDN User Part ("originating access non - ISDN") Value non-significant No. 3 (NOTE 1)
used all the way")
1 1 Progress Indicator Progress Indicator
("ISDN User Part ("originating access ISDN") No. (Value of PI) received in the ATP
used all the way") (NOTE 2) and additional
Progress Indicator with
"Progress Description"
value No. 6 (NOTE 1)
1
1
("ISDN User Part Not present No. 6 (NOTE 1)
("originating access ISDN")
used all the way")
NOTE 1: The Progress Indicator "Coding Standard" parameter shall be set to "00 (ITU-T standardized coding)". The
default value for the Progress Indicator "Location" parameter is "0011 (Transit Network)".
NOTE 2: The entire Progress Indicator, including the "Progress Description", "Coding Standard" and "Location"
parameters shall be copied.
When Number Portability is supported, the method used for signalling of the Called Party E.164 address and the
Number Portability Routing Number determines the parameters of the IAM message used to derive the Request URI
and To header of the INVITE Request.
ITU-T Q.769.1 [92] describes three possible addressing methods for signalling of the Called Party E.164 address and
Number Portability Routing Number (ITU-T Q.769.1 [92] uses the terms directory number and network routing number
respectively). The choice of these methods is based on network operator and national requirements.
The following sub-clauses describe how the Request URI and To header fields are populated, based on these methods,
when a Number Portability Routing Number is available in the IAM.
When the optional Number Portability Routing Number is available and supported, these procedures take precedence
over procedures for coding of the Request URI and To header fields described in clause 7.2.3.2.2.1.
When a Number Portability Routing Number is not available, the Request URI and To Header fields are populated as
described in clause 7.2.3.2.2.1, with the following addition: If a Number Portability Forward Information Parameter
parameter is present in the IAM, containing a value of "number portability query done for called number, non-ported
called subscriber", a tel URI npdi parameter [93] is added.
For the following sub-clauses, the Request URI is a tel URI or SIP URI with "user=phone" and shall contain an
International public telecommunication number prefixed by a "+" sign (e.g. tel: +4911231234567).
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Table 17a: Mapping ISUP to SIP Request-URI and To header field with Number Portability Separate
Directory Number Addressing Method
IAM INVITE
Called Party Number Called Directory Number Request-URI and To Header Field
Address Signal: Address Signal: The ‘telephone-subscriber’ is populated from
Nature of address indicator: Nature of address indicator: the Called Directory Number as follows:
"Network routing number in "National (significant) number". Insert "+CC" before the Address signals (NOTE 1)
national (significant) number
format" or "National The Tel URI rn= parameter is populated from
(significant) number" or the Called Party Number as follows:
"Network routing number in Insert "+CC" before the Address signals (NOTE 1)
network specific number
format" as described in ITU-T Use of the local form of the rn= parameter is out of
Q.769.1 [92] the scope of the present specification.
(NOTE 2)
Tel URI npdi parameter as defined in IETF RFC
4694 [93] is added.
NOTE 1: CC = Country Code of the network in which the O-IWU is located.
NOTE 2: If the address signals received in the ISUP Called Party Number contain a sending terminated signal
(hexadecimal digit F), then this shall be discarded.
Table 17b: Mapping ISUP to SIP Request-URI and To header field with Number Portability
Concatenated Number Addressing Method
IAM INVITE
Called Party Number Request-URI and To Header Field
Address Signal: The ‘telephone-subscriber’ is populated from the Called Party
Nature of address indicator: Number as follows:
"Network routing number concatenated Remove the prefix representing the Number Portability Routing Number
with called directory number" or "National or the prefix prior to the directory number (NOTE3).
(significant) number" as described in ITU-
T Q.769.1 [92] Insert "+CC" before the Address signals (NOTE 1).
(NOTE 2)
The Tel URI rn= parameter is populated from the Called Party
Number as follows:
Use all address digits contained within the Called Party Number or
remove the digits that follow the prefix representing the Number
Portability Routing Number
Use of the local form of the rn= parameter is out of the scope of the
present specification.
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Table 17c: Mapping ISUP to SIP Request-URI and To header field with Number Portability Separate
Network Routing Number Addressing Method
IAM INVITE
Network Routing Number Called Party Number Request-URI and To Header Field
Address Signal: Address Signal: The "telephone-subscriber" is
Nature of address indicator: Nature of address indicator: populated from the Called Party Number
"Network routing number in national "National (significant) number". as follows:
(significant) number format" or Insert "+CC" before the Address signals
"Network routing number in network (NOTE 1)
specific number format" as
described in ITU-T Q.769.1 [92] The Tel URI rn= parameter is populated
(NOTE 2) from the Network Routing Number as
follows:
Insert "+CC" before the Address signals
(NOTE 1)
The procedures followed in clause 7.2.3.2.2.1 apply with the following addition.
If the Transit Network Selection parameter, defined according to ITU-T Q.761 [4], is included in the IAM message the
O-MGCF, based on network configuration, may send the transit network selection information to the SIP network. In
such a case the "cic=" parameter as defined in IETF RFC 4694 [93] is included in the SIP-Request URI and configured
according to the table below.
Table 17d: Mapping of ISUP "Transit Network Selection" (TNS) to SIP "Carrier Identification Code"
(CIC)
7.2.3.2.2B.2 Void
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O-MGCF
COT(success)
‚9 SDP indicating
pre-conditions
met
When the requested local preconditions (if any) have been met and if possible outstanding continuity procedures have
successfully been completed (COT with the Continuity Indicators parameter set to "continuity check successful" is
received), a SDP offer (e.g. a SIP UPDATE request, as defined in IETF RFC 3311 [55]) shall be sent for each early SIP
dialogue for which the received provisional response indicated support of preconditions confirming that all the required
local preconditions have been met. If the O-MGCF previously offered inactive media stream it shall set the media
stream to active mode.
NOTE: This procedure applies regardless of whether the early SIP dialog existed prior to the preconditions being
fulfilled or is subsequently created.
- the detection of end of address signalling by the expiry of Timer T i/w1 or,
O-MGCF
IAM
SAM T i/w1 running
- the reception of the first 180 Ringing. An O-MGCF supporting the P-Early-Media header as a network option
should initiate the sending of an awaiting answer indication only if according to IETF RFC 5009 [89] backward
early media is not authorized (the most recently received P-Early-Media header is received does not authorize
the backward early media or the P-Early-Media header has not yet been received).
O-MGCF
180 Ringing
ACM (Subscriber Free)
Ring tone
Figure 16: Sending of ACM (Receipt of first 180 Ringing and backward early media is not authorized)
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Based on local knowledge that the call is transited to a PSTN network, the O-MGCF may decide not to generate
the awaiting answer indication when receiving the 180 Ringing message and backward early media is not
authorized according to IETF RFC 5009 [89].
O-MGCF
180 Ringing
[P-Early-Media header:
ACM backward early media
(Subscriber Free) authorized]
Ring tone
Figure 16a: Sending of ACM (Receipt of first 180 Ringing and backward early media is authorized)
O-MGCF
Figure 16b: Sending of ACM (Receipt of first 181 Call is Being Forwarded and backward early media
is not authorized)
O-MGCF
181 Call is Being
Forwarded
ACM [P-Early-Media
(Call Diversion header: backward early
information, OBCI: in- media authorized]
band info available)
Audio
Figure 16c: Sending of ACM (Receipt of first 181 Call is Being Forwarded that includes authorization
of early media)
- At an O-MGCF supporting the P-Early-Media header as a network option, once all the following sub-conditions
have been met: {1} the reception of the first 183 Session Progress that includes a P-Early-Media header
authorizing backward early media, and {2} SDP preconditions are not used, or applicable SDP preconditions
have been met.
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O-MGCF
Figure 16d: Sending of ACM (Receipt of first 183 Session Progress that includes authorization of
early media)
O-MGCF
IAM INVITE
When a 180 (Ringing) response is received with the Alert-Info header field at an O-MGCF supporting capabilities
associated with the Alert-Info header field an O-MGCF may instruct the IM-MGW to play out early media available at
the associated URL to the PSTN leg of the communication.
At an O-MGCF supporting the P-Early-Media header as a network option, if the O-MGCF receives a 18x response with
a P-Early-Media header that changes the authorization of early media, the O-MGCF terminates the sending of the
awaiting answer indication if the header authorizes backward early media, and initiates the sending of the awaiting
answer indication if the header removes authorization of backward early media and if the O-MGCF has received the
180 Ringing response.
10 charge
00 no indication otherwise
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00 no indication
1 interworking encountered
As a network operator option, the value I = 0 "no interworking encountered" is used for TMR = 64 kBit/s unrestricted
NOTE: This avoids sending of a progress indicator with Progress information 0 0 0 0 0 0 1 "Call is not end-to-end
ISDN; further call progress information may be available in-band", so the call will not be released for that
reason by an ISDN terminal.
As a network operator option, the value K = 1 "ISDN user part/BICC used all the way" is used for TMR = 64 kBit/s
unrestricted.
NOTE: This avoids sending of a progress indicator with progress information 0 0 0 0 0 0 1 "Call is not end-to-end
ISDN; further call progress information may be available in-band", so the call will not be released for that
reason by an ISDN terminal.
As a network operator option, the value M = 1 "terminating access ISDN" is used for TMR = 64 kBit/s unrestricted.
NOTE: This avoids sending of a progress indicator with progress information 0 0 0 0 0 1 0 "Destination access is
non-ISDN", so the call will not be released for that reason by an ISDN terminal.
1 incoming echo control device included, for speech calls, e.g., TMR is "3.1KHz audio".
0 incoming echo control device not included, for known data calls, e.g., TMR "64 kBit/s
unrestricted" or HLC "Facsimile Group 2/3".
If the PSTN XML body is supported as a network option, the Backward Call indicators parameters derived as shown in
Table 7.2.3.2.5.1.1 shall take precedence over the above Backward Call indicators parameter setting.
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Table 7.2.3.2.5.1.1: Derivation of Backward Call Indicators from PSTN XML body
Bit A 1 "in-band information or an appropriate pattern is now available" shall be set if 183 Session Progress or
181 Call is Being Forwarded response is received and according to IETF RFC 5009 [89] backward early
media is authorized.
If the O-MGCF supports the PSTN XML body as a network option and if a PSTN XML body is received within the 180
ringing or 183 session progress, the O-MGCF shall store the received PSTN XML elements, replacing any previously
stored PSTN XML elements on that dialog.
NOTE: Multiple 18x responses can be received, both within a single dialog and in multiple dialogs. The PSTN
XML bodies are stored on a per-dialog basis to be mapped to the ATP/TMU parameters on receipt of the
200 OK (see subclause 7.2.3.2.9.2).
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If the O-MGCF supports the PSTN XML body as a network option and receives it in the 180 or 183, the O-MGCF shall
store a "ProgressIndicator" element from the PSTN XML body on a per dialog basis and shall add itionaly map it into a
Progress Indicator in the ACM as shown in table 7.2.3.2.5.4.1.
ACM 180/183
Access transport parameter PSTN XML body with Progress indicator with "Coding
Standard" value "00 (ITU-T standardized coding)" and with
"Progress Description" No. (Value of PI)
Progress indicator (NOTE) Progress indicator No. 1 / 2
Progress indicator (NOTE) Progress indicator No. 8
NOTE: The entire Progress Indicator, including the "Progress Description", "Coding Standard" and "Location"
parameters shall be copied.
7.2.3.2.6.0 General
If the Address Complete Message (ACM) has already been sent, the O-MGCF shall send the Call Progress message
(CPG) in the following cases:
- Upon receipt of the SIP 180 Ringing provisional response. An O-MGCF supporting the P-Early-Media header as
a network option should initiate the sending of an awaiting answer indication only if according to IETF RFC
5009 [89] backward early media is not authorized (the most recently received P-Early-Media header does not
authorize the backward early media or the P-Early-Media header has not yet been received).
O-MGCF
180 Ringing
CPG (Alerting)
Ring tone
Figure 18: Sending of CPG (Alerting) (Receipt of 180 Ringing response and backward early media is
not authorized)
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O-MGCF
180 Ringing
[P-Early-Media header:
backward early media
CPG authorized]
(Alerting)
Ring tone
Figure 18a: Sending of CPG (Alerting) (Receipt of 180 Ringing response with authorization of early
media)
Based on local knowledge that the call is transited to a PSTN network, the O-MGCF may decide not to generate
the awaiting answer indication when receiving the 180 Ringing message and backward early media is not
authorized according to IETF RFC 5009 [89].
- At an O-MGCF supporting the P-Early-Media header as a network option, upon receipt of a 183 Session
Progress that includes the first P-Early-Media header authorizing backward early media.
O-MGCF
Appropriate
announcement
O-MGCF
Audio
At an O-MGCF supporting the P-Early-Media header as a network option, if the O-MGCF receives a 18x response with
P-Early-Media header that changes the authorization of early media, the O-MGCF terminates the sending of the
awaiting answer indication if the header authorizes backward early media and initiates the sending of the awaiting
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answer indication if the header removes authorization of backward early media and if the O-MGCF has received the
180 Ringing response.
If the O-MGCF supports the PSTN XML body as a network option and receives it in the 180 or 183, any
"ProgressIndicator" element in the PSTN XML body shall be stored on a per-dialog basis as well as mapped as shown
in tables 7.2.3.2.6.1.1 and 7.2.3.2.6.1.3.
Table 7.2.3.2.6.1.1: Mapping of progress indicator in PSTN XML body into ATP
CPG 180/183
Access transport parameter PSTN XML body with Progress indicator X
Progress indicator (NOTE 1, NOTE 3) Progress indicator No. 1 / 2
Progress indicator (NOTE 2, NOTE 3) Progress indicator No. 4
Progress indicator No. 4 (NOTE 2, NOTE 4) Progress indicator No. 7
NOTE 1: Values 1 ("call is not end-to-end ISDN: further call progress information may be available in-band") or 2
("destination address is non-ISDN") shall be sent if Value 4 ("Call has returned to the ISDN") has been sent
since value 1 or 2 was previously sent or if no value 1 or 2 was previously sent.
NOTE 2: Value 4 ("Call has returned to the ISDN") shall be sent if value 1 ("call is not end-to-end ISDN: further call
progress information may be available in-band") or 2 ("destination address is non-ISDN") was sent previously
and no value 4 has been signalled since.
NOTE 3: The entire Progress Indicator, including the "Progress Description", "Coding Standard" and "Location"
parameters shall be copied.
NOTE 4: The Progress Indicator "Coding Standard" parameter shall be set to "00 (ITU-T standardized coding)". The
default value for the Progress Indicator "Location" parameter is "0100 (Public Network serving remote user)".
Table 7.2.3.2.6.1.3: Mapping of progress indicator in PSTN XML body into Event Indicator
CPG 180/183
Event indicator PSTN XML body with Progress indicator with "Coding
Standard" value "00 (ITU-T standardized coding)" and
with Progress Description" value No. X
"In-band information or appropriate pattern is now No. 8
available" "In-band information or appropriate pattern is now available"
7.2.3.2.7.0 General
The description of the following ISDN user part parameters can be found in ITU-T Recommendation Q.763 [4].
0 0 0 0 0 1 1 in-band information or an appropriate pattern is now available, if the received 183 Session
Progress response and most recently received P-Early-Media header authorize backward early
media
NOTE: In national networks other values of the Event indicator may be used.
If the O-MGCF supports the PSTN XML body as a network option and receives in the 180 or 183 a "ProgressIndicator"
element in the PSTN XML body with a "Coding Standard" value "00 (ITU-T standardized coding)" and with Progress
Description" value No. 8, instead of the mapping above, the O-MGCF shall map this "Progress Indicator" into the
"Event Indicator" within the CPG as shown in table 7.2.3.2.6.1.3.
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If the O-MGCF supports the PSTN XML body as a network option and if a PSTN XML body is received within the 180
ringing or 183 session progress, the O-MGCF shall store the contained information as described in subclause
7.2.3.2.5.3, and additionally shall map this "Progress Indicator" into the ATP within the CPG as shown in table
7.2.3.2.6.1.1.
NOTE: Multiple 18x responses can be received, both within a single dialog and in multiple dialogs. The PSTN
XML bodies are stored on a per-dialog basis to be mapped to the ATP parameter on receipt of the 200
OK (see subclause 7.2.3.2.9.2).
7.2.3.2.7.3 Void
The Backward Call indicator shall be derived as shown in section 7.2.3.2.5.1. The Backward Call Indicators parameter
is optional in the CPG message and shall only be included if any indicators have changed from those previously sent.
Bit A 1 "in-band information or an appropriate pattern is now available" shall be set if 181 Call is Being
Forwarded response is received and according to IETF RFC 5009 [89] the backward early media is
authorized.
The O-MGCF shall not progress any further early dialogues to established dialogues. Therefore, upon the reception of a
subsequent final 200 (OK) response for any further dialogue for an INVITE request (e.g., due to forking), the O-MGCF
shall:
NOTE: Through connection and the stop of awaiting answer indication are described in clause 9.2.3.3
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O-MGCF
200 OK (INVITE)
ANM
If Backwards Call Indicators are included in the ANM, then the coding of these parameters shall be as described in
clause 7.2.3.2.5.1. The Backward Call Indicators parameter is optional in the ANM message and shall only be included
if any indicators have changed from those previously sent.
If the O-MGCF supports the PSTN XML body as a network option and if a PSTN XML body is received within the 200
OK(INVITE) or has been previously stored from a 18x message, the O-MGCF shall map the most recently received
information for the established dialog (i.e. the dialog for which the first 200 OK has been received) into the ANM as
shown in Table 7.2.3.2.9.2.1 except Progress Indicator No 3. or No. 8.
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NOTE: The TMU is only included if both the TMR and TMR Prime were received in the ISUP IAM and fallback
has occurred.
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O-MGCF
200 OK (INVITE)
CON
The Called Party's status indicator (Bit DC) of BCI parameter is set to "no indication". The other BCI indicators shall be
set as described in clause 7.2.3.2.5.1
If the O-MGCF supports the PSTN XML body as a network option and if a PSTN XML body is received within the 200
OK(INVITE), the O-MGCF shall map the contained information into the CON as shown in Table 7.2.3.2.9.2.1 for the
ANM.
O-MGCF
Status Code
REL
4xx, 5xx or 6xx
If a Reason header as described in IETF draft-jesske-dispatch-reason-in-responses [115] is included in a 4xx, 5xx, 6xx
response, then the Cause Value of the Reason header shall be mapped to the ISUP Cause Value field in the ISUP REL
message. The Reason header field itself is described in IETF RFC 3326 [116]. The mapping of the Reason header to the
Cause Indicators parameter is shown in Table 8a (see subclause 7.2.3.1.7). Otherwise coding of the Cause value field in
the REL message is derived from the SIP Status code received according to Table 18. The Cause Indicators Parameter
Values are defined in ITU-T Recommendation Q.850 [38].
In all cases where SIP itself specifies additional SIP side behaviour related to the receipt of a particular INVITE
response these procedures should be followed in preference to the immediate sending of a REL message to BICC/ISUP.
If there are no SIP side procedures associated with this response, the REL shall be sent immediately.
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NOTE Depending upon the SIP side procedures applied at the O-MGCF it is possible that receipt of certain
4xx/5xx/6xx responses to an INVITE may in some cases not result in any REL message being sent to the
BICC/ISUP network. For example, if a 401 Unauthorized response is received and the O-MGCF
successfully initiates a new INVITE containing the correct credentials, the call will proceed.
If the O-MGCF supports the PSTN XML body as a network option and if a PSTN XML body is received within the
4xx/5xx/6xx, the O-MGCF shall map the contained information into the Access Transport Parameter of the REL as
shown in subclause 7.2.3.2.9.2.
When a 4xx, 5xx or 6xx SIP response to an INVITE request is received from the network containing an Error-Info
header field, the O-MGCF, supporting the capabilities associated with the Error-Info header field, may instruct the IM-
MGW to play out media available at the associated URL towards PSTN.
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7.2.3.2.12.1 Special handling of 404 Not Found and 484 Address Incomplete responses after
sending of INVITE without determining the end of address signalling
This Clause is only applicable when the network option of Sending of INVITE without determining the end of address
signalling is being used (see subclause 7.2.3.2.1.a).
On receipt of a 404 Not Found or 484 Address Incomplete response while Ti/w2 is running, the O-MGCF shall start
timer Ti/w3, if there are no other pending INVITE transactions for the corresponding call.
At the receipt of a SAM, a SIP 1xx provisional responses, or a SIP 200 OK (INVITE), the O-MGCF shall stop Ti/w2
and Ti/w3.
The O-MGCF shall send a REL message with Cause Value 28 towards the BICC/ISUP network if Ti/w3 expires.
O-MGCF
BYE
REL
If a Reason header field with Q.850 Cause Value is included in the BYE request, then the Cause Value shall be mapped
to the ISUP Cause Value field in the ISUP REL as shown in Table 8a (see subclause 7.2.3.1.7). Otherwise, the O-
MGCF sends a REL message with Cause Code value 16 (Normal Call Clearing).
If the O-MGCF supports the PSTN XML body as a network option and if a PSTN XML body is received within the
BYE, the O-MGCF shall map the contained information into the Access Transport Parameter of the REL as shown in
subclause 7.2.3.2.9.2.
A Reason header field containing the received (Q.850) Cause Value of the REL message shall be added to the
CANCEL or BYE request. The mapping of the Cause Indicators parameter to the Reason header is shown in Table 9a
(see subclause 7.2.3.1.8).
If the O-MGCF supports the PSTN XML body as a network option, the O-MGCF shall map the contained information
into a PSTN XML body within the BYE or CANCEL as shown in Table 9aa.
NOTE: For the RSC message, the circuit identified by the CIC is affected.
For the GRS message, the affected circuits are identified by the CIC and the Range subfield of the Range
and Status parameter.
For the CGB message, the affected circuits are identified by the CIC and the Range and Status parameter.
If a final response (i.e. 200 OK (INVITE) has already been received the O-MGCF shall send a BYE method. If a final
response (i.e. 200 OK (INVITE)) has not already been received the O-MGCF shall send a CANCEL method.
A Reason header field containing the (Q.850) Cause Value of the REL message generated by the ISUP procedures shall
be added to the SIP message (BYE or CANCEL request) to be sent by the SIP side of the O-MGCF.
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Editor's Note: It is FFS whether to indicate the cause value for internal error in the network to the user.
NOTE: The MGCF shall send the ACK method before it sends the BYE, if 200 OK (INVITE) is received.
A Reason header field containing the (Q.850) Cause Value of the REL message sent by the O-IWU shall be added to
the SIP Message (BYE or CANCEL request) to be sent by the SIP side of the O-IWU.
Editor's Note: It is FFS whether to indicate the cause value for internal error in the network to the user.
Release with cause code as indicated in table 17 is sent immediately to the BICC/ISUP network.
A BYE request is sent for the early dialog within which the UPDATE was sent.
If all the early dialogs that were generated from the INVITE request have answered the respective UPDATE request
with 580 Precondition failure response then the O-MGCF shall send the Release message with Cause Code '127
Interworking' to the ISUP network.
O-MGCF
COT (failure)
CANCEL
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CANCEL shall be sent if the Continuity message is received with the Continuity Indicators parameter set to "continuity
check failed" or the ISUP (or BICC) timer T8 expires.
A Reason header field containing the (Q.850) Cause Value 41 Temporary Failure shall be added to the CANCEL
request to be sent by the SIP side of the O-MGCF.
O-MGCF
When receiving a SIP response with a response code 3xx, the default behaviour of the O-MGCF is to release the call
with a cause code value 127 (Interworking unspecified).
NOTE: The O-MGCF may also decide for example to redirect the call towards the URIs in the Contact header
field of the response as an operator option, but such handling is outside of the scope of the present
document.
7.2.3.2.20 Sending of INFO for overlap signalling using the in-dialog method
7.2.3.2.20.1 General
SIP INFO request are used to carry additional digits when overlap signalling is sent using the in-dialog method as
described in Clause 7.2.3.2.1a.2, which is a network option. Clause 7.2.3.2.1a.2 also describes when the O-MGCF sends
a SIP INFO request.
Table 18b provides a summary of how the header fields within the outgoing INFO messages are populated when in-
dialog SIP INFO requests are used for overlap dialling.
SAM INFO
Digits See Annex G
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7.2.3.3 Timers
Table 19: Timers for interworking
Symbol Time-out value Cause for initiation Normal termination At expiry Reference
Ti/w1 4 s to 6 s When last address At the receipt of fresh address Send INVITE, 7.2.3.2.1
(default of 4 s) message is received and information. send the 7.2.3.2.4
the minimum number of address (NOTE 1)
digits required for routing complete
the call has been message
received.
Ti/w2 4 s to 20 s When INVITE is sent On reception of 180 Ringing, Send ACM (no 7.2.3.2.4
(default of 4 s) unless the ACM has or 183 Session Progress and indication) 7.2.3.2.1
already been sent. P-Early-Media header (NOTE 2)
authorizing backward early
media, or 181 Call is Being
Forwarded, or 404 Not Found
or 484 Address Incomplete for
an INVITE transaction for
which Ti/w3 is running, or 200
OK (INVITE).
Ti/w3 4-6 seconds On receipt of 404 Not At the receipt of SAM Send REL with 7.2.3.2.1A,
(default of 4 Found or 484 Address Cause Value 7.2.3.2.12.1
seconds) Incomplete if there is no 28 to the (NOTE 3)
other pending INVITE BICC/ISUP
transactions for the side.
corresponding call.
NOTE 1: This timer is used when overlap signalling is received from BICC/ISUP network and converted to en-block
signalling at the MGCF.
NOTE 2: This timer is used to send an early ACM if a delay is encountered in receiving a response from the
subsequent SIP network.
NOTE 3: This timer is known as the "SIP dialog protection timer". This timer is only used where the O-MGCF is
configured to send INVITE before end of address signalling is determined.
Figure 25: Control Plane interworking between CS networks supporting BICC over MTP3 and the IM
CN subsystem
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Figure 26: Control Plane interworking between CS networks supporting BICC over MTP3B over AAL5
and the IM CN subsystem
If ATM transport is applied between the SS7 Signalling function and the Signalling Gateway Function, they shall apply
MTP3B (ITU-T Recommendation Q.2210 [46]) over SSCF (ITU-T Recommendation Q.2140 [45]) over SSCOP (ITU-
T Recommendation Q.2110 [44]) over AAL5 (ITU-T Recommendation I.363.5 [43]) as depicted in figure 26.
If IP transport is applied between the SS7 Signalling function and the MGCF, they shall support and apply M3UA,
SCTP and IP (either IPv4, see RFC 791 [16], or IPv6, see RFC 2460 [39]), as depicted in figure 27.
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STC provides the services for the transparent transfer of STC user information, e.g. BICC, between STC users, i.e. the
SS7 signalling function and the MGCF (see 3GPP TS 29.205 [14]).
STC performs the functions of data transfer service availability reporting and congestion reporting to the STC user and
User part availability control. See ITU-T Recommendation Q.2150.1 [29].
An I-MGCF shall support both incoming INVITE requests containing SIP preconditions and 100rel extensions in the
SIP Supported or Require header fields, and INVITE requests not containing these extensions, unless the Note below
applies.
NOTE: If the I-MGCF is deployed in an IMS network that by local configuration serves no user requiring
preconditions, the MGCF may not support incoming requests requiring preconditions.
The I-MGCF shall interwork forked INVITE requests with different request URIs.
I-MGCF
INVITE
IAM
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The I-MGCF shall reject an INVITE request for a session only containing unsupported media types by sending a status
code 488 "Not Acceptable Here". If audio media streams and non-audio media streams are contained in a single
INVITE request, the non-audio media streams shall be rejected in the SDP answer, as detailed in IETF RFC 3264 [36].
The I-MGCF shall include a To tag in the first backward non-100 provisional response, in order to establish an early
dialog as described in IETF RFC 3261 [19].
00 no COT to be expected, if the received SDP does not contain precondition information or
indicates that all preconditions are fulfilled, and all local resource reservation is completed.
10 COT to be expected, if the received SDP indicates that precondition is not fulfilled or any local
resource reservation is not completed.
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I-MGCF
SDP indicating
preconditions met and/or
active media
COT
If the IAM has already been sent, then the Continuity message shall be sent indicating "continuity check successful",
when all of the following conditions have been met.
- When the requested remote preconditions (if any) in the IMS have been met.
- The media stream previously set to inactive mode is set to active (a s specified in IETF RFC 4566 [56]).
7.3.3.1.4C Sending of 183 Session Progress for overlap signalling using the in-dialog
method
See clause 7.2.3.1.4C.
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NOTE: If the network option of overlap signalling using the in-dialog method is applied at the I-MGCF in
combination with the optional codec negotiation interworking with BICC in clause B.2., in order to meet
the requirement in clause B.2.1 that the I-MGCF suspends the SDP answer procedure until it receives
backward codec information from BICC, the 183 session progress response will not contain the SDP
answer.
If the BICC network sends an APM message with DTMF signal, duration and action indicator to the MGCF, the MGCF
may send this information to the IM-MGW via the Mn interface. The IM-MGW shall send the corresponding DTMF
signal and duration information on the user plane of the IM CN subsystem according to RFC 4733 [105].
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The O-MGCF should defer sending the INVITE request until the BICC bearer setup and any local resource reservation
is completed.
NOTE: If the O-MGCF sends the INVITE request before the BICC bearer setup and any local resource
reservation is completed, the following considerations apply: If the receiving terminal is not supporting
the SIP precondition, clipping may occur. Furthermore, if the MGCF sets the SDP "inactive" attribute in
the initial INVITE request and the receiving terminal is not supporting the SIP precondition and the SIP
UPDATE method, the interworking procedures within the present specification do not describe all
necessary signalling interactions required to set up a call, in particular with respect to the sending of the
re-INVITE that may also cause additional delay in the call setup. In addition, the interworking of the
ringing indication might not be possible if the peer sends the ringing indication only as response to a re-
INVITE.
The BICC bearer setup is completed when one of the following conditions is met:
- The event Bearer Set-up indication – for the forward bearer set-up case where the incoming Connect Type is
"notification not required", which indicate successful completion of bearer set-up, is received by the Incoming
bearer set-up procedure, (ITU-T Recommendation Q.1902.4 [30] clause 7.5);
- Bearer Set-up Connect indication for the backward call set-up case, which indicate successful completion of
bearer set-up, is received by the Incoming bearer set-up procedure, (ITU-T Recommendation Q.1902.4 [30]
clause 7.5);
- BNC set-up success indication for cases using bearer control tunnelling which indicate successful completion of
bearer set-up, is received by the Incoming bearer set-up procedure, (ITU-T Recommendation Q.1902.4 [30]
clause 7.5).
If the O-MGCF sends the INVITE request without waiting for the BICC bearer setup and any local resource reservation
to complete (which is not recommended according to subclause 7.3.3.2.1), it shall indicate that SIP preconditions are
not met when the initial INVITE request indicates support of the SIP preconditions.
The SDP media description shall contain precondition information as per 3GPP TS 24.229 [9]. If the local preconditions
are not met the O-MGCF should set the media stream to inactive mode (by including an attribute "a=inactive"). If the
local configuration indicates that O-MGCF is deployed in the IMS network that serves users supporting SIP
precondition mechanism, the attribute "a=inactive" may be omitted when the initial SDP offer indicates local
preconditions are not met. If the initial SDP offer indicates local preconditions are fulfilled, the O-MGCF shall not set
the media stream to inactive mode.
The O-MGCF shall include the AMR codec transported according to IETF RFC 3267 [23] with the options listed in
clause 5.1.1 of 3GPP TS 26.236 [32] in the SDP offer. Within the SDP offer, the O-MGCF should also provide SDP RR
and RS bandwidth modifiers specified in IETF RFC 3556 [59] to disable RTCP, as detailed in Clause 7.4 of 3GPP TS
26.236 [32].
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For speech and video calls, the O-MGCF shall insert an IMS Communication Service Identifier, indicating the IMS
Multimedia Telephony Communication Service.
The IMS Communication Service Identifier for the IMS Multimedia Telephony Communication Service is defined in
3GPP TS 24.173 [88].
O-MGCF
COT (success)
UPDATE
The UPDATE request with a new SDP offer shall be sent for each early SIP dialogue for which the received provisional
response indicated support of preconditions confirming that all the required local preconditions have been met when all
the following conditions are met.
1. A Continuity message, with the Continuity Indicators parameter set to "continuity check successful" shall be
received.
In addition, depending on which bearer set-up procedure used for the call one of the following condition shall be met:
- The event Bearer Set-up indication – for the forward bearer set-up case where the incoming Connect Type is
"notification not required", which indicate successful completion of bearer set-up, is received by the Incoming
bearer set-up procedure, (ITU-T Recommendation Q.1902.4 [30] clause 7.5);
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- Bearer Set-up Connect indication for the backward call set-up case, which indicate successful completion of
bearer set-up, is received by the Incoming bearer set-up procedure, (ITU-T Recommendation Q.1902.4 [30]
clause 7.5);
- BNC set-up success indication for cases using bearer control tunnelling which indicate successful completion of
bearer set-up, is received by the Incoming bearer set-up procedure, (ITU-T Recommendation Q.1902.4 [30]
clause 7.5).
If the O-MGCF previously offered inactive media stream it shall set the media stream to active mode.
NOTE: This procedure applies regardless of whether the early SIP dialog existed prior to the preconditions being
fulfilled or is subsequently created.
The sending of an awaiting answer indication is described in clause 9.2.3.1. and clause 9.2.3.2.
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If the BICC network sends an APM message with DTMF signal, duration and action indicator to the MGCF, the MGCF
may send this information to the IM-MGW via the Mn interface. The IM-MGW shall send the corresponding DTMF
signal and duration information on the user plane of the IM CN subsystem according to 4733 [105].
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7.3.3.2.21 Sending of INFO for overlap signalling using the in-dialog method
See Clause 7.2.3.2.20.
7.3.3.3 Timers
See clause 7.2.3.3.
In the specific case of ISUP originated calls, use of the CLIP service additionally requires the ability to determine
whether the number was network provided or provided by the access signalling system. Due to the possible SIP
indication of the P-Asserted-Identity the Screening indicator is set to network provided as default. For the CLIP-CLIR
service the mapping of the APRI from privacy header at the O-MGCF is described within table 16 in Clause 7.2.3.2.2.6.
At the O-MGCF the presentation restricted indication shall be mapped to the privacy header = "id" and "header". This is
described in table 5 in clause 7.2.3.1.2.6.
NOTE: This implies that all outgoing calls will invoke the COLP service.
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Table 21: Mapping of connected number parameter to SIP P-Asserted-Identity header fields
Table 22: Mapping of BICC/ISUP APRIs into SIP privacy header fields
The O-MGCF has to store the status of the "Connected Line Identity Request indicator".
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ANM or CON message if required by the procedures described in subclause 7.4.2.2.3. In accordance with ISUP
procedures, a connected number shall not be included within the ACM message. The mapping of the of the P-Asserted-
Identity and Privacy header fields is shown in tables 23 and 24.
If no P-Asserted-Identity header field is provided within the 200 OK (INVITE) message, the stored information
previously received in the last provisional 1xx response of the same SIP dialogue shall be used.
NOTE: Due to forking, other P-Asserted-Identities might have been received in different SIP dialogues.
If the Calling Party has requested the COLP service (as indicated by the stored request status) but the 200 OK (INVITE)
and previous 1XX provisional responses do not include a P-Asserted-Identity header field, the O-MGCF shall set up a
network provided Connected Number with an Address not Available indication.
If the P-Asserted-Identity is available then the Connected number has to be setup with the screening indication network
provided. The mapping of the P-Asserted-Identity and Privacy (if available) is shown in table 24.
Table 24: Mapping of P-Asserted-Identity and privacy headers to the ISUP/BICC connected number
parameter
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Release 8 104 3GPP TS 29.163 V8.16.0 (2011-09)
7.4.5.1 General
The ISDN Subaddress in ISUP is transported within the Access Transport Parameter. The Coding of the Subaddress
parameter within the Access Transport Parameter is described within ETSI EN 300 403-1 [96]. The isdn-subaddress
parameter carried within a tel or sip URI is defined within RFC 3966 [97]. The isdn-subaddress encoding type carried
within a tel or sip URI is defined within RFC 4715 [108].
The mapping in Table 24bb of the Subaddress received within an ANM Message containing the ISUP Access Transport
Parameter to the isdn-subaddress of a tel or sip URI to be sent within a 200 OK (INVITE) shall be applied.
Table 24ba: Mapping of the Subaddress received in an initial INVITE to the Subaddress sent in the
IAM
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Release 8 105 3GPP TS 29.163 V8.16.0 (2011-09)
Table 24bb: Mapping of the Subaddress received in an ANM to the Subaddress sent in the 200 OK
(INVITE)
The mapping in Table 24bd of the Subaddress received within an ANM Message containing the ISUP Access Transport
Parameter to the isdn-subaddress of a tel or sip URI to be sent within a 200 OK (INVITE) shall be applied.
Table 24bc: Mapping of the Subaddress received in an IAM to the Subaddress sent in the INVITE
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Table 24bd: Mapping of the Subaddress received in a 200OK to the Subaddress sent in the ANM
7.4.6.1 General
A MGCF within an IMS network applying the Communication Diversion service uses the procedures defined in clause
7.5.4.
This clause describes additional interworking of call forwarding service information between a PSTN/PLMN network
and an IMS network. This can also be used when interworking between SIP networks (e.g., IMS network interworking
with enterprise networks making use of the History-Info header with an escaped Reason header as described in IETF
RFC 4244 [91]). The procedures support the interworking of the diversion reason within the History-Info header using
an escaped Reason header defined by IETF RFC 3326 [116] as described in IETF RFC 4244 [91] and also support the
diversion cause using the cause-parm as described by IETF RFC 4458 [113].
If the MGCF is supporting the interworking of Call Forwarding and also applying the Communication Diversion
services as defined by 3GPP TS 24.604 [60], it uses both IETF RFC 4244 [91] and IETF RFC 4458 [113] to signal the
diversion reason. An IMS network supporting the interworking of Call Forwarding and not applying the IMS
Communication Diversion supplementary service according to 3GPP TS 24.604 [60] uses IETF RFC 4244 [91] and can
also use IETF RFC 4458 [113] based on network option.
When interworking the SIP History-Info header to ISUP and the diversion reason based on IETF RFC 4244 [91] and
IETF RFC 4458 [113] are both present for the same diversion instance, the MGCF should use the cause-parm URI
parameter per IETF RFC 4458 [113] for deriving the ISUP. When the cause-parm URI parameter per IETF RFC 4458
[113] is used, the corresponding interworking as described in cause 7.5.4 shall be applied for that diversion instance.
Otherwise, the interworking as defined in the present clause should be applied.
When interworking from ISUP to the History-Info header, to signal the diversion reason the MGCF should use the
escaped Reason header from IETF RFC 4244 [91] and apply the interworking procedures in the present clause. The
MGCF should also apply the interworking procedures described in clause 7.5.4 to use the cause-parm from IETF RFC
4458 [113] to signal the diversion reason.
In the event that the interworking procedures described in this clause as well as the interworking procedures in clause
7.5.4 are not applied, the actions of the MGCF at the ISUP/BICC side as described in ITU-T Recommendation Q.732.2-
4 [42] under the clause "Interactions with networks not providing any call diversion information" may be applied as a
network option.
7.4.6.2.1 General
This sub-clause describes the optional mapping of Call Forwarding information at the O-MGCF to the protocol-cause
specified in IETF RFC 3326 [116].
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Source SIP header field Source Redirection number Derived value of parameter field
and component Component
value
hi-targeted-to-uri of the CC Nature of address indicator If CC is equal to the country code of the
History-Info header country where O-MGCF is located AND
following the last History- the next ISUP node is located in the
Info entry field entry same country, then set to "national
containing a Reason (significant) number" else set to
header as defined in IETF "international number".
RFC 4244 [91] with cause CC, NDC, SN Address signals If NOA is "national (significant) number"
value as listed in table then set to
7.4.6.2.2.4 NDC + SN.
appropriate global number If NOA is "international number"
portion of the hi-targeted-to- then set to CC + NDC + SN.
uri, assumed to be in form
"+" CC + NDC +
SN.(NOTE)
NOTE: If it is SIP URI and doesn‟t contain "user=phone", mapping to redirection number is impossible,
therefore no need to generate Redirection number and Redirection number restriction indicator (per
Table 7.4.6.2.2.3), Notification subscription options can‟t be set as "presentation allowed with
redirection number".
Table 7.4.6.2.2.3: Mapping of History-Info header field to ISUP Redirection number restriction
Source SIP header field Source Redirection number Derived value of parameter field
and component Component restriction
value
Privacy header field, "history" or Presentation restricted "Presentation restricted"
priv-value component "session" or indicator
"header"
Privacy "Presentation allowed" or absent
header field
absent or
"none"
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Release 8 108 3GPP TS 29.163 V8.16.0 (2011-09)
Source SIP header field Source Component Call Diversion Derived value of parameter field
and component value Information
Privacy header field, "history" or "session" or Notification If the priv-value "history" or "session" or
priv-value component "header" subscription "header" is set for the History-Info
options header field or to the hist-info element
entries concerning the redirecting (see
Table 7.4.6.3.2.2) and diverted to uri
(see Table 7.4.6.2.2.2) then
presentation not allowed shall be set
If the priv-value "history" or "session" or
"header" is set only to the hist-info
element concerning the diverted-to uri
then presentation allowed without
redirection number shall be set
Privacy header field Presentation allowed with redirection
absent number
or "none"
hi-targeted-to-uri; Reason Cause parameter Call diversion Redirecting Reason
header as defined in IETF value information
RFC 4244 [91] with cause 302 Deflection immediate response
parameter 486 User busy
408 No reply
503 Mobile subscriber not reachable
all other cause values Unknown
Table 7.4.6.2.2.6: Mapping of 181 (Call Is Being Forwarded) ACM when ACM was not previously
sent
Source SIP header field Source Component ISUP Parameter Derived value of parameter
and component value field
181 (Call Is Being ACM
Forwarded)
Generic notification Call is diverting
indicators
History-Info header field See table 7.4.6.2.2.2 Redirection number See table 7.4.6.2.2.2
Priv-value See table 7.4.6.2.2.3 Redirection number See table 7.4.6.2.2.3
restriction
Priv-value See table 7.4.6.2.2.4 Call diversion information See table 7.4.6.2.2.4
Notification subscription
options
hi-targeted-to-uri; Reason See table 7.4.6.2.2.4 Call diversion information Redirecting Reason
header as defined in IETF See table 7.4.6.2.2.4
RFC 4244 [91] with cause
parameter
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Release 8 109 3GPP TS 29.163 V8.16.0 (2011-09)
Table 7.4.6.2.2.7: Mapping of 181 (Call Is Being Forwarded) CPG if ACM was already sent
Source SIP header field Source Component ISUP Parameter Derived value of parameter
and component value field
181 (Call Is Being CPG
Forwarded) response
Generic notification Call is diverting
indicators
hi-targeted-to-uri; Reason 486 Event indicator CFB (national use)
header as defined in IETF 408 (see NOTE) CFNR (national use)
RFC 4244 [91] with cause all other values, or if Progress
parameter appropriate national
use value CFB, CFNR
or CFU is not used in a
network, or if no hi-
targeted-to-uri cause-
param URI parameter
is contained in the SIP
181.
History-Info header field See table 7.4.6.2.2.2 Redirection number See table 7.4.6.2.2.2
Priv-value See table 7.4.6.2.2.3 Redirection number See table 7.4.6.2.2.3
restriction
Priv-value See table 7.4.6.2.2.4 Call diversion information See table 7.4.6.2.2.4
Notification subscription
options
hi-targeted-to-uri; Reason See table 7.4.6.2.2.4 Call diversion information See table 7.4.6.2.2.4
header as defined in IETF Redirecting Reason
RFC 4244 [91] with cause
parameter
NOTE: This appears in the cases of CFNR.
Table 7.4.6.2.2.8: Mapping of 180 (Ringing) ACM when ACM was not previously sent
Source SIP header field Source Component ISUP Parameter Derived value of parameter
and component value field
180 (Ringing) response ACM
History-Info header field If hi-index indicate that Generic notification Call is diverting
there is a call indicators
forwarding.
History-Info header field See table 7.4.6.2.2.2 Redirection number (NOTE) See table 7.4.6.2.2.2
Priv-value See table 7.4.6.2.2.3 Redirection number See table 7.4.6.2.2.3
restriction (NOTE)
Priv-value See table 7.4.6.2.2.4 Call diversion information See table 7.4.6.2.2.4
Notification subscription
options (NOTE)
hi-targeted-to-uri; Reason See table 7.4.6.2.2.4 Call diversion information See table 7.4.6.2.2.4
header as defined in IETF Redirecting Reason (NOTE)
RFC 4244 [91] with cause
parameter
NOTE: Parameter shall only be supplied if hi-targeted-to-uri contains a cause-param URI parameter, as
defined in IETF RFC 4244 [91].
The mapping described within table 7.4.6.2.2.1 can only appear if the communication has already undergone a Call
Forwarding in the ISDN/PSTN and the 180 is the first provisional response sent in backward direction.
The IWU can indicate the call diversion in the mapping of 180 (Ringing) to CPG in fact if the response before was
a 181 (Call is being forwarded).
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Release 8 110 3GPP TS 29.163 V8.16.0 (2011-09)
Table 7.4.6.2.2.9: Mapping of 180 (Ringing) CPG if ACM was already sent
Source SIP header field Source Component ISUP Parameter Derived value of parameter
and component value field
180 (Ringing) response CPG
Generic notification Call is diverting
indicators
History-Info header field Event indicator ALERTING
History-Info header field See table 7.4.6.2.2.2 Redirection number (NOTE) See table 7.4.6.2.2.2
Priv-value See table 7.4.6.2.2.3 Redirection number See table 7.4.6.2.2.3
restriction (NOTE)
Priv-value See table 7.4.6.2.2.4 Call diversion information See table 7.4.6.2.2.4
Notification subscription
options (NOTE)
hi-targeted-to-uri; Reason See table 7.4.6.2.2.4 Call diversion information See table 7.4.6.2.2.4
header as defined in IETF Redirecting Reason (NOTE)
RFC 4244 [91] with cause
parameter
NOTE: Parameter shall only be supplied if hi-targeted-to-uri contains a cause-param URI parameter, as
defined in IETF RFC 4244 [91].
The mapping in table 7.4.6.2.2.9 appears when a 181 previously was mapped to an ACM. Therefore the state machine
of the MGCF knows that a CDIV is in Progress.
Source SIP header field Source Component ISUP Parameter Derived value of parameter
and component value field
200 (OK) response ANM/CON
History-Info header field See table 7.4.6.2.2.2 Redirection number See table 7.4.6.2.2.2
Priv-value See table 7.4.6.2.2.3 Redirection number See table 7.4.6.2.2.3
restriction
To interwork the redirection number at the O-MGCF it can be needed to create placeholder History entries. Such a
History entry has to provide a hi-targeted-to-uri with a placeholder value "unknown@unknown.invalid" a Cause
parameter and a hi-index as described within table 7.4.6.2.3.1.
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7.4.6.3.1 General
This sub-clause describes the interworking of the Call Forwarding information at the I-MGCF.
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Release 8 113 3GPP TS 29.163 V8.16.0 (2011-09)
Source SIP header field Source Redirecting number Derived value of parameter field
and component Component
value
latest History-Info header Redirecting number
field entry containing a
Reason header as defined
in IETF RFC 4244 [91] with
cause parameter value as
listed in the
cause parameter row in
Table 7.4.6.2.2.4 (Note 1)
hi-targeted-to-uri CC Nature of address indicator If CC is equal to the country code of the
appropriate global number country where MGCF is located AND
portion of the URI, the next ISUP node is located in the
assumed to be in form same country, then set to "national
"+" CC + NDC + SN (significant) number" else set to
"international number"
CC, NDC, SN Address signals If NOA is "national (significant) number"
then set to
NDC + SN.
If NOA is "international number"
then set to CC + NDC + SN
Privacy header field, "history" or APRI "presentation restricted"
priv-value component "session" or
in History-Info header field "header"
as specified in this Privacy "presentation allowed"
table(NOTE 2) header field
absent or
"none"
NOTE 1: If it is SIP URI and doesn‟t contain "user=phone", mapping to redirecting number is impossible,
therefore no need to generate Redirecting number
NOTE 2: It is possible that an entry of the History-Info header field itself is marked as restricted or the whole
History-Info header.
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Release 8 114 3GPP TS 29.163 V8.16.0 (2011-09)
Source SIP header field Source Component Redirection Derived value of parameter field
and component value Information
Privacy header field, "history"or "session" or Redirection Call diverted, all redirection info
priv-value component of the "header" indicator presentation restricted
History In History-Info for the Privacy SIP
header field of the History- header or for any of the
Info Entry following the last two hi-targeted-to-uri
hi-targeted-to-uri containing entries
a Reason header as Privacy header field Call diverted
defined in IETF RFC 4244 absent
[91] with cause parameter or
value as listed in the cause "none"
parameter row in this table for the Privacy SIP
or as header itself header and for both the
two hi-targeted-to-uri
entries
Original Unknown
redirection reason
Cause parameter in the last Cause parameter Redirecting Redirecting Reason
hi-targeted-to-uri containing value Reason
a Reason header as 302 Deflection immediate response
defined in IETF RFC 4244 486 User busy
[91] 408 No reply
503 Mobile subscriber not reachable
All other values unknown
Hi-index Redirection number of History entries containing a
counter cause-param with value as listed in the
cause-param row in this table
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Release 8 115 3GPP TS 29.163 V8.16.0 (2011-09)
Table 7.4.6.3.2.4: Mapping of History-Info header field to ISUP Original Called number
Source SIP header field Source Original called number Derived value of parameter field
and component Component
value
Numbering Plan Indicator "ISDN (Telephony) numbering plan
(Recommendation E.164)"
st
hi-targeted-to-uri of 1 hi- CC Nature of address indicator If CC is equal to the country code of the
targeted-to-uri containing a country where MGCF is located AND
Reason header as defined the next ISUP node is located in the
in IETF RFC 4244 [91] with same country, then set to "national
cause parameter; (significant) number" else set to
appropriate global number "international number"
portion of the URI, CC, NDC, SN Address signals If NOA is "national (significant) number"
assumed to be in form then set to
"+" CC + NDC + SN NDC + SN.
(NOTE 1) If NOA is "international number"
then set to CC + NDC + SN
priv-value component "history" or APRI "presentation restricted"
in History-Info header field "session" or
of the History-Info header "header"
field entry as defined above Privacy "presentation allowed"
in this table (NOTE 2) header field
absent or
"none"
NOTE 1: If it is SIP URI and doesn‟t contain "user=phone", mapping to Original Called number is impossible,
therefore no need to generate Original Called number
NOTE 2: It is possible that an entry of the History-Info header field itself is marked as restricted or the whole
History-Info header.
INVITE IAM
History-Info header See table 7.4.6.3.2.2 Redirecting See table 7.4.6.3.2.2
field number
History-Info header See table 7.4.6.3.2.3 Redirection See table 7.4.6.3.2.3
field Information
Cause parameter in Cause parameter value Redirection Redirecting Reason
the last hi-targeted-to- 486 Information User busy
uri containing a 408 No reply
Reason header as 302 Deflection immediate response
defined in IETF RFC 503 Mobile subscriber not reachable
4244 [91]
All other values unknown
History-Info header See table 7.4.6.3.2.4 Original Called See table 7.4.6.3.2.4
field Number
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Release 8 116 3GPP TS 29.163 V8.16.0 (2011-09)
In the ISUP destination exchange of the diverted-to user only the Redirection Number Restriction parameter will be
included into the ACM, CPG, ANM or CON message. Therefore only the mapping of this parameter is shown in the
following table.
Table 7.4.6.3.3.2: Mapping of ISUP Redirection Number Restriction to History-Info header field
A received CPG shall be mapped t a 180 (Ringing) response if the CPC indicates an Alerting is due to the mapping
rules defined within the basic call.
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Release 8 117 3GPP TS 29.163 V8.16.0 (2011-09)
3GPP
Release 8 118 3GPP TS 29.163 V8.16.0 (2011-09)
3GPP
Release 8 119 3GPP TS 29.163 V8.16.0 (2011-09)
Table 7.4.6.3.3.6: Mapping of ANM 200 (OK) response (to INVITE request)
In the event that the interworking procedures described in this clause are not applied, the actions of the MGCF at the
ISUP/BICC side described in ITU-T Recommendation Q.732.5 [42] under the clause "Interactions with other networks"
may be applied as a network option.
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Release 8 120 3GPP TS 29.163 V8.16.0 (2011-09)
ISUP/BICC side are described in ITU-T Recommendation Q.732.7 [42] under the clause "Interactions with other
networks".
Table 24be: Mapping between ISUP and SIP for the Explicit Communication Transfer supplementary
service
The user plane interworking of the hold/resume request is described in the clause 9.2.9.
MGCF
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When an MGCF receives a CPG message with a "remote hold" Generic notification indicator and the media on the IMS
side are not "sendonly" or "inactive", the MGCF shall forward the hold request by sending an UPDATE or re-INVITE
message on the early dialog which was last established containing SDP with "sendonly" or "inactive" media, as
described in RFC 3264 [36].
If additional early dialog is established during the "remote hold" condition the MGCF shall send UPDATE or re-
INVITE request containing an SDP with "sendonly" or "inactive" media on the new established dialog , as described in
RFC 3264 [36].
If an UPDATE request with an SDP offer is received on one of the early dialogs for a call in the "remote hold"
condition the MGCFshall send an appropriate SDP answer followed by a new UPDATE requests including SDP with
"sendonly" or "inactive" media on the dialog , as described in RFC 3264 [36].
If a MGCF receives a 200 OK response on a dialog for which the call is "remote hold" condition the MGCF shall send
UPDATE or re-INVITE request containing an SDP with "sendonly" or "inactive" media on the dialog where 200 OK
was received, as described in RFC 3264 [36].
If the MGCF receives a CPG with Generic Notification Indicator "remote retrieval" for a early dialog then a SIP
UPDATE request (indicating call retrieval) shall be sent if the call hold service had been invoked on the early dialog
before. For each subsequent early dialog for which the MSC receives an 18x response or an UPDATE request with an
SDP offer, the MSC shall sent SIP UPDATE indicating call retrieval after a possible SDP answer to the SDP offer, if
that dialog had received a call hold indication before.
If the MGCF receives a CPG with Generic Notification Indicator "remote retrieval" on a confirmed dialog then a SIP
RE-INVITE or UPDATE request (according to implementation option]) shall be sent for this dialog only if the call hold
service had been invoked for this dialog before.
When an MGCF receives a CPG message with a "remote retrieval" Generic notification indicator and the media on the
IMS side are not "sendrecv" or "recvonly", the MGCF shall forward the resume request by sending an UPDATE or re-
INVITE message containing SDP with "sendrecv" or "recvonly" media, as described in RFC 3264 [36].
If the MGCF receives a CPG with "remote hold"’ or "remote retrieval" before answer, it shall forward the request using
an UPDATE message. If the MGCF receives a CPG with "remote hold" or "remote retrieval" after answer, it should
forward the request using re-INVITE but may use UPDATE.
If link aliveness information is required at the IM-MGW while the media are on hold, the O-MGCF should provide
modified SDP RR and RS bandwidth modifiers specified in IETF RFC 3556 [59] within the UPDATE or re-INVITE
messages holding and retrieving the media to temporarily enable RTCP while the media are on hold, as detailed in
Clause 7.4 of 3GPP TS 26.236 [32]. If no link aliveness information is required at the IM-MGW, the O-MGCF should
provide the SDP RR and RS bandwidth modifiers previously used.
The interworking does not impact the user plane, unless the MGCF provides modified SDP RR and RS bandwidth
modifiers within the UPDATE or re-INVITE messages. If the MGCF provides modified SDP RR and RS bandwidth
modifiers to the IMS side, the MGCF shall also provide modified SDP RR and RS bandwidths to the IM-MGW, as
described in the clause 9.2.10.
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Release 8 122 3GPP TS 29.163 V8.16.0 (2011-09)
MGCF
A Suspend message containing the Suspend/Resume indicators set to "ISDN subscriber initiated" shall be treated like a
CPG with "remote hold" in Section 7.4.10 Resume message containing the Suspend/Resume indicators set to "ISDN
subscriber initiated" shall be treated like a CPG with "remote retrieval" in Section 7.4.10.
Alternatively, the MGCF may apply the interworking to the Conference supplementary service described in subclause
7.5.6.
3GPP
Release 8 123 3GPP TS 29.163 V8.16.0 (2011-09)
Table 24aa: Mapping between ISUP and SIP for the Conference Calling (CONF) and Three-Party
Service (3PTY) supplementary service
7.4.15 Void
Editors Note: IETF draft-johnston- sipping-cc-uui needs more detail on encoding of the UUI as defined within
ITU-T Q.763 i.e., the protocol discriminator and the encoding syntax.
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Release 8 124 3GPP TS 29.163 V8.16.0 (2011-09)
The format of the uuidata field shall be the hexadecimal representation of binary data coded in ascii alphanumeric
characters. For example, the 8- bit binary value 0011- 1111 is 3F in hexadecimal. To code this in ascii, one 8- bit byte
containing the ascii code for the character '3' (0011- 0011 or 033H) and one 8- bit byte containing the ascii code for the
character 'F' (0100- 0110 or 046H) are required. For each byte value, the high-order hexadecimal digit is always the first
digit of the pair of hexadecimal digits. The ascii letters used for the hex digits shall always be capital form.
For example:
User-to-User: 00C81031313232333334343535363637373838FA08303900064630E9E0;encoding=hex
Interworking procedures between the user-user information element and User-to-User header for the User-to-user
signalling service 1 are defined in the following subsections.
Editors Note: For 3GPP Release 8 the encoding of data should be described in more detail.
Editors Note: For 3GPP Release 8 the interworking of the encoded data should be described in more detail.
The "length of user-user contents" parameter shall be set by the I-MGCF according to the normal procedures.
The I-MGCF maps the messages transporting the user-user information according to the normal interworking
procedures.
Table 24ab Mapping of the User-to-User header to the ISUP user-to-user information parameter
The O-MGCF shall set the encoding header field parameter of the User-to-User header to the "hex" value
The O-MGCF maps the messages transporting the user-user information according to the normal interworking
procedures.
Table 24ac Mapping of the ISUP user-to-user information parameter to the User-to-User header
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Release 8 125 3GPP TS 29.163 V8.16.0 (2011-09)
7.5.2.1 General
The protocol specification of the Terminating Identification Presentation and Terminating Identification Restriction
supplementary services is described in 3GPP TS 24.608 [64].
If an Optional forward call indicators parameter in the IAM is received where the bit H Connected line identity request
indicator is set to "requested", then the option tag "from-change" shall be add to the Supported header field. See table
7.5.2.2.1.
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ISUP Parameter Derived value of parameter Source SIP header field Source Component
field and component value
Optional forward call indicator Connected line identity request Supported "from-change"
indicator is set to "requested"
If a provisional or final response including the option tag "from-change" is received , then the O-MGCF shall:
- if a 200 (OK) response to the INVITE request is received, start timer T TIR1; and
Otherwise the 200 (OK) response (to the INVITE request) shall be mapped as described in subclause 7.2.3.2.7a.
If an UPDATE request is received containing a changed From header field before the timer TTIR1 expired, then the
O-MGCF shall:
- map the From header field received in the UPDATE request to the Generic number in the ANM as shown in
table 7.5.2.2.2 and table 7.5.2.2.3;
- if the UPDATE request includes a P-Asserted-Identity header field that is different from the one within the
stored 200 (OK) response, the latest received P-Asserted-Identity header field shall be mapped to the connected
number as described table 24; and
- map the parameters needed to be mapped of the stored 200 (OK) response to an ANM as described in subclause
7.4.2.2.3, modified by the changed mapping steps of the From and P-Asserted-Identity header fields.
When TTIR1 expires, then the stored 200 (OK) response (to the INVITE request) response shall be mapped as described
in subclause 7.4.2.2.3.
ANM/CON UPDATE
Generic number From header field
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Table 7.5.2.2.3: Mapping of SIP From header field to ISUP Generic Number
("additional connected number") parameter
Source SIP header Source Generic Number Derived value of parameter field
field and component component value parameter field
– – Number Qualifier Indicator "additional connected number"
From, userinfo CC Nature of Address Indicator If CC is equal to the country code of the
component of URI country where I-MGCF is located AND
assumed to be in form the next ISUP node is located in the
"+" CC + NDC + SN same country, then set to "national
(significant) number" else set to
"international number"
– – Number Incomplete "complete"
Indicator
– – Numbering Plan Indicator "ISDN (Telephony) numbering plan
(Recommendation E.164)"
Privacy, priv-value Privacy header Address Presentation "presentation allowed"
component field absent Restricted Indicator (APRI)
"none" "presentation allowed"
"header" "presentation restricted"
"user" "presentation restricted"
– – Screening Indicator "user provided, not verified"
From, userinfo CC, NDC, SN Address Signals If NOA is "national (significant) number"
component assumed to then set to NDC + SN.
be in form If NOA is "international number"
"+" CC + NDC + SN then set to CC + NDC + SN
- if an option tag "from-change" is included within the Supported header field of the received INVITE request,
then the bit H Connected line identity request indicator of the Optional forward call indicators parameter in the
IAM shall be set to "requested".
Table 7.5.2.3.1: Mapping of a SIP Supported header field of a SIP INVITE request
to the ISUP Optional forward call indicator of a ISUP IAM
Source SIP header field Source Component value ISUP Parameter Derived value of
and component parameter field
Supported "from-change" Optional forward call indicator Connected line identity
request indicator is set to
"requested".
If a received ISUP ANM includes a ISUP Generic Number ("additional connected number") parameter, then the
I-MGCF shall send a 200 (OK) response (to the INVITE request) including a option tag "from-change" If the initial
INVITE was received and the Supported header contains the "from-change" tag, the 200 (OK) response is followed by
an UPDATE request, containing the "additional connected number" copied into the From header as shown in
table 7.5.2.3.2.
The to header field of the UPDATE request is derived from the P-Asserted-Identity header field received within the
initial INVITE request.
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A received connected number in an ANM shall be mapped to the P-Asserted-Identity header field as shown in table 21
of the UPDATE request.
7.5.2.4 Timer
Table 7.5.2.4.1 TIR timer definition
7.5.3 (void)
7.5.4.1 General
The protocol specification of the Communication Diversion supplementary service is described in 3GPP TS 24.604
[60]. The mapping of Communication Diversion supplementary service with Call Diversion services PSTN/ISDN
supplementary service including the mapping of the optional History-Info header field as defined in IETF RFC 4244
[91] is described.
In case of interworking with networks which do not provide any notification of the communication diversion or
communication redirection information (e.g. redirection counter) in the signalling system, the communication continues
according to the basic call procedures.
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7.5.4.2.1 General
For the mapping of IAM to the INVITE request no additional procedures beyond the basic call and interworking
procedures are needed unless Call forwarding within the ISUP Network appeared.
Source SIP header field Source Redirection Derived value of parameter field
and component Component number
value
hi-targeted-to-uri of the last CC Nature of address If CC is equal to the country code of the
History-Info hi-entry indicator country where O-MGCF is located AND the
containing a cause-param next ISUP node is located in the same country,
URI parameter, as defined then set to "national (significant) number" else
in IETF RFC 4458 [113]. set to "international number".
The global number portion CC, NDC, SN Address signals If NOA is "national (significant) number" then
of the hi-targeted-to-uri is set to
assumed to be in form NDC + SN.
"+" CC + NDC + SN. If NOA is "international number"
(NOTE) then set to CC + NDC + SN.
NOTE: If it is SIP URI and doesn‟t contain "user=phone", mapping to redirection number is impossible,
therefore no need to generate Redirection number and Redirection number restriction indicator (per
Table 7.5.4.2.1.3), Notification subscription options can‟t be set as "presentation allowed with
redirection number".
Table 7.5.4.2.1.3: Mapping of History-Info header field to ISUP Redirection number restriction
Source SIP header Source Component value Redirection number Derived value of
field and component restriction parameter field
Privacy header field, "history" or "session" or "header" Presentation restricted "Presentation restricted"
priv-value component Privacy header field absent indicator "Presentation allowed" or
or "none" absent
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Source SIP header field Source Component Call Diversion Derived value of parameter field
and component value Information
Privacy header field, "history" or "session" or Notification If the priv-value "history" or "session" or
priv-value component "header" subscription "header" is set for the History-Info
options header field or to the hist-info element
entries concerning the redirecting (see
Table 7.5.4.3.2) and diverted to uri (see
Table 7.5.4.2.1.2) then presentation not
allowed shall be set.
If the priv-value "history" or "session" or
"header" is set only to the hist-info
element concerning the diverted-to uri
then presentation allowed without
redirection number shall be set.
Privacy header field Presentation allowed with redirection
absent number
or "none"
hi- targeted-to-uri cause- cause-param value Call diversion Redirecting Reason
param URI parameter, as 404 information Unknown
defined in IETF RFC 4458 302 Unconditional
[113] of the last History-Info 486 User busy
hi-entry containing such 408 No reply
cause-param. 480 Deflection immediate
503 Mobile subscriber not reachable
487 Deflection during alerting
Table 7.5.4.2.1.6: Mapping of 181 (Call Is Being Forwarded) ACM if no ACM was sent before
Source SIP header field Source Component ISUP Parameter Derived value of parameter
and component value field
181 (Call Is Being ACM
Forwarded)
Generic notification Call is diverting
indicators
History-Info header field See table 7.5.4.2.1.2 Redirection number See table 7.5.4.2.1.2
Priv-value See table 7.5.4.2.1.3 Redirection number See table 7.5.4.2.1.3
restriction
Priv-value See table 7.5.4.2.1.4 Call diversion information See table 7.5.4.2.1.4
Notification subscription
options
hi-targeted-to-uri;cause- See table 7.5.4.2.1.4 Call diversion information Redirecting Reason
param URI parameter as See table 7.5.4.2.1.4
defined in IETF RFC 4458
[113] of the last History-Info
hi-entry containing such
cause-param.
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Table 7.5.4.2.1.7: Mapping of 181 (Call Is Being Forwarded) CPG if ACM was already sent
Source SIP header field Source Component ISUP Parameter Derived value of parameter
and component value field
181 (Call Is Being CPG
Forwarded) response
Generic notification Call is diverting
indicators
hi-targeted-to-uri; cause- 486 Event indicator CFB (national use)
param URI parameter, as 408 (see NOTE) CFNR (national use)
defined in IETF RFC 4458 302 CFU (national use)
[113] of the last History-Info Any other value, or if PROGRESS
hi-entry containing such appropriate national
cause-param. use value CFB, CFNR
or CFU is not used in a
network. or if no
agreement exists
between operators to
use theses values, or if
no hi-targeted-to-uri
cause-param URI
parameter is contained
in the SIP 181.
History-Info header field See table 7.5.4.2.1.2 Redirection number See table 7.5.4.2.1.2
Priv-value See table 7.5.4.2.1.3 Redirection number See table 7.5.4.2.1.3
restriction
Priv-value See table 7.5.4.2.1.4 Call diversion information See table 7.5.4.2.1.4
Notification subscription
options
hi-targeted-to-uri; cause- See table 7.5.4.2.1.4 Call diversion information See table 7.5.4.2.1.4
param URI parameter, as Redirecting Reason
defined in IETF RFC 4458
[113] of the last History-Info
hi-entry containing such
cause-param.
NOTE: This appears in the cases of CFNR.
Table 7.5.4.2.1.8: Mapping of 180 (Ringing) ACM if no ACM was sent before
Source SIP header field Source Component ISUP Parameter Derived value of parameter
and component value field
180 (Ringing) response ACM
History-Info header field If hi-targeted-to-uri of at Generic notification Call is diverting
least one History-Info indicators
hi-entry contains a
cause-param URI
parameter, as defined
in IETF RFC 4458
[113].
History-Info header field See table 7.5.4.2.1.2 Redirection number (NOTE) See table 7.5.4.2.1.2
Priv-value See table 7.5.4.2.1.3 Redirection number See table 7.5.4.2.1.3
restriction (NOTE)
Priv-value See table 7.5.4.2.1.4 Call diversion information See table 7.5.4.2.1.4
Notification subscription
options (NOTE)
hi-targeted-to-uri; cause- See table 7.5.4.2.1.4 Call diversion informationSee table 7.5.4.2.1.4
param URI parameter, as Redirecting Reason (NOTE)
defined in IETF RFC 4458
[113] of the last History-Info
hi-entry containing such
cause-param.
NOTE: Parameter shall only be supplied if hi-targeted-to-uri of at least one History-Info hi-entry contains a
cause-param URI parameter, as defined in IETF RFC 4458 [113].
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The mapping described within table 7.5.4.2.1.1 can only appear if the communication has already undergone a Call
Forwarding in the ISDN/PSTN and the 180 is the first provisional response sent in backward direction.
The IWU can indicate the call diversion in the mapping of 180 (Ringing) to CPG in fact if the response before was
a 181 (Call is being forwarded).
Table 7.5.4.2.1.9: Mapping of 180 (Ringing) CPG if ACM was already sent
Source SIP header field Source Component ISUP Parameter Derived value of parameter
and component value field
180 (Ringing) response CPG
History-Info header field If hi-targeted-to-uri of at Generic notification Call is diverting
least one History-Info indicators
hi-entry contains a
cause-param URI
parameter, as defined
in IETF RFC 4458
[113].
Event indicator ALERTING
History-Info header field See table 7.5.4.2.1.2 Redirection number See table 7.5.4.2.1.2
(NOTE)
Priv-value See table 7.5.4.2.1.3 Redirection number See table 7.5.4.2.1.3
restriction (NOTE)
Priv-value See table 7.5.4.2.1.4 Call diversion information See table 7.5.4.2.1.4
Notification subscription
options (NOTE)
hi-targeted-to-uri; cause- See table 7.5.4.2.1.4 Call diversion information See table 7.5.4.2.1.4
param URI parameter, as Redirecting Reason (NOTE)
defined in IETF RFC 4458
[113] of the last History-Info
hi-entry containing such
cause-param.
NOTE: Parameter shall only be supplied if hi-targeted-to-uri of at least one History-Info hi-entry contains a
cause-param URI parameter, as defined in IETF RFC 4458 [113].
The mapping in table 7.5.4.2.1.9 appears when a 181 previously was mapped to an ACM. Therefore the state machine
of the MGCF knows that a CDIV is in Progress.
Source SIP header field Source Component ISUP Parameter Derived value of parameter
and component value field
200 (OK) response ANM/CON
History-Info header field See table 7.5.4.2.1.2 Redirection number See table 7.5.4.2.1.2
Priv-value See table 7.5.4.2.1.3 Redirection number See table 7.5.4.2.1.3
restriction
For the mapping of 180 (Ringing) response and 200 (OK) response to the regarding ISUP messages and parameters no
additional procedures beyond the basic call procedures are needed.
To interwork the redirection number at the O-MGCF it can be needed to create placeholder History entries. Such a
History entry has to provide a hi-targeted-to-uri with a placeholder value "unknown@unknown.invalid" a cause-param
and a hi-index as described within table 7.5.4.2.2.1.
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Source SIP header field Source Redirecting number Derived value of parameter field
and component Component
value
In History-Info SIP header Redirecting number
field, hi-targeted-to-uri in hi-
entry before last hi-entry
containing a cause-param
URI parameter, as defined
in IETF RFC 4458
[113].(NOTE 1)
hi-targeted-to-uri CC Nature of address indicator If CC is equal to the country code of the
appropriate global number country where MGCF is located AND
portion of the URI, the next ISUP node is located in the
assumed to be in form same country, then set to "national
"+" CC + NDC + SN (significant) number"
else set to "international number"
CC, NDC, SN Address signals If NOA is "national (significant) number"
then set to NDC + SN.
If NOA is "international number"
then set to CC + NDC + SN
Privacy header field, "history" or APRI "presentation restricted"
priv-value component "session" or
in History-Info header field "header"
as specified in this table Privacy "presentation allowed"
(NOTE 2) header field
absent or
"none"
NOTE 1: If it is SIP URI and doesn‟t contain "user=phone", mapping to redirecting number is impossible,
therefore no need to generate Redirecting number
NOTE 2: It is possible that an entry of the History-Info header field itself is marked as restricted or the whole
History-Info header.
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Source SIP header field Source Component Redirection Derived value of parameter field
and component value Information
Privacy SIP header field "history"or "session" or Redirection Call diverted, all redirection info
and priv-value of hi-entries "header" indicator presentation restricted
of the History-Info SIP for the Privacy SIP
header field. In History-Info header or for any of the
header field last hi-entry two hi-targeted-to-uri
containing a cause-param entries
URI parameter as defined Privacy header field Call diverted
in IETF RFC 4458 [113] absent
and hi-entry before. or
"none"
for the Privacy SIP
header and for both the
two hi-targeted-to-uri
entries
Original Unknown
redirection reason
Cause-param value in the cause-param value Redirecting Redirecting Reason
last hi-targeted-to-uri 404 Reason Unknown/not available
containing a cause-param 302 Unconditional
as defined in IETF RFC 486 User busy
4458 [113] 408 No reply
480 Deflection immediate response
487 Deflection during alerting
503 Mobile subscriber not reachable
Hi-index Redirection number of History entries containing a
counter cause-param with value as listed in the
cause-param row in this table
Table 7.5.4.3.4: Mapping of History-Info header field to ISUP Original Called number
Source SIP header field Source Original called number Derived value of parameter field
and component Component
value
Numbering Plan Indicator "ISDN (Telephony) numbering plan
(Recommendation E.164)"
hi-targeted-to-uri of hi-entry CC Nature of address indicator If CC is equal to the country code of the
st
preceding the 1 hi- country where MGCF is located AND
targeted-to-uri containing a the next ISUP node is located in the
cause-param URI same country, then set to "national
parameter, as defined in (significant) number" else set to
IETF RFC 4458 [113]; the "international number"
global number portion of CC, NDC, SN Address signals If NOA is "national (significant) number"
the URI, is assumed to be then set to
in form NDC + SN.
"+" CC + NDC + SN If NOA is "international number"
(NOTE 1) then set to CC + NDC + SN
priv-value component "history" or APRI "presentation restricted"
in History-Info header field "session" or
of the History-Info header "header"
field entry as defined above Privacy "presentation allowed"
in this table (NOTE 2) header field
absent or
"none"
NOTE 1: If it is SIP URI and doesn‟t contain "user=phone", mapping to Original Called number is impossible,
therefore no need to generate Original Called number
NOTE 2: It is possible that an entry of the History-Info header field itself is marked as restricted or the whole
History-Info header.
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INVITE IAM
History-Info header See table 7.5.4.3.2 Redirecting See table 7.5.4.3.2
field number
History-Info header See table 7.5.4.3.3 Redirection See table 7.5.4.3.3
field Information
cause-param in the cause-param value Redirection Redirecting Reason
last hi-targeted-to-uri 404 Information Unknown/not available
containing a cause- 302 Unconditional
param as defined in 486 User busy
IETF RFC 4458 [113] 408 No reply
480 Deflection immediate response
487 Deflection during alerting
503 Mobile subscriber not reachable
History-Info header See table 7.5.4.3.4 Original Called See table 7.5.4.3.4
field Number
In the ISUP destination exchange of the diverted-to user only the Redirection Number Restriction parameter will be
included into the ACM, CPG, ANM or CON message. Therefore only the mapping of this parameter is shown in the
following table.
Table 7.5.4.3.7: Mapping of ISUP Redirection Number Restriction to History-Info header field
A received CPG shall be mapped to a 180 (Ringing) response if the CPC indicates an Alerting is due to the mapping
rules defined within the basic call.
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Table 7.5.4.3.11: Mapping of ANM 200 (OK) response (to INVITE request)
7.5.6.1 General
The protocol description of the CONF supplementary service is described in 3GPP TS 24.605 [61]. In this subclause the
interworking from the conference event package RFC 4575 [100] to the messages of the PSTN/ISDN CONF
supplementary service is described. Note that an interworking from the PSTN/ISDN to the IMS is out of scope.
When the conference event package option is implemented, and one of the following events occurs at the MGCF:
- a 200 (OK) response is received as a response to an initial INVITE request originated by the MGCF, where the
Contact header field contains an "isfocus" parameter; or
- an ACK message is received which acknowledges a 200 (OK) response to the initial INVITE request, and the
initial INVITE request is originated by the conferencing AS and contains an "isfocus" parameter in the Contact
header field;
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2) the Request URI is set to the Contact address of the conferencing AS;
3) the P-Asserted-Identity header field, the From header field and the Privacy header field are set with the same
value as:
- the P-Asserted-Identity header field, the From header field and the Privacy header field in the initial INVITE
request originated by the MGCF; or
- the P-Asserted-Identity header field, the To header field and the Privacy header field in a 1xx or 2xx response
sent by the MGCF to the initial INVITE request from the conferencing AS.
When a full type of notification is received a check is made of the content. If the changes with respect a previous
version of the notification have not been sent on to the PSTN/ISDN for this session, the MGCF shall perform an ISUP
interaction towards the PSTN/ISDN. If the changes with respect a previous version of the notification have been sent to
the PSTN/ISDN for this session, the MGCF shall not perform an ISUP interaction towards the PSTN/ISDN.
When a partial notification is received then it is assumed that a value of a received notification has changed, so the
MGCF performs an ISUP interaction towards the PSTN/ISDN, as follows:
- Conference established:
Upon the receipt of a conference information document with the <conference-state-type> element active is set to
"true", the MGCF shall send a CPG message to the PSTN/ISDN with a notification "conference established".
- Participant added:
Upon the receipt of a conference information document with the <endpoint-type> and the element status of
endpoint-status-type is set to "connected" and it was not set to "on-hold" before and the Contact URI in the
element entity is not the address of the served PSTN/ISDN participant, the MGCF shall send a CPG message to
the PSTN/ISDN with a notification "other party added".
Upon the receipt of a conference information document with the <endpoint-type> and the element status of
endpoint-status-type is set to "on-hold" and it was set to "connected" before and the Contact URI in the element
entity is the address of the served PSTN/ISDN participant, the MGCF shall send a CPG message to the
PSTN/ISDN with a notification "isolated".
Upon the receipt of a conference information document with the <endpoint-type> and the element status of
endpoint-status-type is set to "on-hold" and it was set to "connected" before and the Contact URI in the element
entity is not the address of the served PSTN/ISDN participant, the MGCF shall send a CPG message to the
PSTN/ISDN with a notification "other party isolated".
Upon the receipt of a conference information document with the <endpoint-type> and the element status of
endpoint-status-type is set to "connected" and it was set to "on-hold" before and the Contact URI in the element
entity is the address of the served PSTN/ISDN participant, the MGCF shall send a CPG message to the
PSTN/ISDN with a notification "reattached".
Upon the receipt of a conference information document with the <endpoint-type> and the element status of
endpoint-status-type is set to "connected" and it was set to "on-hold" before and the Contact URI in the element
entity is not the address of the served PSTN/ISDN participant, the MGCF shall send a CPG message to the
PSTN/ISDN with a notification "other party reattached".
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Upon the receipt of a conference information document with the <endpoint-type> and the element status of
endpoint-status-type is set to "disconnected" and the element joining-method of joining-type is not set to "focus-
owner", the MGCF shall send a CPG message to the PSTN/ISDN with a notification "other party disconnected".
The mapping of Anonymous Communication Rejection supplementary service with Anonymous Call Rejection
PSTN/ISDN Supplementary Service is described in subclause 7.4.23.
The mapping for Communication Barring is in accordance with the basic call procedures as described in subclauses
7.2.3.1.8 and 7.2.3.2.12.
7.5.9.0 General
The protocol specification of the Malicious Communication Identification supplementary service is described in 3GPP
TS 24.616 [102]. The XML MCID body used in related SIP messages is also specified in 3GPP TS 24.616 [102].
7.5.9.1.0 General
If the MGCF supports the interworking of the MCID service the O-MGCF shall map a SIP INFO request containing a
XML mcid body with MCID XML Request schema to an Identification Request (IDR) message and an Identification
response (IRS) message to a SIP INFO request containing a XML mcid body with MCID XML Response schema in
accordance with table 7.5.9.1.1.
The IDR message shall be generated upon reception of the SIP INFO request containing a XML mcid body with MCID
XML Request schema.
The SIP INFO request containing a XML mcid body with MCID XML Response schema shall be generated upon
reception of the IRS message.
Table 7.5.9.1.1 Mapping between ISUP IDR and IRS and SIP messages
7.5.9.1.1 Interworking of the MCID XML Request schema with the ISUP MCID request
indicators
If the MGCF supports the interworking of the MCID service O-MGCF shall map the codes in the MCID XML elements
to MCID request indicator and holding indicator parameter fields in accordance with Table 7.5.9.1.1.1.
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Table 7.5.9.1.1.1 Mapping between ISUP MCID request and holding indicators and MCID XML
elements
7.5.9.1.2 Interworking of the ISUP MCID response indicators with the MCID XML
Response schema
If the MGCF supports the interworking of the MCID service the O-MGCF shall map the codes in the MCID response
indicator and hold provided indicator parameter field to the MCID XML elements in accordance with Table 7.5.9.1.2.1.
Table 7.5.9.1.2.1 Mapping between ISUP MCID response and hold provided indicators and MCID XML
elements
7.5.9.1.3 Interworking of the ISUP Calling Party Number in an Identification Response with
the OrigPartyIdentity within the MCID XML Response schema
If the O-MGCF supports the interworking of the MCID service and receives an ISUP Identification Response
containing a Calling Party Number with the screening indicator set to "user provided, verified and passed" or "network
provided", the O-MGCF shall map the Calling Party Number to the MCID XML Response schema OrigPartyIdentity
element applying the same mapping procedures as described in Table 14 for the mapping into the SIP P-Asserted-
Identity header and shall map the Calling Party Number APRI to the MCID XML Response schema
OrigPartyPresentationRestriction element. If the Calling Party Number APRI has a value of "presentation allowed" then
the MCID XML Response schema OrigPartyPresentationRestriction element shall be set to "false", otherwise it shall be
set to "true".
7.5.9.1.4 Interworking of the ISUP Generic Number in an Identification Response with the
GenericNumber within the MCID XML Response schema
If the O-MGCF supports the interworking of the MCID service and receives an ISUP Identification Response
containing a Generic Number with the screening indicator set to "user provided, verified and passed", or "user
provided, not verified", or "network provided", the O-MGCF shall map the Generic Number to the MCID XML
Response schema GenericNumber element applying the same mapping procedures as described in Table 13 for the
mapping into the SIP From header and shall map the Generic Number APRI to the MCID XML Response schema
GenericNumberPresentationRestriction element. If the Generic Number APRI has a value of "presentation allowed"
then the MCID XML Response schema GenericNumberPresentationRestriction element shall be set to "false",
otherwise it shall be set to "true".
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7.5.9.2.1 General
If the MGCF supports the interworking of the MCID service the I-MGCF shall map an Identification Request (IDR)
message to a SIP INFO request containing a XML mcid body with MCID XML Request schema and a SIP INFO
request containing a XML mcid body with MCID XML Response schema to an Identification response (IRS) message
in accordance with table 7.5.9.1.1.
If the received MCID XML Response schema contains an OrigPartyIdentity element, the I-MGCF shall map the
OrigPartyIdentity to the Calling Party Number within the IRS applying the same mapping procedure as described in
Table 5 for the mapping from the SIP P-Asserted-Identity header, with the exception that the I-MGCF shall map the
MCID XML Response schema OrigPartyPresentationRestriction element to the Calling Party Number APRI. If the
MCID XML Response schema OrigPartyPresentationRestriction element has the value "true", the Calling Party
Number APRI shall be set to "presentation restricted ", and otherwise the Calling Party Number APRI shall be set to
"presentation allowed".
If the received MCID XML Response schema contains an GenericNumber element, the I-MGCF shall map the
GenericNumber to the Generic Number within the IRS applying the same mapping procedure as described in Table 6
for the mapping from the From header, with the exception that the I-MGCF shall map the MCID XML Response
schema GenericNumber PresentationRestriction element to the Generic Number APRI. If the MCID XML Response
schema GenericNumberPresentationRestriction element has the value "true", the Generic Number APRI shall be set to
"presentation restricted ", and otherwise the Generic Number APRI shall be set to "presentation allowed".
7.5.10.0 General
The protocol specification of the Closed User Group supplementary service is described in 3GPP TS 24.654 [101].
If the MGCF supports the interworking of CUG supplementary service, the I-MGCF shall interwork the CUG XML
schema with the ISUP Closed user group interlock code parameter and the Closed user group call indicator of the
optional forward call indicator parameter in accordance with tables 7.5.10.1.2 and 7.5.10.1.3.
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Table 7.5.10.1.2: Mapping of the SIP XML CUG Element to the ISUP closed usergroup interlock code
parameter
CUG XML Element derived value of ISUP Closed user group Source component value
parameter field interlock code Parameter
networkIndicator networkIdentityType "Network Identity" Octet 1 & Octet 2 including 4 binary
= 4 Digit decimal coded digits derived from XML Network
value Identity in decimal format
cugInterlockBinaryCode sixteenbitType = 16 "Binary Code" Octet 3 & Octet 4 including a 16 bit
bit coded value Binary Code derived from the XML
Binary Code
Table 7.5.10.1.3 Mapping of the SIP XML CUG Element to the ISUP closed user group call indicator
included in the optional Forward Call Indicator Parameter
CUG XML Element derived value of ISUP "Optional Forward Source component value
parameter field Call Indicator" Parameter
cugCommunicationIndicator Type=00 "closed user group call" non-CUG call
Type=01 indicator spare
Type=10 closed user group call, outgoing
access allowed
Type=11 closed user group call, outgoing
access not allowed
If the I-MGCF supports the interworking of CUG supplementary service, then if an INVITE request with the MIME
including a cug XML element is received and the terminating network is not supporting CUG, the I-MGCF shall behave
as shown in table 7.5.10.1.4.
Table 7.5.10.1.4: Action at the I-MGCF with a PSTN/ISDN network without CUG capability
If the MGCF supports the interworking of CUG supplementary service, the MGCF shall interwork the CUG XML
schema with the ISUP Closed user group interlock code parameter and the Closed user group call indicator of the
optional forward call indicator parameter in accordance with tables 7.5.10.2.2 and table 7.5.10.2.3.
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Table 7.5.10.2.2 Mapping of the ISUP closed usergroup interloccode to SIP XML CUG element
ISUP Closed user group Source component value CUG XML Element derived value of parameter
interlock code Parameter field
"Network Identity" Octet 1 & Octet 2 including networkIndicator networkIdentityType = 4 Digit
4 binary coded digits decimal value derived from
Network Identity
"Binary Code" Octet 3 & Octet 4 including cugInterlockBinaryCode sixteenbitType = 16 bit coded
a 16 bit Binary Code value derived from Binary Code
Table 7.5.10.2.3: Mapping of the ISUP Closed user group call indicator to SIP XML CUG element
ISUP Optional Forward Call Source component value CUG XML Element derived value of
Indicator Parameter parameter field
closed user group call non-CUG call cugCommunicationIndicator Type=00
indicator spare Type=01
closed user group call, outgoing Type=10
access allowed
closed user group call, outgoing Type=11
access not allowed
If the MGCF supports the interworking of CUG supplementary service, but the IMS is not supporting CUG, the
procedures described in ITU Q.735.1 [42] shall apply if an INVITE request with the MIME body including a cug XML
element is sent and the O-MGCF supports CUG supplementary service.
7.5.11 CCBS/CCNR
7.5.11.0 General
The protocol specification of the Completion of Communications to Busy Subscriber and Completion of
Communications by No Reply supplementary services is described in 3GPP TS 24.642 [112].
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If the I-MGCF supports the interworking of CCBS/CCNR supplementary services, the I-MGCF shall map between the
SIP and TCAP messages in accordance with table 7.5.11.1.2 and table 7.5.11.1.3.
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If the O-MGCF supports the interworking of CCBS/CCNR supplementary services, the O-MGCF shall map between
the TCAP and SIP messages in accordance with table 7.5.11.2.2 and table 7.5.11.2.3.
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7.5.12.0 General
The protocol specification of the Communication Waiting supplementary service is described in 3GPP TS 24.615 [111].
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Transcoding
voice
AMR
codec
RTP
UDP TNL
IP
L2
L1 L1
Mb IM-MGW
Figure 31/1: IM CN subsystem to bearer independent CS network user plane protocol stack
Transcoder-less
user plane
interworking
RTP
Nb FP
UDP
IP L3
L2 L2
L1 L1
Mb IM-MGW Nb
Figure 31/2: IM CN subsystem to bearer independent CS network user plane protocol stack (optional
in the event the codecs on both sides are the same)
If no transcoder is inserted, the IM-MGW shall interwork the following procedures between the Nb and Mb interfaces.
8.1.1.1 Initialisation
There is no need to interwork initialisation procedures between Nb and Mb interfaces see 3GPP TS 29.415 [26].
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Interworking of rate control procedures at an IM-MGW between an Mb interface and a corresponding Nb interface only
applies when the IM-MGW bridges compatible codec configurations between the interfaces without applying a
transcoding function. An IM-MGW receiving a CMR from an Mb interface shall initiate the IuFP rate control procedure
on the corresponding Nb interface. An IM-MGW receiving a rate control request on an Nb interface shall adjust the
CMR field of outgoing speech frames on the corresponding Mb interface.
Tables 24a and 24b provide the mapping between Mb and Nb interfaces.
Mb - Qbit Mb - FT Nb - FQC
1 x 0
0 x 1
Nb - FQC Mb - Qbit Mb – FT
0 1 NC
1 0 NO_DATA
2 0 NC
8.1.1.5 Framing
Even when the IM-MGW bridges compatible codec configurations between the Nb and Mb interfaces, the IM-MGW
shall perform translation between the frame formats defined for the two interfaces, since all codec configurations have
different framing procedures for the two interfaces. The framing details for Nb are defined in 3GPP TS 26.102 [50] and
3GPP TS 29.415 [26], although they do not describe the framing for ITU-T codecs other than G.711. The framing
details for Mb are defined in RFC 3267 [23], RFC 3550 [51], RFC 3551 [52] and RFC 3555 [53].
8.1.1.6 Transcoding
Transcoding at the IM-MGW is avoided when the IM-MGW bridges compatible codec configurations between the Nb
and Mb interfaces. Otherwise transcoding is necessary, which eliminates the need to interwork other user plane
procedures between the interfaces.
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When the IuFP frame numbers are based on time and if the IM-MGW bridges compatible codec configurations between
the Nb and Mb interfaces, it shall either correct jitter before forwarding PDUs or interwork the RTP timestamp (see
RFC 3550 [51]) with the IuFP Frame Number (see 3GPP TS 29.415 [26]) so that both the RTP timestamp and IuFP
frame number similarly reflect the nominal sampling instant of the user data in the packet.
The RTP sequence number (see RFC 3550 [51]) is handled independently on Mb, i.e. it is not interworked with the
IuFP Frame Number (see 3GPP TS 29.415 [26]).
Transcoding
G.711
AMR PCM
Coding
RTP
TDM
UDP CS
IP Bearer
Channel
L2
L1 L1
Mb IM-MGW
It shall be possible for the IM CN subsystem to interwork with the CS networks (e.g. PSTN, ISDN or a CS domain of a
PLMN) by supporting AMR to G.711 transcoding (see ITU-T Recommendation G.711 [1]) in the IM-MGW. The IM-
MGW may also perform transcoding between AMR and other codec types supported by CS networks.
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The IETF Differentiated Services architecture (see RFC 2475 [22]) shall be used to provide QoS for the external bearer
service.
When detecting DTMF digits arriving from the CS side, the MGW shall comply with TS 23.014 [103] (by checking that
a valid digit with minimum duration and minimum gap has been received) before initiating an RTP Telephony Event to
the IMS interface.
When sending DTMF towards the IMS side according to the IETF RFC 4733 [105] RTP Payload format, the MGW
shall comply with the DTMF encoding requirements of Annex G.2 of 3GPP TS 26.114 [104], in particular the
minimum duration of 65ms shall be ensured. It is optional if the RTP Telephony Event is sent as a number of "RTP
Events" with interim durations (e.g. every 20ms or 40ms in line with the speech packetisation time) or as a single "RTP
Event" with the at least 65ms duration.
9.1 Overview
The MGCF shall control the functions of the IM-MGW, which are used to provide the connection between media
streams of an IP based transport network and bearer channels from a CS network.
The MGCF shall interact with the IM-MGW across the Mn reference point. The MGCF shall terminate the signalling
across the Mn interface towards the IM-MGW and the IM-MGW shall terminate the signalling from the MGCF.
The signalling interface across the Mn reference point shall be defined in accordance with ITU-T Recommendation
H.248.1 [2] and shall conform to 3GPP specific extensions as detailed in 3GPP TS 29.332 [15].
The present specification describes Mn signalling procedures and their interaction with BICC/ISUP and SIP signalling
in the control plane, and with user plane procedures. 3GPP TS 29.332 [15] maps these signalling procedures to H.248
messages and defines the required packages and parameters.
The SIP signalling occurring at the MGCF is described in 3GPP TS 24.229 [9].
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Release 8 156 3GPP TS 29.163 V8.16.0 (2011-09)
MGCF
CS
NETWORK
T1 T2
C1
IM-MGW
- Shall send the appropriate remote codec(s), the remote UDP port and the remote IP address to the IM-MGW.
The remote UDP port and IP address refer to the destination of user plane data sent towards the IM CN
subsystem. The remote codec(s) are the codec(s) the IM-MGW may select for user plane data sent towards the
IM CN subsystem.
- Shall indicate to the IM-MGW the appropriate local codec(s) and request a local IP address and UDP port. The
local IP address and UDP port are used by the IM-MGW to receive user plane data from the IM CN subsystem.
The local codec(s) are the codec(s) the IM-MGW may select to receive user plane data from the IM CN
subsystem.
- If DTMF support together with speech support is required, the reserve value indicator shall be set to "true".
The IM-MGW
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- Shall reply to the MGCF with the selected local codec(s) and the selected remote codec(s) and the selected local
UDP port and IP address.
The MCGF shall send the local codec(s), UDP port and IP address to the IMS in the Session Progress (signal 9 in figure
34).
- The appropriate remote codec(s), the remote UDP port and the remote IP address.
- If DTMF support together with speech support is required, the reserve value indicator shall be set to "true".
- Reply to the MGCF with the selected local codec(s)if the MGCF supplied local codec(s),
- Update the codec reservation and remote IP address and remote UDP port in accordance with the received
information.
The MGCF shall include the selected codec(s) and UDP port and IP address in an 200 OK (PRACK) (signal 11 in
figure 34) sent back to the IMS.
9.2.2.1.5 Through-connection
During the Prepare Bearer and Establish Bearer procedures, the MGCF shall either use the Change Through-Connection
procedure to request the IM-MGW to backward through-connect the BICC terminations, or the MGCF shall use this
procedure to both-way through-connect the BICC termination already on this stage (signal 7 in figure 34). During the
Reserve IMS Connection Point procedure, the MGCF shall use the Change IMS Through-Connection procedure to
request the IM-MGW to backward through-connect the IMS termination (signal 3 in figure 34).
When the MGCF receives the BICC:ANM answer indication, it shall request the IM-MGW to both-way through-
connect the termination using the Change Through-Connection or Change IMS Through-Connection procedures (signal
22 in figure 34), unless those terminations are already both-way through-connected.
NOTE: As an implementation option the MGCF may also decide for example to only release the resources in the
IM-MGW that caused the failure, possibly select a new IM-MGW for the connection and continue the
call establishment using new resources in the selected IM-MGW but such handling is outside of the scope
of the present document.
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Release 8 159 3GPP TS 29.163 V8.16.0 (2011-09)
MGCF IM-MGW
1. SIP: INVITE
if SIP 9. SIP:183
Session Progress
100 rel 5. BICC: IAM
is
used 6. BICC: APM
Bearer Establishment
Only if
SIP 14. SIP: UPDATE
Precon- 15. SIP :200 OK Only if BICC
ditions (UPDATE) continuity check
16. BICC: COT
are and/or SIP
used Preconditions
17. BICC: ACM are used
18. SIP :180 Ringing
Optional
19. SIP: PRACK
Figure 34: Basic IM CN Subsystem originating session, BICC forward bearer establishment
(message sequence chart)
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- Shall send the appropriate remote codec(s), the remote UDP port and the remote IP address to the IM-MGW.
The remote UDP port and IP address refer to the destination of user plane data sent towards the IM CN
subsystem. The remote codec(s) are the codec(s) the IM-MGW may select for user plane data sent towards the
IM CN subsystem.
- Shall indicate to the IM-MGW the appropriate local codec(s)and request a local IP address and UDP port. The
local UDP port and IP address are used by the IM-MGW to receive user plane data from the IM CN subsystem.
The local codec(s) are the codec(s) the IM-MGW may select to receive user plane data from the IM CN
subsystem.
- If DTMF support together with speech support is required, the reserve value indicator shall be set to "true".
- Reply to the MGCF with the selected local codec(s) and the selected remote codec(s) and the selected local UDP
port and IP address.
The MCGF shall send the local codec(s), UDP port and IP address to the IMS in the Session Progress (signal 5 in figure
35).
- the appropriate remote codec(s), the remote UDP port and the remote IP address.
- optionally if DTMF support together with speech support is required, the reserve value indicator shall be set to
"true".
- Reply to the MGCF with the selected local codec(s), if the MGCF supplied local codec(s).
- Update the codec reservation and remote IP address and remote UDP port in accordance with the received
information.
The MGCF shall include the selected codec(s), IP address and UDP port in an 200 OK (PRACK) (signal 12 in figure
35) sent back to the IMS.
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9.2.2.2.5 Through-connection
During the Prepare Bearer procedure, the MGCF shall either use the Change Through-Connection procedure to request
the IM-MGW to backward through-connect the BICC termination, or the MGCF shall use this procedure to both-way
through-connect the BICC termination already on this stage (signal 6 in figure 35). During the Reserve IMS Connection
Point procedure, the MGCF shall use the Change IMS Through-Connection procedure to request the IM-MGW to
backward through-connect the IMS termination (signal 3 in figure 35).
When the MGCF receives the BICC:ANM answer indication, it shall request the IM-MGW to both-way through-
connect the terminations using the Change Through-Connection or Change IMS Through-Connection procedures
(signal 21 in figure 35), unless those terminations are already both-way through-connected.
NOTE: As an implementation option the MGCF may also decide for example to only release the resources in the
IM-MGW that caused the failure, possibly select a new IM-MGW for the connection and continue the
call establishment using new resources in the selected IM-MGW but such handling is outside of the scope
of the present document.
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Release 8 162 3GPP TS 29.163 V8.16.0 (2011-09)
MGCF IM-MGW
1. SIP: INVITE
2. SIP: 100 Trying
8. BICC: IAM
Bearer Establishment
9. SIP: PRACK
10. H.248: MOD.req [Context ID = C1,Termination ID = T1]
if SIP
100 rel Configure IMS Resources
11. H.248: MOD.resp [Context ID = C1, Termination ID = T1 ]
is used
12. SIP :200
OK (PRACK)
Figure 35: Basic IM CN Subsystem originating session, BICC backward bearer establishment
(message sequence chart)
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Release 8 163 3GPP TS 29.163 V8.16.0 (2011-09)
9.2.2.3 ISUP
- shall send the appropriate remote codec(s), the remote UDP port and the remote IP address to the IM-MGW. The
remote UDP port and IP address refer to the destination of user plane data sent towards the IM CN subsystem.
The remote codec(s) are the codec(s) the IM-MGW may select for user plane data sent towards the IM CN
subsystem.
- shall indicate to the IM-MGW the appropriate local codec(s) and request a local IP address and UDP port. The
local IP address and UDP port are used by the IM-MGW to receive user plane data from the IM CN subsystem.
The local codec(s) are the codec(s) the IM-MGW may select to receive user plane data from the IM CN
subsystem.
- If DTMF support together with speech support is required, the reserve value indicator shall be set to "true".
- reply to the MGCF with the selected local codec(s) and the selected remote codec(s) and the selected local UDP
port and IP address.
The MCGF shall send selected local codec(s) and the selected remote codec and the selected local UDP port and IP
address to the IMS in the Session Progress (signal 5 in figure 36).
- the appropriate remote codec(s), the remote UDP port and the remote IP address.
- If DTMF support together with speech support is required, the reserve value indicator shall be set to "true".
- reply to the MGCF with the selected local codec(s), if the MGCF supplied local codec(s).
- update the codec reservation and remote IP address and UDP port in accordance with the received information.
The MGCF shall include the selected codec(s) UDP port and IP address in 200 OK (PRACK) (signal 12 in figure 36)
sent back to the IMS.
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Release 8 164 3GPP TS 29.163 V8.16.0 (2011-09)
9.2.2.3.5 Through-connection
During the Reserve TDM Circuit and Reserve IMS Connection Point procedures, the MGCF shall either use the Change
TDM Through-Connection procedure to request the IM-MGW to backward through-connect the termination, or the
MGCF shall use this procedure to both-way through-connect the TDM termination already on this stage (signal 6 in
figure 36). During the Reserve IMS connection Point procedure, the MGCF shall use the Change IMS through-
connection procedure to request the IM-MGW to backward through-connect the IMS termination (signal 3 in figure 36).
When the MGCF receives the ISUP:ANM answer indication, it shall request the IM-MGW to both-way through-
connect the terminations using the Change IMS Through-Connection or Change TDM Through-Connection procedures
(signal 21 in figure 36), unless those terminations are already both-way through-connected.
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MGCF IM-MGW
1. SIP: INVITE
2. SIP: 100 Trying
8. ISUP: IAM
9. SIP:
PRACK
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Release 8 166 3GPP TS 29.163 V8.16.0 (2011-09)
Figure 36: Basic IM CN Subsystem originating session, ISUP (message sequence chart)
The MGCF may also indicate that the IP interface type is for MboIP.
The IM-MGW shall reply to the MGCF with the selected local codec(s) and the selected local IP address and UDP port.
The MGCF shall send this information in the INVITE (signal 4 in figure 37) to the IM CN subsystem.
- The MGCF shall indicate the remote IP address and UDP port, i.e. the destination IP address and UDP port for
data sent in the user plane towards the IM CN subsystem,
- The MGCF shall indicate the remote codec(s), i.e. the speech codec(s) for data sent in the user plane towards the
IM CN subsystem.
- The MGCF may indicate the local codec(s) and the local IP address and UDP port. The MGCF shall indicate the
local codec(s) if a change is required.
- IF DTMF support together with speech support is required, the reserve value indicator shall be set to "true".
The IM-MGW shall reply with the selected remote codec(s) and reserve resources for these codec(s). If local codec(s)
were received, the IM-MGW shall also reply with the selected local codec(s) and reserve the corresponding resources.
If the selected local codec(s) differ from the codec(s) received in the SDP of signal 6 in figure 37 (if any), the MGCF
shall send the local reserved codec(s), and the local IP address and UDP port in the PRACK (signal 9 in figure 37) to
the IMS.
If the selected local codec(s) differ from the codec(s) received in the SDP of signal 23 in figure 37 (if any), the MGCF
shall send the local reserved codec(s), and the local IP address and UDP port in an re-INVITE or UPDATE (not
depicted in figure 37) to the IMS.
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Release 8 167 3GPP TS 29.163 V8.16.0 (2011-09)
provide the IM-MGW with the bearer characteristics that was received from the preceding node in the IAM. After the
IM-MGW has replied with the bearer address and the binding reference, the MGCF provides the APM message (signals
13 in figure 37) to the preceding node. The MGCF may also provide the IM-MGW-id in the APM message.
- the MGCF receives the first 180 Ringing message, unless this message or some previous SIP provisional
response contained a P-Early-Media header that authorizes early media and the MGCF supports the P-Early-
Media header as a network option.
9.2.3.1.7 Through-Connection
During the Prepare Bearer procedure, the MGCF shall either use the Change Through-Connection procedure to request
the IM-MGW to backward through-connect the BICC termination, or the MGCF shall use this procedure to both-way
through-connect the BICC termination already on this stage (signals 11 and 12 in figure 37). During the Reserve IMS
Connection Point procedure, the MGCF shall use the Change IMS Through-Connection procedure to request the IM-
MGW to backward through-connect the IMS termination (signals 2 and 3 in figure 37).
When the MGCF receives the SIP 200 OK(INVITE) (signal 23 in figure 37), it requests the IM-MGW to both-way
through-connect the terminations using the Change IMS Through-Connection or Change Through-Connection
procedures (signals 28 and 29 in figure 37), unless those terminations are already both-way through-connected.
NOTE: As an implementation option the MGCF may also decide for example to only release the resources in the
IM-MGW that caused the failure, possibly select a new IM-MGW for the connection and continue the
call establishment using new resources in the selected IM-MGW but such handling is outside of the scope
of the present document.
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Release 8 168 3GPP TS 29.163 V8.16.0 (2011-09)
IM-MGW MGCF
1. BICC: IAM
6. SIP:183
Session
Progress
7. H.248: MOD.req Context ID = C1,Termination ID = T1]
if SIP
Configure IMS Resources
Precon-
8. H.248: MOD.resp [Context ID = C1, Termination ID = T1 ] ditions
9. SIP:
PRACK and/or
100 rel
are used
10. SIP :200
OK
(PRACK)
Bearer Establishment
17. SIP
:180
Ringing
18. SIP:
19. BICC: ACM PRACK Optional
Figure 37/1: Basic CS Network Originating Session, Forward Bearer Establishment (message
sequence chart)
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Release 8 169 3GPP TS 29.163 V8.16.0 (2011-09)
IM-MGW MGCF
Stop If tone
Tone was
25. H.248: MOD.resp [Context ID = C1, Termination ID = T2]
inserted
The MGCF may also indicate that the IP interface type is for MboIP.
The IM-MGW shall reply to the MGCF with the selected local codec(s) and the selected local IP address and UDP port.
The MGCF shall send this information in the INVITE (signal 6 in figure 38) to the IM CN subsystem.
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Release 8 170 3GPP TS 29.163 V8.16.0 (2011-09)
- The MGCF shall indicate the remote IP address and UDP port, i.e. the destination IP address and UDP port for
data sent in the user plane towards the IM CN subsystem
- The MGCF shall indicate the remote codec(s), i.e. the speech codec(s) for data sent in the user plane towards the
IM CN subsystem.
- The MGCF may indicate the local codec(s) and the local IP address and UDP port. The MGCF shall indicate the
local codec(s) if a change is required.
- If DTMF support together with speech support is required, the reserve value indicator shall be set to "true".
The IM-MGW shall reply with the selected remote codec(s) and reserve resources for this codec. If local codec(s) were
received, the IM-MGW shall also reply with the selected local codec(s) and reserve the corresponding resources.
If the selected local codec(s) differ from the codec(s) received in the SDP of signal 8 in figure 38 (if any), the MGCF
shall send the reserved speech codec(s), and the local IP address and UDP port in the PRACK (signal 11 in figure 38) to
the IMS.
If the selected local codec(s) differ from the codec(s) received in the SDP of signal 22 in figure 38 (if any), the MGCF
shall send the local reserved codec(s), and the local IP address and UDP port in an re-INVITE or UPDATE (not
depicted in figure 38) to the IMS.
- the MGCF receives the first 180 Ringing message, unless this message or some previous SIP provisional
response contained a P-Early-Media header that authorizes early media and the MGCF supports the P-Early-
Media header as a network option.
9.2.3.2.7 Through-Connection
During the Establish Bearer procedure, the MGCF shall either use the Change Through-Connection procedure to
request the IM-MGW to backward through-connect the BICC termination, or the MGCF shall use this procedure to
both-way through-connect the BICC termination already on this stage (signals 2 and 3 in figure 38). During the Reserve
IMS Connection Point procedure, the MGCF shall use the Change IMS Through-Connection procedure to request the
IM-MGW to backward through-connect the IMS termination (signals 4 and 5 in figure 38).
When the MGCF receives the SIP 200 OK(INVITE) (signal 22 in figure 38), it shall request the IM-MGW to both-way
through-connect the bearer using the Change IMS Through-Connection or Change Through-Connection procedure
(signals 25 and 26 in figure 38), unless those terminations are already both-way through-connected.
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Release 8 171 3GPP TS 29.163 V8.16.0 (2011-09)
NOTE: As an implementation option the MGCF may also decide for example to only release the resources in the
IM-MGW that caused the failure, possibly select a new IM-MGW for the connection and continue the
call establishment using new resources in the selected IM-MGW but such handling is outside of the scope
of the present document.
3GPP
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IM-MGW MGCF
1. BICC: IAM
Bearer Establishment
4. H.248: ADD.req [Context ID = C1, Termination ID=?] Reserve IMS Connection Point,
Change IMSThrough-
5. H.248: ADD.resp [Context ID = C1, Termination ID = T1] Connection = backward
6. SIP: INVITE
7. SIP: 100 Trying
8. SIP:183
Session
Progress
9. H.248: MOD.req [Context ID = C1,Termination ID = T1]
if SIP
Configure IMS Resources
Precon-
10.H.248: MOD.resp [Context ID = C1, Termination ID = T1 ] 11. SIP: ditions
PRACK
and/or
12. SIP 100 rel
:200 OK are used
(PRACK)
Figure 38/1: Basic CS Network Originating Session, BICC Backward Bearer Establishment (message
sequence chart)
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IM-MGW MGCF
Figure 38/2: Basic CS Network Originating Session, BICC Backward Bearer Establishment
(message sequence chart continue)
9.2.3.3 ISUP
The MGCF may also indicate that the IP interface type is for MboIP.
The IM-MGW shall reply to the MGCF with the selected local codec(s) and the selected local IP address and UDP port.
The MGCF shall send this information in the INVITE (signal 6 in figure 39) to the IM CN subsystem.
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- The MGCF shall indicate the remote IP address and UDP port, i.e. the destination IP address and UDP port for
data sent in the user plane towards the IM CN subsystem.
- The MGCF shall indicate the remote codec(s), i.e. the speech codec(s) for data sent in the user plane towards the
IM CN subsystem.
- The MGCF may indicate the local codec(s) and the local IP address and UDP port. The MGCF shall indicate the
local codec(s) if a change is required.
- If DTMF support together with speech support is required, the reserve value indicator shall be set to "true".
The IM-MGW shall reply with the selected remote codec(s) and reserve resources for these codec(s). If local codec(s)
were received, the IM-MGW shall also reply with the selected local codec(s) and reserve the corresponding resources.
If the selected local codec(s) differ from the codec(s) received in the SDP of signal 8 in figure 39 (if any), the MGCF
shall send the reserved speech codec(s), and the local IP address and UDP port in the PRACK (signal 11 in figure 39) to
the IMS.
If the selected local codec(s) differ from the codec(s) received in the SDP of signal 22 in figure 39 (if any), the MGCF
shall send the local reserved codec(s), and the local IP address and UDP port in an re-INVITE or UPDATE (not
depicted in figure 39) to the IMS.
- the MGCF receives the first 180 Ringing message, unless this message or some previous SIP provisional
response contained a P-Early-Media header that authorizes early media and the MGCF supports the P-Early-
Media header as a network option.
9.2.3.3.7 Through-Connection
Within the Reserve TDM Circuit procedure, the MGCF shall either use the Change TDM Through-Connection
procedure to request the IM-MGW to backward through-connect the TDM termination, or the MGCF shall use this
procedure to both-way through-connect the TDM termination already on this stage (signals 2 and 3 in figure 39).
During the Reserve IMS Connection Point procedure, the MGCF shall use the Change IMS Through-Connection
procedure to request the IM-MGW to backward through-connect the IMS termination (signals 4 and 5 in figure 39).
When the MGCF receives the SIP 200 OK(INVITE) message, it shall request the IM-MGW to both-way through-
connect the terminations using the Change IMS Through-Connection or Change TDM Through-Connection procedure
(signals 25 and 26 in figure 39), unless those terminations are already both-way through-connected.
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IM-MGW MGCF
1. ISUP: IAM
6. SIP: INVITE
8. SIP:183
Session
Progress
9. H.248: MOD.req [Context ID = C1,Termination ID = T1]
if SIP
Configure IMS Resources Precon-
10.H.248: MOD.resp [Context ID = C1, Termination ID = T1 ] 11. SIP: ditions
PRACK and/or
100 rel
are used
12. SIP
:200 OK
(PRACK)
if
continuity 13. ISUP: COT
check is 14. SIP:
used UPDATE if SIP
Precon-
ditions
15. SIP and/or
:200 OK 100 rel
(UPDATE) are used
17. SIP:
18. ISUP: ACM PRACK
20. H.248: MOD.resp [Context ID = C1, Termination ID = T2] Send TDM Tone
Optional
Figure 39/1: Basic CS Network Originating Session, ISUP (message sequence chart)
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IM-MGW MGCF
Stop TDM
If tone
23. H.248: MOD.req [Context ID=C1, Termination ID = T2] Tone ,
was
Activate
inserted or
24. H.248: MOD.resp [Context ID = C1, Termination ID = T2] TDM Voice
for voive
Processing
processing
Function
Figure 39/2: Basic CS Network Originating Session, ISUP (message sequence chart continue)
- If the MGCF receives a SIP provisional response containing a P-Early-Media header that authorizes early media
and if the MGCF supports the P-Early-Media header as a network option, the MGCF shall provide the remote IP
address and UDP port, and the remote codec selected from the received SDP and local configuration data,
corresponding to the SIP dialogue of the SIP provisional response containing a P-Early-Media header that
authorizes early media. The IM-MGW may be configured to use the remote IP address and port information as
source filter for incoming packages to prevent that early media from other early SIP dialogues interfere. The
MGCF may also provide an IP address and port source filter that disallows early media from other early
dialogues without an early media authorization in order to prevent that such unauthorized early media interfere
with the authorized early media. (NOTE 1, NOTE 2)
- If the MGCF did not receive any SIP provisional response containing a P-Early-Media header that authorizes
early media or if the MGCF does not support the P-Early-Media header as a network option, the MGCF may
compare the selected local codecs of the different dialogues (which the MGCF selects due to the received SDP
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answer and local configuration data). If different local codecs are selected for the different dialogues, the MGCF
may include all these codecs in the "local IMS resources", and set the "reserve value" to indicate that resources
for all these codecs shall be reserved. Alternatively, the MGCF may only include the codecs received in the last
SDP in the "local IMS resources".
- If the MGCF did not receive any SIP provisional response containing a P-Early-Media header that authorizes
early media or if the MGCF does not support the P-Early-Media header as a network option, the MGCF may
update the "remote IMS resources" with the information received in the latest SDP. The MGCF should provide
the remote IP address and UDP port, and the remote codec selected from the received SDP and local
configuration data. (NOTE 3)
NOTE 1: The O-MGCF can use the P-Early-Media header (IETF RFC 5009 [89]) to determine whether the media
associated with a forked dialog is authorized and thus eligible for a through connection. In the presence of
early media for multiple dialogs due to forking, if the IM-MGW is able to identify the media associated
with a dialog (i.e., if symmetric RTP is used by the peer and the IM-MGW can use the remote SDP
information to determine the source of the media), then the O-MGCF/IM-MGW can selectively establish
a through-connection for an authorized early media flow.
NOTE 2: The behaviour of an O-MGCF supporting the P-Early-Media header as a network option upon the
possible reception of SIP provisional responses containing P-Early-Media headers that authorize early
media for several early dialogues is left unspecified.
NOTE 3: The behaviour in the third bullet is beneficial if forking is applied in a sequential manner.
- If the remote IMS resources configured at the IM-MGW do not match the remote resources selected for the
established dialogue of the final response, the MGCF shall provide the remote IP address and UDP port from the
latest received SDP of this established dialogue, and the remote codec selected from the latest received SDP of
this established dialogue and local configuration data within the "remote IMS resources".
- If the local IMS resources configured at the IM-MGW contain more codecs than selected for the established
dialogue of the final response, the MGCF should update the "local IMS resources" with the selected local codec
derived from the latest SDP of this established dialogue and local configuration data. The "reserve value" may be
cleared unless it is required for DTMF.
- The IM-MGW may be configured to use the remote IP address and port information as source filter for incoming
packages to prevent that early media from other early SIP dialogues interfere with the media of the established
dialogue. The MGCF may also provide an IP address and port source filter that disallows early media from other
early dialogues in order to prevent that such early media interfere with the media of the established dialogue. If
the MGCF has provided a source filter selecting media of another SIP early dialogue, it shall remove or update
this source filter.
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IM-MGW MGCF
1. ISUP: IAM
Reserve IMS Connection Point, 4. H.248: ADD.req [Context ID = C1, Termination ID=?]
Change IMS Through-
Connection = backward 5. H.248: ADD.resp [Context ID = C1, Termination ID = T1]
6. SIP: INVITE
8. SIP:183 Session
Progress
(to:xxx, tagA)
9. H.248: MOD.req [Context ID = C1,Termination ID = T1]
Configure IMS Resources
10.H.248: MOD.resp [Context ID = C1, Termination ID = T1 ] 11. SIP: PRACK
(to:xxx, tagA)
Figure 39a/1: CS Network Originating Session with forking, ISUP (message sequence chart)
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IM-MGW MGCF
Stop TDM Tone , 33. H.248: MOD.req [Context ID=C1, Termination ID = T2]
Activate TDM Voice
Processing Function 34. H.248: MOD.resp [Context ID = C1, Termination ID = T2]
Figure 39a/2: CS Network Originating Session with forking, ISUP (message sequence chart continue)
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9.2.4.1 BICC
MGCF IM-MGW
1. SIP: BYE
3. BICC: REL
4. BICC: RLC
Figure 40: Session release from IM CN subsystem side for BICC (message sequence chart)
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9.2.4.2 ISUP
MGCF IM-MGW
1. SIP: BYE
8. ISUP: RLC
Figure 41: Session release from IM CN subsystem side for ISUP (message sequence chart)
9.2.5.1 BICC
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MGW that the CS network side bearer termination shall be removed and the bearer shall be released towards the
preceding MGW (signal 3 to 6 in figure 42). After completion of resource release, the MGCF shall send a RLC message
towards the preceding node.
IM-MGW MGCF
1.BICC: REL
2. SIP: BYE
9. BICC: RLC
Figure 42: Session release from CS network side for BICC (message sequence chart)
9.2.5.2 ISUP
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IM-MGW MGCF
1.ISUP: REL
2. SIP: BYE
7. ISUP: RLC
Figure 43: Session release from CS network side for ISUP (message sequence chart)
9.2.6.1 BICC
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Release 8 185 3GPP TS 29.163 V8.16.0 (2011-09)
MGCF IM-MGW
1. BICC: REL
2. SIP: BYE
3. BICC: RLC
Release Termination
7. H.248: SUB.resp [Context ID = C1, Termination ID = T2 ]
Figure 44: Session release initiated by MGCF for BICC (message sequence chart)
9.2.6.2 ISUP
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Release 8 186 3GPP TS 29.163 V8.16.0 (2011-09)
IM-MGW
MGCF
1. SIP: BYE
2. ISUP: REL
7. ISUP: RLC
8. SIP: 200 OK [BYE]
Figure 45: Session release initiated by MGCF for ISUP (message sequence chart)
9.2.7.1 BICC
NOTE: Other actions related to MGW Out-Of-Service procedure is defined in 3GPP TS 23.205 [27].
NOTE: Other actions related to MGW-Out-Of-Service procedure is defined in 3GPP TS 23.205 [27]
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MGCF IM-MGW
or
1b. H.248: Notify.req [Context ID = C1, Termination ID = T1]
IMS Bearer
Released 2b. H.248: Notify.resp [Context ID = C1, Termination ID = T1]
3. BICC: REL
4. SIP: BYE
5. BICC: RLC
Figure 46: Session release initiated by the IM-MGW for BICC (message sequence chart)
9.2.7.2 ISUP
NOTE: Other actions related to "MGW-Out-Of-Service" procedure is defined in 3GPP TS 23.205 [27].
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procedure (signals 5 and 6 in figure 47). The MGCF also expects to receive a 200 OK [BYE] message from the IM CN
subsystem side (signal 10 in figure 47).
NOTE: Other actions related to "MGW-Out-Of-Service" procedure are defined in 3GPP TS 23.205 [27].
MGCF IM-MGW
or
1b. H.248: Notify.req [Context ID = C1, Termination ID = T2]
Bearer Released
2b. H.248: Notify.resp [Context ID = C1, Termination ID = T2]
or
1c. H.248: Notify.req [Context ID = C1, Termination ID = T1]
IMS Bearer
Released 2c. H.248: Notify.resp [Context ID = C1, Termination ID = T1]
3. ISUP: REL
4. SIP: BYE
9. ISUP: RLC
Figure 47: Session release initiated by the IM-MGW for ISUP (message sequence chart)
Before the actual usage of the telephony signals can occur the sending/receiving of telephone events need to be agreed
with the SDP offer-answer mechanism defined in RFC 3264 [36]. The outcome of the negotiation can be e.g. that no
telephone events are sent in RTP payload, telephone events are sent only in one direction or in both directions. If the
outcome of the negotiation is that RTP payload telephone-events are sent in both directions, the IM-MGW may
nevertheless be configured to interwork only mobile originated telephone-events.
When the offer-answer mechanism based session parameters negotiation results in an agreement that telephone events
are sent in the RTP payload and the needed preconditions are fulfilled, telephone events can be sent in RTP payload.
This negotiation can be done at call control signalling phase or during an ongoing call.
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If the MGCF and IM-MGW support the reception and/or transmission of the RTP MIME type "telephone event" (as
defined in RFC 4733 [105]) with the IMS, the following applies:
- For CS Network Originating Sessions, the MGCF shall include the MIME type "telephone events" with default
events in the first SDP offer. After the usage of telephone events is agreed in the subsequent offer-answer
parameter exchanges and the needed preconditions defined in RFC 3312 [37] are fulfilled, telephone events can
be sent as RTP payload.
- In case of IM CN Subsystem Originating Sessions, the MGCF shall accept the MIME type "telephone events"
with default events in any SDP answer when it received such an offer.
Furthermore, the MGCF shall use the "Detect IMS RTP Tel Signal" procedure to request the MGW to detect incoming
telephone events from the IMS and notify the MGCF about the detected events. The MGW shall use the "Notify IMS
RTP Tel Event" procedure for this notification. The termination used to receive DTMF shall be placed in the same
context used for the speech of the same call. The MGCF shall request to be notified when the MGW detects the end of a
digit and may also request to be notified when the MGW detects the start of a digit. An IM-MGW not supporting the
notification about the detection of the start of a digit may ignore the request to provide this notification. If the IM-MGW
received a "Detect IMS RTP Tel Event" procedure for a termination, the IM-MGW shall not forward inband to the CS
network any DTMF received at this termination.
Figure 48 shows an example message sequence chart when a DTMF digit is received from the IM CN subsystem in the
RTP payload and the MGCF has requested to be notified only about the detection of the end of a digit. Figure 48a
shows an example message sequence chart when a DTMF digit is received from the IM CN subsystem in the RTP
payload and the MGCF has requested to be notified about the detection of the start and the end of a digit.
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Release 8 190 3GPP TS 29.163 V8.16.0 (2011-09)
IM-MGW MGCF
… 1 Digit
9. RTP encoded continued DTMF event
(tone-event, end=1, duration= Duration)
Figure 48: Activation of notification of DTMF digits received in RTP and examples of sending the
digits out-of-band to CS CN, a whole digit received by IM-MGW before sending further (message
sequence chart)
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Release 8 191 3GPP TS 29.163 V8.16.0 (2011-09)
IM-MGW MGCF
4. H.248 Notify.ind
[ Context ID = C1, Termination ID = T1, Tone_ID,
Event=”Start tone detected”]
Figure 48a: Activation of notification of DTMF digits received in RTP and examples of sending the
digits out-of-band to CS CN, IM-MGW starts sending the digit further when the start of the digit is
recognized (message sequence chart)
9.2.8.2 Sending and receiving DTMF digits inband to/from CS CN (ISUP or BICC)
For the IM CN subsystem terminated session, the MGCF shall use the "Configure IMS Resources" procedure as
described in Clause 9.2.3. For the IM CN subsystem originating session , the MGCF shall use the "Reserve IMS
Connection Point and Configure Remote Resources" procedure as described in Clause 9.2.2. If DTMF is supported and
the MGCF wants to configure the IM_MGW to send and receive DTMF to/from the CS network side, the MGCF shall
include "telephone event" along with the selected speech codecs within the "local IMS resources" parameter of these
procedures to request the MGW to detect incoming telephone events and transform them into speech signals on the CS
side and shall not apply the "Detect IMS RTP Tel Event" procedure. When receiving this configuration, an MGW
supporting DTMF shall detect DTMF encoded according as RTP Tel Event and transform this into DTMF tones
encoded within the speech codec used at the CS CN network and may in addition optionally detect incoming telephone
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events received inband from the CS CN network and transform them into telephone events on the IMS side. The same
termination shall be used to receive and transmit DTMF and speech of the same call.
Figure 49 shows the message sequence chart to configure the IM-MGW to receive DTMF detection on the IMS side
and transfer the DTMF inband on the CS side. When receiving this configuration, the IM-MGW may in addition
optionally detect DTMF inband on the CS side and transmit DTMF on the IMS side.
IM-MGW MGCF
1. H.248 Add/Mod.req
[Context ID = C1, Termination ID = T1,
Codec = “telephone event, AMR”,]
Configure IMS Resources
Or
Reserve IMS Conection Point
and configure Remote 2. H.248 Add/Mod.resp [Context ID = C1, Termination ID = T1]
Resources
Figure 49: Activation of processing of DTMF digits received in RTP for sending the digits inband to
CS CN (message sequence chart)
Furthermore, the MGCF shall use the "Send IMS RTP Tel Event" and may use the "Stop IMS RTP Tel Event"
procedures to request the MGW to play out DTMF to the IM CN subsystem whenever it receives out-of-band DTMF
indications from the BICC network.
Figure 49a shows an example message sequence chart when DTMF digits are transmitted to the IM CN subsystem in
the RTP payload. For the first digit, the received APM message contains all information including the duration and only
a single notification is received. For the second digit, the start and the end of the DTMF digit are notified separately.
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IM-MGW MGCF
…
7. RTP encoded new DTMF event
(tone-event, end=1, duration)
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Hold request
When the IMS network makes a hold request by sending an UPDATE or re-INVITE message (signal 1 of figure 50),
the MGCF shall request the IM-MGW to suspend sending media towards the IMS side by changing the through-
connection of the IM CN subsystem side termination to ‘not through-connected’ (signal 2 of figure 50). If the IMS side
provides modified SDP RR or RS bandwidth modifiers, as specified in IETF RFC 3556 [59], within the hold request,
the MGCF shall use the Configure IMS Resources Mn procedure to forward this information to the IM-MGW (not
depicted in figure 50, but may be combined with signal 2). The MGCF shall send a CPG (Hold) message to the
succeeding CS network node to indicate that the session is on hold (signal 4 of figure 50). Simultaneously a SIP
message acknowledging the Hold request is sent to the IMS side (signal 7 of figure 50, acknowledged by signal 7.a if
the INVITE method is used). Announcements may be applied to the party on hold, depending on the held party’s status,
using the Play Announcement procedure (for BICC) or the Play TDM Announcement procedure (for ISUP, signal 5 in
figure 50). The hold operation shall not block RTCP flows.
Resume request
When the IMS network makes a request to retrieve the session on hold by sending an UPDATE or re-INVITE message
(signal 8 of figure 50), the MGCF shall request the IM-MGW to re-establish communication towards the IMS network
by changing the through-connection of the IM CN subsystem side termination to both-way through-connected (signal
11 of figure 50). If the IMS side provides modified SDP RR or RS bandwidth modifiers, as specified in IETF RFC 3556
[59], within the retrieve request, the MGCF shall use the Configure IMS Resources Mn procedure to forward this
information to the IM-MGW (not depicted in figure 50, but may be combined with signal 11). Possible announcements
to the party on hold shall be stopped using the Stop Announcement procedure (for BICC) or the Stop TDM
Announcement procedure (for ISUP, signal 9 in figure 50). The MGCF shall send a CPG (Retrieve) message to the
succeeding CS network node to indicate that the session is retrieved (signal 13 of figure 50).
Figure 50 shows the message sequence chart for the call hold and retrieval procedures.
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Release 8 195 3GPP TS 29.163 V8.16.0 (2011-09)
MGCF IM-MGW
5. H.248: MOD.req
Play TDM [Context ID = C1, Termination ID = T2]
Announcement
6. H.248: MOD.resp
[Context ID = C1, Termination ID = T2]
7. SIP: 200 OK [SDP]
9. H.248: MOD.req
[Context ID = C1, Termination ID = T2]
When an MGCF receives a CPG message with a ’remote retrieval’ Generic notification indicator (signal 6 of figure 51),
the MGCF forwards the resume request by sending an UPDATE or re-INVITE message containing SDP with
"sendrecv" or "recvonly" media (signal 9 of figure 51).
If the MGCF receives a CPG with ‘remote hold’ or ‘remote retrieval’ before answer, it shall forward the request using
an UPDATE message. If the MGCF receives a CPG with ‘remote hold’ or ‘remote retrieval’ after answer, it should
forward the request using re-INVITE but may use UPDATE.
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If link aliveness information is required at the IM-MGW while the media are on hold, the MGCF should provide to the
modified SDP RR and RS bandwidth modifiers specified in IETF RFC 3556 [59] within the SDP offers in the UPDATE
or re-INVITE messages holding and retrieving the media to temporarily enable RTCP while the media are on hold, as
detailed in Clause 7.4 of 3GPP TS 26.236 [32]. If no link aliveness information is required at the IM-MGW, the MGCF
should provide the SDP RR and RS bandwidth modifiers previously used.
The interworking does not impact the user plane, unless the MGCF provides modified SDP RR and RS bandwidth
modifiers in the UPDATE or re-INVITE messages. If the MGCF provides modified SDP RR and RS bandwidth
modifiers in the UPDATE or re-INVITE messages, the MGCF shall also provide modified SDP RR and RS bandwidths
to the IM-MGW using the Configure IMS Resources procedures (signals 2-3 and 7-8 of figure 51).
Figure 51 shows the message sequence chart for the call hold and retrieval procedures.
MGCF IM-MGW
2. H.248: MOD.req
Configure IMS [Context ID = C1, Termination ID = T1]
Resources
(optional) 3. H.248: MOD.resp
[Context ID = C1, Termination ID = T1]
7. H.248: MOD.req
Configure IMS [Context ID = C1, Termination ID = T1]
Resources
(optional) 8. H.248: MOD.resp
[Context ID = C1, Termination ID = T1]
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Table 25: Procedures toward the IM Subsystem: Reserve IMS connection point
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Table 27: Procedures toward the IM Subsystem: reserve local, reserve remote IMS connection point
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9.3.1.7 Void
Table 30a: Procedures between (G)MSC server and MGW: Hanging termination indication
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This procedure is the same as Activate Voice Processing Function in 3GPP TS 23.205 [27].
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Table 35a: Procedures between (G)MSC server and MGW: Hanging termination indication
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IM-MGW
CS CN1
IM CN1
Termination Termination
1 2
Shown in Figure 9.3.5, the IM CN1 and CS CN1 represent separate IP realms. The definition of IP realm is specified in
IETF RFC 2663[90].
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Release 8 207 3GPP TS 29.163 V8.16.0 (2011-09)
The termination1 and termination2 are connected with different IP realms in the IM-MGW separately.
For establishing session when multiple IP realms are used in the IM-MGW, the MGCF may indicate the IP realm
identifier to the IM-MGW. The IM-MGW shall assign the IP termination with the indicated IP realm.
A default IP realm may be configured in IM-MGW such that if the IM-MGW has not received the IP realm identifier
and the IM-MGW supports multiple IP realms then the default IP realm shall be used.
If the IM-MGW does not support the option to indicate an IP realm then it is free to select any IP port.
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Release 8 208 3GPP TS 29.163 V8.16.0 (2011-09)
Annex A (informative):
Summary of differences items between 3GPP TS 29.163
and ITU-T Q.1912.5
The present document specifies the principles of interworking between the 3GPP IM CN subsystem and BICC/ISUP
based legacy CS networks, in order to support IMS basic voice calls. A specification exists in the ITU-T that covers
similar work: Interworking between Session Initiation Protocol (SIP) and Bearer Independent Call Control Protocol or
ISDN User Part. (ITU-T Q.1912.5 [49]) in order to support services that can be commonly supported by BICC or ISUP
and SIP based network domains. Three profiles are described in the ITU-T specification: A, B, and C. Profile B and C
are out of the scope of the present specification.
3GPP intends to strive for alignment with ITU-T Q.1912.5 [49], however some differences exist. This annex contains a
list of these differences. Future revisions of this document will seek to incorporate text to address these differences.
This Annex is intended as an informative tool for the designer community and operators to understand the main
differences between 3GPP and ITU recommendations for the SIP-BICC/ISUP interworking.
The list of differences between TS 29.163 and ITU-T Q.1912.5 [49] is referred to profile A of the latter.
Extra entry comprising the case when SIP procedures result in release after answer.
6. Satellite indicator
It is set to "00 No satellite circuit in the connection". While in ITU-T Q.1912.5 [49] is set to "01 one satellite
circuit in the connection"
7. The mapping of the Reason Header and the Location Field mapping is missing in the 3GPP specification,
whereas in ITU is specified.
The reason for this is that the Reason Header was included in IMS only as optional. As the reason header is
optional, it can be proprietary interworked and in that case ITU-T mapping recommendation can be used.
8. COLP/COLR Service interworked is included in 29.163, and left FFS in ITU-T Q.1912.5 [49]
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Annex B (normative):
Codec Negotiation between a BICC CS network and the IM
CN subsystem
B.1 Introduction
This annex describes optional procedures for interworking of codec negotiation between a BICC CS network and the
IM CN subsystem.
When the I-MGCF receives an INVITE without SDP offer, the I-MGCF shall continue call establishment without
interworking of codec negotiation procedures. The mid-call interworking procedures of clause B.2.3 and clause B.2.4
may still apply.
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Certain BICC timers or events can force completion of the incoming bearer set-up procedure before the O-MGCF
receives the SDP answer from the IM CN subsystem. In this case, the O-MGCF shall perform the terminating codec
negotiation procedure according to clause 8.3.3 of ITU-T Q.1902.4 [30], including all supported codecs in the Available
Codec List, and shall resume the incoming bearer set-up procedure without waiting any longer for the SDP answer.
When an SDP answer arrives from the IM CN subsystem in response to the SDP offer in an INVITE request after the
BICC incoming bearer set-up procedure has started, the O-MGCF shall select a codec configuration for use on the
bearer interface to the IM CN subsystem from the codecs in the SDP answer, choose a new Selected Codec for the
BICC CS network from the codecs in the Available Codec List constructed during incoming bearer set-up, and initiate
the second SDP offer/answer exchange with the IM CN subsystem using the codec selected for the IM CN subsystem, if
necessary. The O-MGCF should select codecs for the bearer interfaces to the BICC CS network and IM CN subsystem
in such a way as to avoid transcoding at the IM-MGW and minimize speech degradation, if possible, according to
clause B.2.5. Otherwise the O-MGCF should select the highest priority codecs from the available options for the two
bearer interfaces. If the SDP answer only included a single voice codec, then there is no need for a second SDP
offer/answer exchange, and the codec selected for the IM CN subsystem is the codec in the SDP answer. When the call
in the BICC CS network enters a state capable of supporting codec modification, if the new Selected Codec is different
from the Selected Codec chosen during the incoming bearer set-up procedure for the BICC CS network, the O-MGCF
should initiate the codec modification procedure towards the BICC CS network using the new Selected Codec,
according to clause 10.4.1 of ITU-T Q.1902.4 [30].
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into a Supported Codec List, delete those codecs in the Supported Codec List not supported at the IM-MGW, and
initiate the mid-call codec negotiation procedure according to clause 10.4.4 of ITU-T Q.1902.4 [30], by sending an
APM with the Supported Codec List and an Action indicator set to "mid-call codec negotiation". When generating the
Supported Codec List, the MGCF should add to the SDP offer all codec configurations for which it can provide
transcoding.
When the MGCF receives a SIP message with an SDP offer that is not associated with incoming call bearer
establishment or preconditions, if the call is not in a state capable of supporting BICC codec negotiation, the MGCF
shall respond to the SDP offer with existing procedures for the IM CN subsystem. When the call is in a state capable of
supporting BICC codec negotiation, the MGCF may send a re-INVITE request without SDP towards the IM CN
subsystem, soliciting a response with an SDP offer, thereby restarting the codec negotiation interworking procedure.
If the succeeding serving node returns an Action indicator set to "mid-call codec negotiation failure", the MGCF either
should send a 488 response to the SDP offerer indicating rejection of the initial SDP offer, or should select the highest
priority codec from the codecs in the received SDP offer supported by the IM-MGW, format an SDP answer based on
this selected codec, and send the SDP answer to the offerer in the appropriate SIP message. If the MGCF sends a 488
response to the SDP offerer, it should continue the call with the bearer configuration in place before initiating this codec
negotiation procedure.
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towards the IM CN subsystem in an attempt to recreate the bearer configuration in place before this codec negotiation
procedure began.
If the MGCF receives a 488 response or other failure response (e.g. 3xx-6xx) to the SDP offer, either it should reject the
mid-call codec negotiation from the BICC CS network by sending an APM with an Action indicator set to "mid-call
codec negotiation failure" towards the preceding serving node, or it should continue as if it received an SDP answer
with no change in codec selected for the IM CN subsystem. If the MGCF sends an APM with an Action indicator set to
"mid-call codec negotiation failure", it should continue the call with the bearer configuration in place before initiating
this codec negotiation procedure.
If the MGCF receives an APM from a BICC CS network that includes an Action indicator set to "modify codec" and
the new selected codec in the message is different from the Selected Codec at the IM-MGW bearer interface to the
BICC CS network, the MGCF either may act as a serving node terminating codec modification, according to clause
10.4.2 of ITU-T Q.1902.4 [30], without interworking the procedure with the IM CN subsystem, or may follow the
procedures of clause B.2.5 to convert the new Available Codec List (with new priority order) from the APM into an
SDP offer for transmission in an appropriate SIP message (e.g. re-INVITE request) towards the IM CN subsystem,
according to RFC 3264 [36], deleting those codecs not supported at the IM-MGW. When generating the SDP offer, the
MGCF should include all codec configurations for which it can provide transcoding in addition to those converted from
the new Available Codec List. The MGCF shall include at least one AMR codec configuration in the SDP offer.
If the MGCF sends a SIP message with an SDP offer towards the IM CN subsystem in response to receipt of a BICC
codec modification request, then it shall delay responding to the BICC codec modification request until it receives a
response to the SDP offer from the IM CN subsystem. When the MGCF receives either an SDP answer or a rejection of
the SDP offer within the appropriate SIP message (e.g. 200 OK (INVITE)) from the IM CN subsystem, it shall decide
whether to accept or reject the BICC codec modification procedure and complete the procedure for a BICC serving
node terminating codec modification, according to clause 10.4.2 of ITU-T Q.1902.4 [30].
If the MGCF sends an APM with an Action indicator set to "codec modification failure" in response to receipt of a
codec modification request, the preceding BICC serving node may retry the request with a mid-call codec negotiation
using an APM including an Action indicator set to "mid-call negotiation" and a Supported Codec List with a new
priority order encouraging selection of a new codec.
The bearer independent CS network uses the Single Codec and Codec List information elements of the Application
Transport Mechanism (APM) defined in ITU-T Recommendation Q.765.5 [35] to negotiate (offer and select) and
potentially re-negotiate the codec type and configuration and associated bearer format attributes to be used in the user
plane. 3GPP TS 29.414 [25] and 3GPP TS 29.415 [26] define the IuFP framing protocol for all codecs in the user plane
for both ATM and IP transport, and 3GPP TS 26.102 [50] provides the framing details for AMR and PCM family
codecs. The Codec List information element of the APM comprises multiple instances of the Single Codec information
element in priority order, as shown in Figure 13 of ITU-T Recommendation Q.765.5 [35]. Figure 14 of ITU-T
Recommendation Q.765.5 [35] defines the Single Codec information element. Clause 11.1.7.2 of ITU-T
Recommendation Q.765.5 [35] defines the encoding of the Single Codec information element for the ITU-T codecs.
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3GPP TS 26.103 [57] defines the encoding of the Single Codec information element for the 3GPP codecs, and Table
7.11.3.1.3-2 of 3GPP TS 28.062 [58] defines the preferred configurations of the narrowband AMR codecs (Config-NB-
Code) for interoperation with TFO. The signalling plane of the bearer independent CS network uses the APM to
negotiate the desired codec type and configuration for the user plane from the prioritized list of codec types and to re-
negotiate the user plane attributes as necessary.
The following subclauses define the translations between the SDP payload format parameters of the IM CN subsystem
and the corresponding subfields of the Single Codec information element of the bearer independent CS network for
certain 3GPP and ITU-T codecs. Following these translation rules will in many cases allow the IM-MGW to perform
interworking between the framing protocols on the bearer interfaces to the BICC CS network and the IM CN subsystem
without transcoding. Implementations may signal other codec types not listed herein or other codec configurations of
codec types listed herein. Implementations may also choose to perform transcoding between codec configurations
signalled separately for the bearer interfaces to the networks, if necessary, but voice quality may suffer.
Table B.1: Mapping between Single Codec subfields and SDP parameters for 3GPP AMR-NB codecs
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Definitions:
Supported Codec List: contains the offered Codec Types and Configuration-possibilities of the node initiating codec
negotiation in BICC (see also TS 23.153). The Supported Codec List is sent from the initiating node forward to the
terminating node. The Supported Codec List corresponds to an SDP offer during codec negotiation.
Available Codec List: contains the offered Codec Types and Configuration-possibilities of the contiguous portion of
the connection between initiating and terminating BICC nodes, including all intermediate nodes through the BICC
network(s). The Available Codec List is sent from the BICC node terminating codec negotiation backward to the
initiating node. The Available Codec List corresponds to information sometimes available in a first-round SDP answer.
The Available Codec List might not represent an end-to-end view of the available Codec Types and Configuration-
possibilities when traversing both BICC and SIP networks.
Selected Codec Type: is determined by the node terminating codec negotiation. It specifies exactly the Codec Type and
one unique Codec Configuration for the call. The Selected Codec Type corresponds to the final SDP answer.
When translating from a Single Codec information element to the equivalent SDP payload format parameters, where
either OM=0 (in the Supported or Available Codec List) or the information element is the Selected Codec Type, the
SDP shall include a single payload type and any associated parameters from the corresponding row in Table B.1. When
translating from a Single Codec information element to the equivalent SDP payload format parameters, where OM=1 in
the Supported or Available Codec List, the SDP shall only include payload formats corresponding to Codec
Configurations compatible with the offered ACS, SCS and MACS, according to Table B.1. Since the number of
compatible payload formats can be large, implementations should select a reasonable subset of the higher-priority
payload formats for inclusion in the SDP. When translating a list of Single Codec information elements into SDP,
duplicate payload types (matching on all parameters) shall be removed.
The following guidelines shall apply when translating from an SDP payload format specification to a Single Codec
information element:
- If there is no "mode-set" parameter for a payload format in the SDP and the SDP is to be translated into a
Supported or Available Codec List, then the corresponding Single Codec subfields shall be OM=1, MACS=8, all
SCS modes offered, and ACS modes offered. Alternatively it is sufficient to specify only the Codec Type (see
below) and omit the other parameters.
- If there is no "mode-set" parameter for a payload format in an SDP answer that is to be translated into a Selected
Codec Type, then the corresponding Single Codec subfields shall be derived from the payload type in the SDP
offer (to which the SDP answer was sent in response).
- If there is a "mode-set" parameter for a payload format in the SDP, then the corresponding Single Codec
subfields shall be OM=0 and ACS modes selected according to the value of "mode-set". The SCS shall be set
identical to the ACS and MACS shall be set to the number of modes in the ACS. If this "mode-set" does not
represent a valid configuration for the Codec Type (determined by OoBTC procedures), then the payload format
shall not be translated.
- If a payload format in an SDP offer that is to be translated into a Supported Codec List includes "mode-change-
period=2", then the Codec IDentification value for the corresponding Single Codec shall be FR_AMR.
- If a payload format in an SDP answer that is to be translated into a Selected Codec Type or Available Codec List
includes "mode-change-period=2", then the Codec IDentification value for the corresponding Single Codec shall
be one of FR_AMR, HR_AMR, OHR_AMR or UMTS_AMR_2, if offered in the Supported Codec List.
- If a payload format in an SDP offer that is to be translated into a Supported Codec List does not include "mode-
change-period=2", then the Codec IDentification value for the corresponding Single Codec shall be
UMTS_AMR.
- If a payload format in an SDP answer that is to be translated into a Selected Codec Type or Available Codec List
does not include "mode-change-period=2", then the Codec IDentification value for the corresponding Single
Codec shall be one of UMTS_AMR_2, FR_AMR, HR_AMR, OHR_AMR or UMTS_AMR, if offered in the
Supported Codec List.
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Table B.2: Mapping between Single Codec subfields and SDP parameters for 3GPP AMR-WB codecs
When translating from a Single Codec information element to the equivalent SDP payload format parameters, the SDP
shall include a distinct payload type and any associated parameters for each row in the table that matches the Config-
WB-Code parameter. For example, OFR_AMR-WB with Config-WB-Code=3 can generate three SDP payload types
for AMR-WB, each including the "mode-change-period=2" parameter, the "mode-change-neighbor=1" parameter, and
the "mode-set" parameter with value sets "0,1,2,4", "0,1,2,8", and "0,1,2", respectively. When translating a list of Single
Codec information elements into SDP, duplicate payload types (matching on all parameters) shall be removed.
The following guidelines shall apply when translating from one or more SDP payload format specifications to a Single
Codec information element:
- Payload formats that match except for different values of "mode-set" shall be represented with the fewest values
of Config-WB-Code, while retaining the priority represented by the order of the payload formats in the SDP. For
example, three SDP payload types for AMR-WB, each including the "mode-change-period=2" parameter, the
"mode-change-neighbor=1" parameter, and the "mode-set" parameter with value sets "0,1,2,4", "0,1,2,8", and
"0,1,2", respectively, will generate Config-WB-Code=3.
- If there is no "mode-set" parameter for a payload format in the SDP and the SDP is to be translated into a
Supported or Available Codec List, then the corresponding Single Codec shall have a Config-WB-Code value of
1.
- If there is no "mode-set" parameter for a payload format in an SDP answer that is to be translated into a Selected
Codec Type, then the corresponding Config-WB-Code value shall be derived from the payload type in the SDP
offer (to which the SDP answer was sent in response).
- If a payload format in an SDP offer that is to be translated into a Supported Codec List includes "mode-change-
period=2", then the Codec IDentification value for the corresponding Single Codec shall be OFR_AMR-WB.
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- If a payload format in an SDP answer is to be translated into a Selected Codec Type or Available Codec List,
then the Codec IDentification value for the corresponding Single Codec shall be one of OFR_AMR-WB,
FR_AMR-WB, OHR_AMR-WB or UMTS_AMR-WB, if offered in the Supported Codec List.
- If a payload format in an SDP offer that is to be translated into a Supported Codec List does not include "mode-
change-period=2", then the payload format shall not be translated.
Table B.3: Mapping between Single Codec subfields and SDP parameters for 3GPP non-AMR codecs
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Table B.4: Mapping between Single Codec subfields and SDP parameters for ITU-T codecs
When translating from a Single Codec information element to the equivalent SDP payload format parameters, the SDP
shall include a distinct payload type and any associated parameters for each matching instance of the Codec
Configuration subfield. For example, G.726 (ADPCM) with Codec Configuration subfield "0101" shall generate SDP
payload types for G726-32 and G726-16.
When translating from an SDP payload format specification to the Single Codec information element, each SDP
payload type should be represented by one matching Single Codec information element. For example, SDP payload
types for G729 and G729E may generate one Single Codec information element for "G.729 Annex B" with Codec
Configuration subfield "110". The G729 and G729E codecs may alternately be represented by two Single Codec
information elements for "G.729 Annex B" with Codec Configuration subfields "100" and "010", respectively, if it is
necessary to indicate preference between them.
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- Shall send the appropriate remote codec(s), the remote UDP port and the remote IP address to the IM-MGW.
The remote UDP port and IP address refer to the destination of user plane data sent towards the IM CN
subsystem. The remote codec(s) are the codec(s) the IM-MGW may select for user plane data sent towards the
IM CN subsystem.
- Shall indicate to the IM-MGW the appropriate local codec(s) and request a local IP address and UDP port. The
local IP address and UDP port are used by the IM-MGW to receive user plane data from the IM CN subsystem.
The local codec(s) are the codec(s) the IM-MGW may select to receive user plane data from the IM CN
subsystem.
- If DTMF support together with speech support is required, the reserve value indicator shall be set to "true".
The IM-MGW
- Shall reply to the MGCF with the selected local codec(s) and the selected remote codec(s) and the selected local
UDP port and IP address.
The MCGF shall send the local codec(s), UDP port and IP address to the IMS in the Session Progress (signal 9 in figure
B.1).
B.3.1.1.4 Through-connection
During the Prepare Bearer and Establish Bearer procedures, the MGCF shall either use the Change Through-Connection
procedure to request the IM-MGW to backward through-connect the BICC terminations, or the MGCF shall use this
procedure to both-way through-connect the BICC termination already on this stage (signal 5 in figure B.1). During the
Reserve IMS Connection Point procedure, the MGCF shall use the Change IMS Through-Connection procedure to
request the IM-MGW to backward through-connect the IMS termination (signal 7 in figure B.1).
When the MGCF receives the BICC:ANM answer indication, it shall request the IM-MGW to both-way through-
connect the termination using the Change Through-Connection or Change IMS Through-Connection procedures (signal
20 in figure B.1), unless those terminations are already both-way through-connected.
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NOTE: As an implementation option the MGCF may also decide for example to only release the resources in the
IM-MGW that caused the failure, possibly select a new IM-MGW for the connection and continue the
call establishment using new resources in the selected IM-MGW but such handling is outside of the scope
of the present document.
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MGCF IM-MGW
1. SIP: INVITE
[supported codec list 1]
Bearer Establishment
Figure B.1: Basic IM CN Subsystem originating session, BICC forward bearer establishment
(message sequence chart)
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The IM-MGW shall reply to the MGCF with the selected local codec(s) and the selected local IP address and UDP port.
The MGCF shall send this information in the INVITE (signal 4 in figure B.2/1) to the IM CN subsystem.
- The MGCF shall indicate the remote IP address and UDP port, i.e. the destination IP address and UDP port for
data sent in the user plane towards the IM CN subsystem,
- The MGCF shall indicate the remote codec(s), i.e. the speech codec(s) for data sent in the user plane towards the
IM CN subsystem.
- The MGCF may indicate the local codec(s) and the local IP address and UDP port. The MGCF shall indicate the
local codec(s) if a change is required.
- IF DTMF support together with speech support is required, the reserve value indicator shall be set to "true".
The IM-MGW shall reply with the selected remote codec(s) and reserve resources for these codec(s). If local codec(s)
were received, the IM-MGW shall also reply with the selected local codec(s) and reserve the corresponding resources.
If the selected local codec(s) differ from the codec(s) received in the SDP of signal 6 in figure B.2/1, the MGCF shall
send the local reserved codec(s), and the local IP address and UDP port in the PRACK (signal 9 in figure B.2/1) to the
IMS.
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MGW has replied with the bearer address and the binding reference, the MGCF provides the APM message (signal 13
in figure B.2/1) to the preceding node. The MGCF may also provide the IM-MGW-id in the APM message.
B.3.2.1.7 Through-Connection
During the Prepare Bearer procedure, the MGCF shall either use the Change Through-Connection procedure to request
the IM-MGW to backward through-connect the BICC termination, or the MGCF shall use this procedure to both-way
through-connect the BICC termination already on this stage (signals 11 and 12 in figure B.2/1). During the Reserve
IMS Connection Point procedure, the MGCF shall use the Change IMS Through-Connection procedure to request the
IM-MGW to backward through-connect the IMS termination (signals 2 and 3 in figure B.2/1).
When the MGCF receives the SIP 200 OK(INVITE) (signal 23 in figure B.2/2), it requests the IM-MGW to both-way
through-connect the terminations using the Change IMS Through-Connection or Change Through-Connection
procedures (signals 28 and 29 in figure B.2/2), unless those terminations are already both-way through-connected.
NOTE: As an implementation option the MGCF may also decide for example to only release the resources in the
IM-MGW that caused the failure, possibly select a new IM-MGW for the connection and continue the
call establishment using new resources in the selected IM-MGW but such handling is outside of the scope
of the present document.
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IM-MGW MGCF
4. SIP: INVITE
[supported codec list 2]
5. SIP: 100 Trying
6. SIP:183
[available codec list 2]
7. H.248: MOD.req Context ID = C1,Termination ID = T1]
Configure IMS Resources
8. H.248: MOD.resp [Context ID = C1, Termination ID = T1 ] 9. SIP: PRACK
[selected codec 2]
Bearer Establishment
Figure B.2/1: Basic CS Network Originating Session, BICC forward bearer establishment
(message sequence chart)
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IM-MGW MGCF
Figure B.2/2: Basic CS Network Originating Session, BICC forward bearer establishment
(message sequence chart continued)
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IM-MGW MGCF
3. SIP:200 OK (INVITE)
[selected codec 2]
8. SIP: ACK
9. BICC: APM [selected codec 1]
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IM-MGW MGCF
1. SIP: re-INVITE
[supported codec list 1]
2. BICC: APM [supported codec list 2]
9. SIP:200 OK (INVITE)
[selected codec 1]
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Annex C (normative):
Interworking of CPC parameter
The CPC extension to tel-URI Parameter is defined in sub-clause 7.2A.12 of 3GPP TS 24.229 [9]. The SIP "Accept-
Language" header is defined in IETF RFC 3261 [19].
Table C.1-1: Mapping of the CPC parameter to the ISUP Calling party's category parameter
In case the Accept-Language header field is not received or contains values that are not in this table then based on
operator policy the Calling party's category parameter shall contain the CPC value "operator, language X" (where X is
one of the following languages: French, English, German, Russian or Spanish) or national/regional specific value.
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Table C.2-1: Mapping of the ISUP Calling party's category parameter to the CPC parameter
Annex D:
Void
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Annex E (normative):
Multimedia interworking between the IP Multimedia Core
Network (CN) Subsystem (IMS) and Circuit Switched (CS)
networks
3GPP TS 26.111
Call Set-up
Figure E.1.1.1 Overview of relevant CS-Domain Protocols (from 3GPP TS 26.110 [78])
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E.2.1 General
In addition to the control plane Interworking between SIP and ISUP or BICC, interactions between the H.245 signalling
at the CS side and SIP/SDP signalling are described.
The establishment of the H.223 multiplexing protocol and the subsequent H.245 signalling procedures take place after
the set-up and both-way through-connection of the CS bearer.
NOTE: Detailed procedures for the mapping between SDP parameters and H.245 parameters are not specified in
the present release.
Figure E.2.3.1.1.1 assumes that the IMS peer uses the SIP precondition extension to indicate that preconditions have not
yet been met.
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Interworking Node
IMS (MGCF+ IM MGW) CS Network
1. SIP: INVITE
[SDP offer:,
m = H.263, MP4V-ES, H.261;
m= AMR,
Preconditions not met]
2. IAM
3. SIP:
[SDP answer:
m = H.263;
m= AMR, 4. Bearer Establishment
Preconditions not met ]
5. ANM
Figure E.2.3.1.1.1: Interactions between H.245 and SIP/SDP for IM CN subsystem originated session
IMS peer indicates unmet local preconditions
Upon receipt of a SIP INVITE request containing speech and video Codecs (signal 1 in figure E.2.3.1.1.1) the
Interworking Node (consisting of MGCF and IM-MGW) starts the call set-up for multimedia call at the CS side by
sending an IAM requesting an UDI bearer (signal 2 in figure E.2.3.1.1.1).
If SDP local preconditions, which are not yet met, are contained in signal 1, the Interworking node should immediately
send an SDP answer to allow for the IMS-side bearer set-up to progress. The Interworking node selects codecs
supported by the IM-MGW and likely to be supported within the CS network and communicates the selected codecs
towards the IMS side within an SDP answer message (signal 3 in figure E.2.3.1.1.1). If theses codecs are contained in
the SDP offer, the Interworking Node should select the H.263 codec and may select other codec from the SDP offer in
addition. The interworking node should include a b:AS bandwidth modifier with a bandwidth suitable for the selected
codec(s), but not higher than 64kbit/s plus RTP/UDP/IP overhead, in the SDP answer to request that the peer does not
send media with a higher bandwidth.
The Interworking Node shall engage in an H.223 bearer setup (step 6 in figure E.2.3.1.1.1). If the interworking Node
supports MONA (Media Oriented Negotiation Acceleration), it shall first attempt a MONA Channel establishment
method negotiation according to Annex K of ITU-T Recommendation H.324 [81]. If the interworking node does not
support MONA, it shall use the multiplexing level negotiation procedures of Annex C of H.324 [81]. If the
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Interworking Node supports MONA, but the remote peer does not do so, a fallback to the multiplexing level negotiation
procedures of Annex C of H.324 [81] will occur.
If both the Interworking Node and the remote CS terminal support MONA procedures, the MONA procedures as per
ITU-T Recommendation H.324 Annex K [81] may be used to replace the H.245 negotiation (signals 7 – 9) as shown in
figure E.2.3.1.1.1.
- After the completion of the H.223 bearer setup at the CS side, the Interworking Node shall send a Terminal
Capability Set message describing its own capabilities (signal 7 in figure E.2.3.1.1.1). Unless the Interworking
Node supports transcoding, the Interworking Node shall only send codecs that have been offered at the IM CN
subsystem side (as received in signal 1 in figure E.2.3.1.1.1) within this message.
- The Interworking Node will receive an H.245 Terminal Capability Set message describing the supported Codecs
at the peer's side (signal 8 in figure E.2.3.1.1.1).
- The codecs contained both in the sent and received terminal capability set messages may be selected at the CS
side. The final decision of the selected codecs at the CS side is taken when the H.245 open logical Channels
message (signal 9 in figure E.2.3.1.1.1) is sent or received. The direction of this message is determined by the
H.245 master-slave determination procedure.
If the Interworking Node does not transcode, it should indicate the codecs selected within the H.245 negotiation
(signal 11 in figure E.2.3.1.1.1) or within the MONA procedures and enable any media that have previously been put on
hold at the IMS side after the completion of the H.245 negotiation or MONA procedures.
The interworking node should include in step 11 of figure E.2.3.1.1.1 the SDP ‘a’ attribute "3gpp_MaxRecvSDUSize"
indicating the maximum SDU size of the application data that can be transmitted to the receiver without segmentation,
as specified in clause 12.2.4.6 of 3GPP TS 26.114 [104].
Figure E.2.3.2.1.1 assumes that the IMS peer does not use the SIP precondition extension.
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Interworking Node
IMS (MGCF+ IM MGW) CS Network
1. SIP: INVITE
[SDP offer:,
m = H.263, MP4V-ES, H.261;
m= AMR]
2. IAM
3. Bearer Establishment
4. ANM
Figure E.2.3.2.1.1 Interactions between H.245 and SIP/SDP for IM CN subsystem originated session
IMS peer does not use SIP preconditions.
Upon receipt of a SIP INVITE request containing speech and video Codecs (signal 1 in figure E.2.3.2.1.1) the
Interworking Node (consisting of MGCF and IM-MGW) starts the call set-up for multimedia call at the CS side by
sending an IAM requesting an UDI bearer (signal 2 in figure E.2.3.2.1.1).
If no unmet local SDP preconditions are contained in signal 1, the Interworking node should defer sending an SDP
answer until the H.245 negotiation or MONA procedures is/are completed.
The Interworking Node shall engage in an H.223 bearer setup (step 5 in figure E.2.3.2.1.1). If the interworking Node
supports MONA (Media Oriented Negotiation Acceleration), it shall first attempt a MONA Channel establishment
method negotiation according to Annex K of ITU-T Recommendation H.324 [81]. If the interworking node does not
support MONA, it shall use the multiplexing level negotiation procedures of Annex C of H.324 [81]. If the
Interworking Node supports MONA, but the remote peer does not do so, a fallback to the multiplexing level negotiation
procedures of Annex C of H.324 [81] will occur.
If both the Interworking Node and the remote CS terminal support MONA procedures, the MONA procedures as per
ITU-T Recommendation H.324 [81] Annex K may be used to replace the H.245 negotiation (signals 6 – 8) as shown in
figure E.2.3.2.1.1.
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- After the completion of the H.223 bearer setup at the CS side, the Interworking Node shall send a Terminal
Capability Set message describing its own capabilities (signal 6 in figure E.2.3.2.1.1). Unless the Interworking
Node supports transcoding, the Interworking Node shall only send codecs that have been offered at the IM CN
subsystem side (as received in signal 1 in figure E.2.3.2.1.1) within this message.
- The Interworking Node will receive an H.245 Terminal Capability Set message describing the supported Codecs
at the peer's side (signal 7 in figure E.2.3.2.1.1).
- The codecs contained both in the sent and received terminal capability set message may be selected at the CS
side. The final decision of the selected codecs at the CS side is taken when the H.245 open logical Channels
message (signal 8 in figure E.2.3.2.1.1) is sent or received. The direction of this message is determined by the
H.245 master-slave determination procedure.
If the Interworking Node does not transcode, it shall send an SDP answer (signal 9 in figure E.2.3.2.1.1) indicating the
codecs selected within the H.245 negotiation or within the MONA procedures after the completion of the H.245
negotiation or MONA procedures. The interworking node should include an b:AS bandwidth modifier with a bandwidth
suitable for the selected codec(s), but not higher than 64kbit/s plus RTP/UDP/IP overhead, in the SDP answer to request
that the peer does not send media with a higher bandwidth.
The interworking node should include in Step 9 of figure E.2.3.2.1.1 the SDP ‘a’ attribute "3gpp_MaxRecvSDUSize"
indicating the maximum SDU size of the application data that can be transmitted to the receiver without segmentation,
as specified in clause 12.2.4.6 of 3GPP TS 26.114 [104].
If in a non-SCUDIF case (ISUP or BICC without SCUDIF) the called CS terminal or network rejects the video call
setup by sending a REL message to the MGCF, the MGCF releases the CS video call being established, re-establishes
the CS call in a speech only mode sending a new IAM with a speech BCIE to the CS network and updates the IM CN
leg codecs to a speech only codec. Then the call/session continues as in a speech only case. If the interworking node
does not support the fallback, it shall release the session.
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Interworking Node
CS Network (MGCF+ IM MGW) IMS
2. SIP: INVITE
1. IAM [SDP offer:,
m = H.263, MP4V-ES;
m= AMR
3. SIP:
[SDP answer:
4. Bearer Establishment m = H.263, MP4V-ES;
m= AMR]
14. SIP :
[SDP answer
m = H.263;
m= AMR]
Figure E.2.4.1.1: Interactions between H.245 and SIP/SDP for CS network originated session
Upon receipt of an IAM request for a multimedia Call (signal 1 in figure E.2.4.1.1) the Interworking Node (consisting
of MGCF and IM-MGW) starts the call set-up for multimedia call at the IM CN subsystem side by sending an INVITE
request (signal 2 in figure E.2.4.1.1). For the INVITE request, the Interworking Node selects codecs supported by the
IM-MGW and likely to be supported within the CS Network. The Interworking Node should select the H.263 codec and
may select other codec in addition. The interworking node should add an b:AS bandwidth modifier with a bandwidth
suitable for the selected codec(s), but not higher than 64kbit/s plus RTP/UDP/IP overhead, in the SDP offer to request
that the peer does not send media with a higher bandwidth.
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NOTE: The SDP coding to express that either a combined voice and video call, or a voice call, or a Clearmode
codec, or some other data call is desired is not defined in the present release.
The Interworking Node shall engage in an H.223 bearer setup (step 7 in figure E.2.4.1.1). If the interworking Node
supports MONA (Media Oriented Negotiation Acceleration), it shall first attempt a MONA Channel establishment
method negotiation according to Annex K of ITU-T Recommendation H.324 [81]. If the interworking node does not
support MONA, it shall use the multiplexing level negotiation procedures of Annex C of H.324 [81]. If the
Interworking Node supports MONA, but the remote peer does not do so, a fallback to the multiplexing level negotiation
procedures of Annex C of H.324 [81] will occur.
If both the Interworking Node and the remote CS terminal support MONA procedures, the MONA procedures as per
ITU-T Recommendation H.324 [81] Annex K may be used to replace the H.245 negotiation (signals 8, 11 and 12) as
shown in figure E.2.4.1.1. Furthermore, the SIP codec renegotiation in signals 9 and 10 is then also not applicable.
- After the completion of the H.223 bearer setup at the CS side the Interworking Node will receive a H.245
Terminal Capability Set message describing the supported Codecs at the peer's side (signal 8 in figure E.2.4.1.1).
- Due to information received in a Terminal Capability Set message (signal 8 in figure E.2.4.1.1), the Interworking
node may send an SDP offer at the IMS side (signal 9 in figure E.2.4.1.1), to offer additional codecs supported at
the CS side but not contained in the first SDP offer (signal 2 in figure E.2.4.1.1), or to restrict the selected codecs
at the IMS side to codecs which are available at the CS side.
NOTE: It is not clear if the addition of codecs not included in previous SDP exchange has any impacts on IMS
procedures, e.g. resource reservation related procedures.
- The Interworking Node shall send a Terminal Capability Set message describing its own capabilities (signal 11
in figure E.2.4.1.1). Unless the Interworking Node supports transcoding, the Interworking node shall only send
codecs that are also negotiated at the IM CN subsystem side (as received in signal 3 in figure E.2.4.1.1) within
this message. The Interworking Node may defer sending the Terminal Capability Set message for some time to
attempt to receive the peer's Terminal Capability set message and perform a possible IMS-side codec re-
negotiation. However, to avoid blocking situations, the Interworking Node shall not defer sending the Terminal
Capability Set message for an excessive period of time.
- The codecs contained both in the sent and received Terminal Capability Set message may be selected at the CS
side. The final decision of the selected codecs at the CS side is taken when the H.245 open logical Channels
message (signal 12 in figure E.2.4.1.1) is sent or received. The direction of this message is determined by the
H.245 master-slave determination procedure.
If the Interworking Node does not transcode, it should indicate the codecs selected within the H.245 negotiation or
within MONA procedures after the completion of the H.245 negotiation (signal 13 in figure E.2.4.1.1) or MONA
procedures.
The interworking node should include in Step 11 of figure E.2.4.1.1 the SDP ‘a’ attribute "3gpp_MaxRecvSDUSize"
indicating the maximum SDU size of the application data that can be transmitted to the receiver without segmentation,
as specified in clause 12.2.4.6 of 3GPP TS 26.114 [104].
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If the interworking node A does not support CS-IMS video interworking, but supports a clearmode codec / clear
channel, it sends the INVITE message with a clearmode codec and UDI & H.223 & H.245 indication, but without a
video codec, to allow the establishment of a CS video call through a clear channel. The interworking node A may also
send an audio codec (alone or with a clearmode codec) to allow a fallback to speech. The interworking node B either
accepts the clearmode and sends the corresponding IAM message with a UDI & H.223 & H.245 request (message 3)
and SIP response with a clearmode codec (message 5 or 11), or accepts the speech mode and sends the corresponding
IAM message with a speech request (message 3) and SIP response with a speech codec (messages 5, 11), or rejects the
INVITE message if the requested codec(s) cannot be supported.
NOTE: The format of the indication of UDI & H.223 & H.245 from interworking node A to interworking node B
is not defined in the present release.
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7. ACM
8. SIP :180 Ringing
9. ACM
10. ANM
11. SIP:200 OK(INVITE)
12. ANM
13. SIP: ACK
Figure E.2.5.2.1.1.1.1 shows an IM CN subsystem originated modification from multimedia to speech during an
ongoing session when the CS leg supports BICC. The interworking node receives an INVITE message that indicates the
dropping of the video media from the session, message 1. The interworking node can only accept the dropping of the
media component and sends a corresponding codec modification request to the BICC network, message 2, and
acknowledges the INVITE with a 200 OK message. The BICC network indicates a successful codec modification,
message E.2.
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Audio + Video
Speech
Editor's note: Handling of a case, where a codec is received in BICC negotiation but not included in the available
codec list negotiated previously, is ffs.
Figure E.2.5.2.1.1.2.1 shows an IM CN subsystem originated modification from speech to multimedia during an
ongoing session when the CS leg supports BICC. The interworking node receives an INVITE message that offers the
adding of a video media to the ongoing speech session, message 1. The interworking node accepts the offer and sends a
corresponding codec modification request to the BICC network, message 2. The BICC network indicates a successful
codec modification, message 3. The interworking node acknowledges the INVITE with a 200 OK message after the
MONA or H.245 in-band negotiation in step 4 is completed.
If the codec modification is not successful in the BICC network, the interworking node responds to the INVITE
message with the speech codec in the 200 OK message to retain the speech only session.
Speech
Audio + Video
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Figure E.2.5.2.1.2.1.1 shows a CS network originated modification from multimedia to speech during an ongoing
session when the CS leg supports BICC. The interworking node receives a Modify Codec message that indicates the
dropping of the video media from the session, message 1. The interworking node accepts the dropping of the video
component and sends a corresponding INVITE message to the IM CN subsystem, message 2, and acknowledges the
codec modification request to the BICC network, message 3. The IM CN subsystem acknowledges the INVITE
dropping the video media with a 200 OK, message 4.
Audio + Video
Speech
Figure E.2.5.2.1.2.2.1 shows a CS network originated modification from speech to multimedia during an ongoing
session when the CS leg supports BICC. The interworking node receives a Modify Codec message that indicates the
adding of a video media to the ongoing speech session, message 1. The interworking node accepts the offer and sends a
corresponding INVITE message to the IM CN subsystem, message 2. The IM CN subsystem acknowledges the INVITE
adding the video media with a 200 OK, message 3, and acknowledges the codec modification request to the BICC
network, message 4. The interworking node may have to update the codecs, messages 7 and 8, after the MONA or
H.245 in-band negotiation in step 6.
If the IM CN subsystem does not accept the addition of the video media to the session, the interworking node rejects the
modify codec request to retain the speech only session.
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Speech
Audio + Video
- The video component stays on in the CS leg. The interworking node may use the video component to send an
announcement to the CS terminal to inform the user about the change of the end-to-end connection to audio only.
Refer to figure E.2.5.2.2.1.1.
- The interworking node initiates an H.245 in-band negotiation to close the video channel.
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Audio + Video
Audio
Video
Speech
3. SIP: ACK
Speech
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NOTE: A call release using only ISUP/BICC signalling at the CS side proceeds faster. H.324 terminals can
handle situations where they do not receive H.245 call release signalling (see Clause 7.5.2 of ITU-T
Recommendation H.324 [81]), and this scenario also occurs e.g. at loss of coverage or when a node
transporting H.223 transparently releases the call.
The procedure of ending the H.245 session is defined in the clause E4. After receiving the BYE message, the MGCF
shall also send a 200 OK [BYE] message (signal 2 in figure E.2.6.1.1) towards the IM CN subsystem.
After ending the H.245 session, the MGCF shall send a REL message (signal 4 in figure E.2.6.1.1) to the succeeding
node. If the IM CN subsystem interworks with ISUP based CS network, the interworking node shall release the
resources for the IMS side and the CS network side (signal 6 in figure E.2.6.1.1) after sending the REL message. If the
IM CN subsystem interworks with BICC based CS network, the interworking node shall release the resources for the
IMS side and the CS network side (signal 6 in figure E.2.6.1.1) upon receiving the RLC message (signal 5 in
figure E.2.6.1.1) from the CS network side. The procedures of releasing the resources for the IMS side and the CS
network side are specified in the present TS in clause 7.
Figure E.2.6.1.1 shows the message sequence chart for the multimedia call release initiated from the IM CN subsystem
side.
Interworking node
IMS (MGCF+ IM-MGW)
CS Network
1. SIP: BYE
2. SIP: 200 OK [BYE]
3. H.245 Signalling to end the H.245 session
between the IM-MGW and the CS network side
5. BICC: RLC
7. ISUP: RLC
NOTE: Signal 7 is omitted when IM CN subsystem interworks with BICC based CS network and Signal 5 is
omitted when IM CN subsystem interworks with ISUP based CS network.
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When the MGCF receives a REL message (signal 2 in figure E.2.6.2.1) from the preceding node, the MGCF sends a
BYE message (signal 3 in figure E.2.6.2.1) to the IM CN subsystem. After receiving the REL message, the
interworking node also releases the resources for the IMS side and the CS network side (signal 4 in figure E.2.6.2.1).
The procedure of the releasing the resources for the IMS side and the CS network side are specified in the present TS in
clause 7. After completion of resource release, the MGCF sends a RLC message (signal 5 in figure E.2.6.2.1) towards
the preceding node. Figure E.2.6.2.1 shows the message sequence chart for the multimedia call release initiated from
the CS network side.
Interworking node
IMS (MGCF+ IM- MGW) CS Network
NOTE: A Call Release using only SIP and ISUP/BICC signalling may proceed faster. H.324 terminals can handle
situations where they do not receive H.245 call release signalling (see Clause 7.5.2 of ITU-T
Recommendation H.324 [81]), and this scenario also occurs e.g. at loss of coverage or when a node
transporting H.223 transparently releases the call.
To release the call, the MGCF shall send a REL message (signal 2 in figure E.2.6.3.1) to the succeeding node on the CS
network side. The MGCF shall also send a BYE message (signal 3 in figure E.2.6.3.1) to the IM CN subsystem side.
If the IM CN subsystem interworks with ISUP based CS network, the interworking node shall release the resources for
the IMS side and the CS network side (signal 5 in figure E.2.6.3.1) after sending the REL message. If the IM CN
subsystem interworks with BICC based CS network, the interworking node shall release the resources for the IMS side
and the CS network side upon receiving the RLC message (signal 4 in figure E.2.6.3.1) from the CS network side. The
procedures of releasing the resources for the IMS side and the CS network side are specified in the present TS in clause
7. Figure E.2.6.3.1 shows the message sequence chart for the multimedia call release initiated from the interworking
node.
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Interworking node
IMS (MGCF+ IM-MGW)
CS Network
6. ISUP: RLC
7. SIP: 200 OK [BYE]
NOTE: Signal 6 is omitted when IM CN subsystem interworks with BICC based CS network and Signal 4 is
omitted when IM CN subsystem interworks with ISUP based CS network.
At the CS side, the IM-MGW needs to terminate the H.223 protocol and multiplex / de-multiplex audio, video and
H.245 signalling. How H.245 related information (e.g. H.245 messages or extracted information) is communicated
between the MGCF and the IM-MGW is described in Clause E4.
E.4.1 Introduction
This clause describes requirements for extensions to the Mn interface protocol in 3GPP TS 29.332 [15] needed to
support the Interworking of multimedia calls. ITU-T Recommendation H.248.1 [2] is used at the Mn interface.
The H.245 signalling shall be handled by the MGCF. Upon reception of the H.245 messages from the CS side at the
IM-MGW, the IM-MGW shall forward those H.245 messages as binary data within H.248 messages over the Mn
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interface towards the MGCF. Upon reception of encapsulated binary H.245 messages within H.248 messages, the IM-
MGW shall forward those H.245 messages towards the CS side.
NOTE: Procedures to support MONA (see Annex K of ITU-T Recommendation H.324 [81]) over the Mn
interface are not defined in the present Release. Furthermore, the signalling flows in Clause E.2 may not
show MONA related signalling in sufficient detail for MONA related Mn interface interactions.
Termination T3
Context C1
Stream 3
Stream 1 Stream 2
Termination T1 Termination T2
CS-Domain IMS
Termination:
T1 CS-Domain (CS-Bearer (BS30) for H.245 control, Speech, Video)
T2 Multiplexing (H.245 control, Speech, Video)
T3 Video (own RTP-stream) + Speech (own RTP-stream)
Stream:
Stream1 (between T1 and T2) data (H.245 control, speech, Video)
Stream2 (between T2 and T3) Video
Stream3 (between T2 and T3) Speech
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E.4.2.3.1 General
H.245 messages shall be transported between the IM-MGW and MGCF over the Mn interface using H.248 Events
(from the IM-MGW towards the MGCF) and H.248 Signals (from the MGCF towards the IM-MGW). The
Events/Signals shall contain the following information:
IM MGW MGCF
1. H.248: MOD.req
[C=C1,T=T2,
signal=H245msgout
{h245mc=XXXX} ]
Signal
H245
2. H.248: MOD.resp Message
3. send H.245 message
.
XXXX in H.223 stream
In Signal 1, the MGCF requests the IM-MGW to send an H.245 message to the CS side. To request the IM-MGW to
send a H.245 message to the CS side, the MGCF shall sent an H.248 signal to the IM-MGW with the complete H.245
message content.
NOTE: In order for this command to succeed, Termination T1 towards the CS side needs to be configured. If a
sending of an H.245 message and a removal of termination T1 is desired, the MGCF needs to apply signal
1 before removing T1.
Upon reception of this signal, the IM-MGW shall send the encapsulated H.245 message within the H.248 signal,
through the H.245 control channel to the CS side (signal 3).
IM MGW MGCF
1. H.248: ADD.req
[C=C1,T= ? ... ,Event=H245msgin] Add
2. H.248: ADD.resp Multiplex
Termination
3. receive H.245
.
message XXXX in H.223
H.248 : Notify.req
,
4.
[C=C1,T=T2
Event=H245msgin
{h245mc=XXXX}] Notify
H245
5. H.248 : Notify.resp
Message
In signal 1, the MGCF requests the IM-MGW to detect received H.245 message from the CS side and forward them to
the MGCF. To request the IM-MGW to detect and forward a H.245 message to the CS side, the MGCF shall send a
suitable H.248 event to the IM-MGW. The event may be indicated through an H.248 ADD command.
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In signal 3, the IM-MGW receives an H.245 message from the CS side. Upon reception of an H.245 message from the
CS side, the IM-MGW shall de-multiplex the H.245 message from the H.223 stream and forward the H.245 message to
the MGCF within an H.248 Notify command (signal 4).
- Request for events for Notification of H.245 message received by the IM-MGW.
- Signals to provide H.245 messages that the IM-MGW shall send towards the CS side.
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IM MGW MGCF
1. H.248: ADD.req
[C=?, T=?, stream=1]
2. H.248: ADD.resp Prepare Bearer or
[C=C1, T=T1] Establish Bearer or
Reserve TDM Circuit
3. H.248: ADD.req
[C=C1, T=?, stream=2{Codec=H.263},
stream=3{Codec=AMR}] Reserve IMS
Connection Point or
4. H.248: ADD.resp
[C=C1, T=T3] Reserve IMS
Connection Point
and Configure
6. H.248: ADD.req Remote Resources
5. Bearer Establishment
[C=C1, T=?, Mux=H223,T1,
LocalControl{ muxlv=2}
Event=H245msgin ] Add
7. H.248: ADD.resp Multiplex
[C=C1, T=T2] Termination
8. H.223: Multiplexing Level Negotiation
[level =2]
NOTE 1: All H.245 messages (from Signal 9 to Signal 18) are transported through the IM-MGW between the MGCF
and the CS side, using the procedures described in Clause E.4.2.3
NOTE 2: Signals 21 and 22 are omitted if the same codec information has already been provisioned in signal 3.
The MGCF shall request terminations towards the CS network (Signal 1 and 2) and towards the IMS (Signal 3 and 4).
For the terminations towards the IMS, the MGCF provides an estimate about the applicable codecs in the required
information elements "Local IMS Resources" (for both "Reserve IMS Connection Point" procedure and "Reserve IMS
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Connection Point and Configure Remote Resources" procedure) and possibly "Remote IMS Resources" (only for
"Reserve IMS Connection Point and Configure Remote Resources" procedure).
The MGCF shall request that the H.223 stream is (de-)multiplexed at the MUX termination T2, and that the H.245
control in H.223 Logical channel 0 is separated (signal 6). Furthermore, the MGCF shall request that the H.223
negotiation is started, and shall request to be notified about all H.245 messages received by the IM-MGW.
The IM-MGW shall start the H.223 Multiplexing Level Negotiation after receipt of the corresponding request from the
MGCF and CS bearer establishment (Signal 8).
Upon reception of a H.245 Terminal Capability Set message (Signal 9), the MGCF sends a H.245 Acknowledgment
message (Signal 10).
The MGCF shall know the H.324 related capabilities of the IM-MGW before starting the H.245 capability negotiation
with the CS side, e.g. through configuration. The H.245 Terminal Capability Set message send by the MGCF (Signal
11) should include the codecs which are supported by both the IMS side and the IM-MGW, and the codecs which could
be transcoded by the IM-MGW from the codecs supported by the IMS side.
The MGCF may defer sending the Terminal Capability Set message (Signal 11) for some time to wait for codec
information from the CS peer's Terminal Capability Set message and perform a possible IMS-side codec re-negotiation.
To avoid blocking situations, the MGCF shall not defer sending the signal for an excessive period of time.
To avoid the CS side selecting the codecs that need to be transcoded at the IM-MGW, the MGCF should aim to be the
master in the H.245 master-slave determination procedure (Signals 9 to 12). The MGCF shall set the Terminal Type
parameter as a number larger than 128 in the H.245 Master Slave Determination message. The H.245 master-slave
determination procedure could be combined with the messages used for the H.245 capability exchange.
The codecs contained both in the sent and received terminal capability set message may be selected at the CS side. The
final decision of the selected codecs at the CS side is taken with the H.245 open logical channel procedure (Signals 13
to 16).
After the completion of the H.245 multiplex table exchange procedure (Signals 17 to 20), the MGCF shall configure the
multiplexing termination T2 by indicating to the IM-MGW the contents of the incoming and outgoing multiplex tables
(Signal 21).
If codec information needs to be changed compared to what has been provisioned in signal 3, the MGCF shall also
configure T3 with the appropriate video and/or speech codec(s) (signal 23).
E.4.2.5.1 Overview
The MGCF shall support the following H.245 indication messages: Function Not Understood Indication / Function Not
Supported Indication, Jitter Indication. The MGCF may support the H.245 User Input Indication message. All these
H.245 messages are conveyed between the MGCF and the CS side through the IM-MGW, as described in clause
E.4.2.3.
If the MGCF receives a Function Not Understood or Function Not Supported message from the CS side, the MGCF
may choose to attempt simpler H.245 interaction than the previous H.245 interaction that caused this indication. If this
is not possible, the MGCF may release the call.
If the MGCF receives a H.245 request, response or command that can not be understood, the MGCF shall send H.245
Function Not Supported indication message to the CS side.
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The MGCF and IM-MGW may support transporting the DTMF information both from the CS side to the IMS side, and
from the IMS side to the CS side.
Upon Receipt of a H.245 User-Input-Indication message, the MGCF may apply the procedures in Clause 9.2.8.3 to
request the IM-MGW to send corresponding RTP telephone-event(s) towards the IMS side.
Upon receipt of RTP telephone events from the IMS, the MGCF will be notified by the IM-MGW using the procedures
in Clause 9.2.8.1, if the MGCF has previously configured the IM-MGW as also described in this Clause. Upon receipt
of the notification, the MGCF may send a H.245 User-Input-Indication message to the CS side.
E.4.2.6.1 Overview
The MGCF shall support the End Session command message. The MGCF may support the Flow Control command
message and the videoFastUpdatePicture message. All these H.245 messages are conveyed between the MGCF and the
CS side by the IM-MGW, as described in clause E.4.2.3.
NOTE: It is very unlikely that the MGCF will receive a Flow control command, as the capability of controlling
the video bit rate during a video call was neither implemented nor tested for UEs offering 3G-324M since
R99, due to the lack of such a necessity in WCDMA where a fixed 64 kbps bearer is maintained to
continuously transport 3G-324M PDUs.
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I M MGW MGCF
2.H.248: Modify.req
C=C1,T=T2,stream{LCN}, stream mode=Recieve Only
3.H.248: Modify.rsp
4. H.245: Close Logical Channel{Old LCN}
Open Logical Channel {New LCN}
5. H.245: Close Logical Channel ACK {Old LCN}
Open Logical Channel ACK{New LCN}
6.H.248: Modify.req
C=C1,T=T2,stream{New LCN, codec}, stream mode = sendreceive
7.H.248: Modify.rsp
9.SIP:reINVITE
12.H.248: Modify.rsp
In Signal 1, the MGCF receives the Flow Control Command from the CS side.
If the minimum bitrate of the current codec is larger than the bitrate requested by the H.245 Flow Control Command
message, the MGCF indicates the IM-MGW to stop the transmission of the media stream over the logical channel
(signal 2). Then the MGCF may select another codec that can satisfy the requested bitrate limit. In signal 4, the MGCF
closes the old logical channel and opens a new logical channel with new codec to satisfy the bitrate limit in the CS side.
In signal 6, the MGCF indicates the IM-MGW to modify the LCN, codec and stream mode of the multiplexing
termination. In signal 8, the MGCF sends flow control indication message to CS side with the current maximum bitrate.
If the MGCF chooses to change the CS-side codec, the MGCF may also adjust the codec at the IMS side accordingly.
To do so, the MGCF may need to re-negotiate the codec at the IMS side using a SIP re-INVITE message (signals 9 and
10). In addition, the MGCF may modify the codec of IMS termination accordingly (signal 11).
The MGCF may also apply interworking procedure towards RTCP at the IMS side as described in Clause E.4.2.8 when
receiving a H.245 Flow Control Command.
The MGCF may send an end session command to the CS side through the IM-MGW to release a call.
If the MGCF receives an end session command from the CS side, it shall release the call if the call is in the active state.
E.4.2.6.4 videoFastUpdatePicture
The MGCF may apply interworking procedure towards RTCP as described in Clause E.4.2.8 when receiving a H.245
videoFastUpdatePicture message. The MGCF may also be triggered by the procedures in Clause E.4.2.8 to send a
H.245 videoFastUpdatePicture message.
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When receiving frequent H.245 videoFastUpdatePicture messages for a SCUDIF call (see 3GPP TS 23.172 [118]), an
MGCF should initiate a service-change from multimedia to speech and remove the IMS MTSI video component with
SIP/SDP signalling. When receiving frequent H.245 videoFastUpdatePicture messages for a non-SCUDIF call, an
MGCF should disable the CS video and remove the IMS MTSI video component with SIP/SDP signalling.
NOTE: In a congestion situation at the IMS side, frequent packet losses might occur and dropping the video
component may serve as a counter-messure to reduce the packet loss frequency. However, CS terminals
might not be able to initiate such a switch-over to voice, but might rather react with repeated H.245
videoFastUpdatePicture message.
E.4.2.7.1 Overview
Media Oriented Negotiation Acceleration (MONA), as specified in ITU-T H.324[81] provides simplified procedures
that allow for a faster call set-up of a H.324 Multimedia call than standard H.324 procedures, and also allow for a
fallback to standard H.324 procedures if either party does not support the enhanced procedures.
The support of MONA is optional for an IM-MGW and MGCF supporting multimedia interworking, as no call failure
but only a fallback to standard H.324 setup procedures will occur if the procedures are not supported.
MONA "preference message" signalling is used instead of H.324 Multiplexing level negotiation. Should standard H.324
Multiplexing level stuffing flags be received, a fallback to standard H.324 procedures is triggered. The sending of
MONA preference messages is repeated by each MONA capable H.324 terminal until a reception is acknowledged by
the peer. During this phase, two PDU types may optionally be attached by MONA terminals to these preference
messages:
- Media Preconfigured Channel (MPC) PDUs: MONA defines a small number of preconfigured H.223 channels
for the most widespread audio and video codecs (AMR, AMR-WB, H.264 MPEG4 and H.263). Media PDUs for
these codecs may be attached to the MONA "preference message" during the call setup.
- Signalling Preconfigured Channel (SPC) PDUs: These PDUs are H.245 generic request messages with special
parameters defined by MONA. These PDUs may also be attached to MONA preference messages.
According to MONA, each MONA capable terminal shall support at least one of these PDU types. The MONA
capability of the IM-MGW can be audited by the MGCF.
The MONA preference message exchange in combination with attached MPC or SPC PDUs may result in the
establishment of the desired media channels without further H.245 signalling. Otherwise, H.245 will be used after the
MONA preference message exchange is acknowledged to negotiate media channels, but MONA defines some
accelerated H.245 procedures (ACP) to speed up these H.245 procedures.
- The H.245 handling should be performed in the MGCF to keep procedures aligned as far as possible with
standard Mn procedures to support H.324 interworking.
- The MGCF should also control MONA preference message exchange procedures in order to maintain the agreed
architectural work split between MGCF and IM-MGW in analogy to the H.245 handling.
- However, the IM-MGW needs to understand the MONA preference messages at least to a sufficient degree to
de-encapsulate the possibly attached MPC and SBC PDUs.
- Furthermore, the frequent retransmissions of MONA preference messages required by MONA procedures are to
be performed by the IM-MGW autonomously to avoid unnecessary load at the Mn interface and the MGCF.
- Furthermore, for resource reservation at the IM-MGW, it is assumed that the IM-MGW has knowledge about
supported predefined Media Preconfigured Channel configurations as specified in Clause K.9.2 of H.324 [81] in
order to limit the amount of information transferred in the Mn interface when establishing MPC channels. The
offered channel resources are reserved by the IM-MGW.
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- In order to avoid increasing call establishment time when interworking with legacy terminals and at the same
time avoid unnecessary load at the Mn interface, the MGCF may initiate both MONA and legacy H.245
procedures simultaneously in parallel. The MGCF shall in this case arm a legacy detection event with an
embedded signal descriptor including the initial legacy H.245 message. The IM-MGW will only send the signal
in case a fallback condition to legacy is detected.
Editor’s note: It is FFS whether the mux-level-indication event can be used as a legacy detection event, and whether
it is more feasible than the mechanism for legacy detection described.
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IM MGW MGCF
1. H.248: ADD.req
[C=?, T=?, stream=1]
2. H.248: ADD.resp Prepare Bearer or
[C=C1, T=T1] Establish Bearer or
Reserve TDM Circuit
3. H.248: ADD.req
[C=C1, T=?, stream=2{Codec=H.263},
stream=3{Codec=AMR}] Reserve IMS
Connection Point or
4. H.248: ADD.resp Reserve IMS
[C=C1, T=T3]
Connection Point
and Configure
6. H.248: ADD.req Remote Resources
5. Bearer Establishment [C=C1, T=?, Mux=H223,T1,
LocalControl{MONA preferences, muxlv=2}
Event=legacyDetected(h245msgout[octet string]),
MONA-preference-reception, MONA-preference- Add
completed, H245msgin, Multiplex
[MPC-reception, SBC-reception] ] Termination
8. MONA preferences 7. H.248: ADD.resp
[ack=00] [C=C1, T=T2]
8. MONA preferences
[ack=00] 11. H.248 : Notify.req
,
[C=C1,T=T2
9. MONA preferences
[ack=00] Event=MONA-preference-receptiom
10. MONA preferences {MONA preferences}] Notify
[ack=01] MONA
12. H.248 : Notify.resp
prefrence
9. MONA preferences reception
[ack=00]
Optional Procedures for reception of SPC PDU , sending of SPC PDU, reception of MPC
14. MONA preferences PDU, sending of MPC PDU
[ack=10]
NOTE 1: MONA preference messages are repeated several times. One repetition is shown for each such message,
with the same signal number as the first message.
NOTE 2: The Context model in figure E.4.2.2.1 is assumed in this call flow.
Figure E.4.2.7.2.1: Mn signalling interactions for MONA preference messages
The MGCF shall request terminations towards the CS network (Signal 1 and 2) and towards the IMS (Signal 3 and 4).
For the terminations towards the IMS, the MGCF provides an estimate about the applicable codecs in the required
information elements "Local IMS Resources" (for both "Reserve IMS Connection Point" procedure and "Reserve IMS
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Connection Point and Configure Remote Resources" procedure) and possibly "Remote IMS Resources" (only for
"Reserve IMS Connection Point and Configure Remote Resources" procedure).
The MGCF shall request that the H.223 stream is (de-)multiplexed at the MUX termination T2 (signal 6). Furthermore,
the MGCF shall request that MONA preferences negotiation is started, and shall provision the MONA preferences to be
indicated by the IM-MGW. The MGCF shall encode the MONA preferences as described in Clause K.6 of ITU-T
H.324 [81]. The MGCF shall take the H.324 related capabilities of the IM-MGW into account in the MONA
preferences. The MGCF can know these capabilities by configuration. The IM-MGW will only support symmetric
codec usage. If several codec alternatives are offered for MPC, it is the responsibility of the MGCF to ensure that
symmetric codecs are established by not selecting transmit codec until the receive channel has been opened by MPC
media. The MGCF shall also request to be notified about the reception of the remote MONA preferences and about the
completion of the MONA preference exchange, or an H.245 message on the H.223 control channel. The MGCF may
also initiate standard H.245 signaling in parallel in order to minimize the time for a legacy interworking fallback. This is
done by arming a legacy detection event including an embedded signal descriptor. The embedded signal is the intial
H.245 message out signal (including H.245 TCS+MSD) to send in case fallback to legacy interworking is detected. The
IM-MGW will only send the embedded signal in case it detects H.223 related indications of a legacy interworking as
specified in Clause K.7.1.2 in H.324 [81]. Upon receiving the legacy detected event, the MGCF continues with standard
H.245 call setup procedures waiting for the reception of a remote H.245 TCS as well as acknowledgements on the sent
H.245 TCS+MSD. If the MGCF indicates the capability to receive SPC PDUs within the MONA preferences, it shall
also request to be notified about incoming SPC PDUs, as detailed in Clause E.4.2.7.4. If the MGCF indicates the
capability to receive any MPC PDUs within the MONA preferences, it shall also request to be notified about incoming
MPC PDUs, as detailed in Clause E.4.2.7.3.
The IM-MGW shall start sending MONA preference messages after receipt of the corresponding request from the
MGCF and CS bearer establishment (Signal 8). The IM-MGW shall repeat sending those messages and increment the
acknowledgment bits of sent MONA preference messages when receiving incoming MONA preference messages
according to MONA procedures (signals 8, 10, 14).
After sending at least 10 MONA preference messages, while the IM-MGW continues to send and receive MONA
preference messages, it shall attach MPC or SPC PDUs if requested to do so by the MGCF as described in Clauses
E.4.2.7.3 and E.4.2.7.4, respectively. If the IM-MGW receives preference messages with an attachment, it shall inspect
the first octet of that attachment that will contain a MUX code according to table K.15 of ITU-T H.324 [81] that
identifies the attached PDU as either a MPC PDU of one of the predefined channels or a SPC PDU. The IM-MGW shall
handle the attached MPC or SPC PDUs as described in Clause E.4.2.7.3 and E.4.2.7.4, respectively.
After sending at least 10 MONA preference messages, the IM-MGW should insert stuffing flags indicating the
multiplexing level received from the MGCF between MONA preference messages as described in Clause K.7.1.1 of
ITU-T H.324 [81].
The IM-MGW shall notify the MGCF when receiving the first incoming MONA preference message (Signals 11 and
12) and forward the received information. Subsequent incoming MONA preference message will be identical apart
from possible increments in the acknowledgement bits. The IM-MGW shall not notify the MGCF about these messages.
Upon reception of the notification of a MONA preference message, the MGCF shall compare the received MONA
preferences message with the preferences message it sent and react as described in Clause 7.1 of ITU-T H.324 [81].
When receiving an incoming MONA preference message with acknowledgment bits 10 or receiving the first non-empty
H.223 MUX PDU, the IM-MGW shall notify the MGCF about the completion of the MONA preference exchange
procedure (signals 16 and 17). The notification shall be only triggered by the one of the two events which occurs first. If
the IM-MGW is not configured to send and detect SPCs, the IM-MGW shall then stop sending MONA preference
messages and, if a MONA preference message with acknowledgment bits 10 has been received, it shall also stop
receiving MONA preference messages. If the IM-MGW is configured to send and detect SPCs, the IM-MGW shall
continue sending and receiving MONA Preference messages encapsulating the SPC/MOS messages until the MGCF
configures the IM-MGW stop sending and detecting SPCs.
NOTE: In the unlikely case that a first non-empty H.223 MUX-PDU is received before the first MONA
Preference Message has been received, the IM-MGW will continue the detection of incoming MONA
Preference Messages and will apply the Notify-MONA-preference-Reception-procedure after the Notify-
MONA-preference-completed procedure.
After receiving the notification about the completion of the MONA preference exchange procedure, and a completion of
the possible subsequent accelerated H.245 setup procedures, the MGCF shall configure the multiplexing termination T2
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by indicating to the IM-MGW the contents of the incoming and outgoing multiplex tables (Signal 19), and may modify
the selected codecs at both the MUX and the IMS side (signal 21).
IM MGW MGCF
1. H.248: ADD.req
[C=C1,T= ?
... ,Event=MPC-reception,
Signal =MPC MUX Code]} Add
2. H.248: ADD.resp Multiplex
Termination
NOTE 1: MONA preference messages are repeated several times. One repetition is shown for each such message,
with the same signal number as the first message.
NOTE 2: The Context model in figure E.4.2.2.1 is assumed in this call flow.
Figure E.4.2.7.3.1: Mn interactions for MONA MPCs
If the MGCF indicates the ability to receive any predefined MPCs channel types in the MONA preferences messages,
the MGCF shall request the IM-MGW to report the channel type of received MPC PDUs (Signal 1 in Figure
E.4.2.7.3.1, Event "MPC Reception").
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If the MGCF intends to use MPCs for sending media during the MONA setup, the MGCF shall request the IM-MGW to
send available media encoded according to the media predefined channel types defined by MONA (signal 1 in Figure
E.4.2.7.3.1, Signal "MPC MUX Code" ) while the MONA preference exchange described in Clause E.4.2.7.2 is
ongoing. The MGCF should select channel types for codecs which are supported by both the IMS side and the IM-
MGW, and/or for codecs which could be transcoded by the IM-MGW from the codecs supported by the IMS side. The
MGCF shall only include one channel type per each desired media type (audio, video) in the "MPC MUX Code" signal.
The MGCF may also configure the MGW to receive these channels at the same time by supplying the Incoming
Multiplex Table IE in the Add Multiplex Termination Procedure.
Upon reception of this request, the IM-MGW shall forward any media received from the IMS side in MPC PDUs of the
corresponding predefined channel type attached to MONA preference messages, transcoding the media if required
(Signal 4).
According to the procedures in Clause E.4.2.7.2, the IM-MGW will notify the MGCF (signal 6) when receiving the first
incoming MONA preference message (Signals 5). The MGCF shall then analyse the Media Preconfigured Channel
Receive bits within the MONA preference message and configure the MGW accordingly:
- If MPC(s) that the MGCF has previously configured within the "MPC MUX Code" signal are supported
according to these bits, the MGCF shall configure the IM-MGW to send these MPC using the predefined
channels (signal 10) by supplying the corresponding Outgoing Multiplex Table in the Configure Multiplex
Termination procedure (signal 8).
- If some of the MPC(s) that the MGCF has previously configured within the "MPC MUX Code" signal are not
supported according to these bits, the MGCF shall configure the IM-MGW to terminate the media on those
MPC(s) by modifying the MPC Mux Code signal using the Configure Multiplex Termination procedure. In
addition, the MGCF may configure the IM-MGW to send media on other supported MPC(s) by supplying the
corresponding Outgoing Multiplex Table in the Configure Multiplex Termination procedure (not shown in
Figure E.4.2.7.3.1).
- If all of the MPC(s) that the MGCF has previously configured within the "MPC MUX Code" signal are not
supported according to these bits, the MGCF shall configure the IM-MGW to terminate MPC operations in those
unsupported channels. The MGCF shall either remove the MPC Mux Code signal using the Stop MPC procedure
or configure the IM-MGW to send media on other supported MPC(s) by supplying the corresponding Outgoing
Multiplex Table in the Configure Multiplex Termination procedure (not shown in Figure E.4.2.7.3.1).
When the MGCF provides an Outgoing Multiplex Table, the IM-MGW shall terminate sending any MPC PDUs as
attachments to MONA preference message.
Note: Any previously provisioned "MPC MUX Code" signal becomes irrelevant.
When being notified about the receipt of the first incoming MONA preference message, the MGCF may also analyse
the Media Preconfigured Channel Receive bits within the MONA preference message and configure the MGW with a
suitable Incoming Multiplex Table using the Configure Multiplex Termination procedure (not shown in Figure
E.4.2.7.3.1).
If the IM-MGW receives the first MONA preference message with attached MPC PDU of a given predefined channel
type (signal 12), or if the IM-MGW receives the first non-empty H.223 MUX PDU of a given predefined channel type
which has been configured with the Incoming Multiplex Table, and the MGCF has requested a notification about such
an event, the IM-MGW shall notify the MGCF about the received channel type (signal 13). The IM-MGW shall not
notify the MGCF about subsequent receptions of MPC PDUs of the same channel type.
Upon reception of such a notification, if the IM-MGW supports the indicated channel type and has not yet been
configured to receive media of that channel type, and if the MGCF has previously indicated the capability to receive
MPCs of that channel type within MONA preference messages, the MGCF shall configure the IM-MGW to receive
media of that channel type and forward them to the IMS side by supplying the Incoming Multiplex Table IE in the
Configure Multiplex Termination procedure (Signal 15).
If the MGCF determines (e.g. after receipt of signal 5 or signal 12) that some desired media channels can not be
established using the MPC procedures, the MGCF should use accelerated H.245 procedures as defined by MONA to set
up media channels. Corresponding H.245 messages shall be transported transparently between the IM-MGW and the
MGCF using the "Signal H.245 message" and "Notify H.245 message" procedures.
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The MGCF can receive a forwarded MONA preference message from the IM-MGW when the MGCF has already
configured the IM-MGW to send media on an MPC. If the processing of this message in the MGCF leads to the
conclusion that MPC procedures need to be aborted because SPC preferred (SPP) is determined, the MGCF shall
configure the IM-MGW to terminate MPC operations by removing the MPC related event MPC-reception and MPC
Mux Code signal using the Stop MPC procedure.
E.4.2.7.4.1 General
H.245 PDUs for SPC shall be transported between the IM-MGW and MGCF over the Mn interface using H.248 Events
(from the IM-MGW towards the MGCF) and H.248 Signals (from the MGCF towards the IM-MGW). The
Events/Signals shall contain the following information:
The related procedures are distinct from the procedures in Clause E.4.2.3, since the PDUs are received or sent by the
IM-MGW using the SPC, i.e. as attachment to MONA preference messages.
If the MGCF supports SPCs, it shall comply with the SPC procedures in Clause K.8 of ITU-T H.324 [81]. The
repetition of sending the same SPCs will be handled by the IM-MGW. When the MGCF receives an acknowledgement
from the CS side it shall request the IM-MGW to stop the repetition sending of the SPCs,
Within the sent SPC PDUs, the MGCF should include the codecs which are supported by both the IMS side and the IM-
MGW, and the codecs which could be transcoded by the IM-MGW from the codecs supported by the IMS side.
NOTE 1: MONA preference messages are repeated several times. One repetition is shown for each such message,
with the same signal number as the first message.
NOTE 2: The Context model in figure E.4.2.2.1 is assumed in this call flow.
Figure E.4.2.7.4.2.1: Mn interactions for sending MONA SPCs
In Signal 1, the MGCF requests the IM-MGW to send an H.245 message to the CS side. To request the IM-MGW to
send a H.245 message to the CS side, the MGCF shall sent an H.248 signal to the IM-MGW with the complete H.245
message content.
Upon reception of this signal, the IM-MGW shall send the encapsulated H.245 message within the H.248 signal, as
attachment to a MONA preference message as described in Clause K.9.4 of ITU-T H.324 [81] (signal 3). It should
repeat sending this H.245 message as attachment to subsequent MONA preference messages.
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IM MGW MGCF
1. H.248: ADD.req
[C=C1,T= ? ... ,Event=SPC-reception] Add
2. H.248: ADD.resp Multiplex
Termination
receive H.245 message attached to
3.
MONA. preference message
H.248 : Notify.req
4. ,
[C=C1,T=T2
Event=SPCin
{h245mc=XXXX}]
Notify
5. H.248 : Notify.resp SPC
NOTE 1: MONA preference messages are repeated several times. One repetition is shown for each such message,
with the same signal number as the first message.
NOTE 2: The Context model in figure E.4.2.2.1 is assumed in this call flow.
Figure E.4.2.7.4.3.1: Mn interactions for receiving MONA SPCs
In signal 1, the MGCF requests the IM-MGW to detect received H.245 message from the CS side in SPC PDUs
attached to MONA preference messages and forward them to the MGCF. To request the IM-MGW to detect and
forward these H.245 message, the MGCF shall send a suitable H.248 event to the IM-MGW. The event may be
indicated through an H.248 ADD command.
In signal 3, the IM-MGW receives an H.245 message from the CS side attached as SPC PDU to a MONA preference
message. Upon reception of such an H.245 message from the CS side, the IM-MGW may check, based on bitwise
comparison of the previously received H.245 message, if it has already forwarded the same H.245 message to the
MGCF, in which case the IM-MGW may choose not to forward the same H.245 message to the MGCF. Otherwise the
IM-MGW shall forward the H.245 message to the MGCF within an H.248 Notify command (signal 4).
NOTE: According to H.324 [81] a MOS requestAck message shall be sent to every received MOS request. If the
IM-MGW chooses, based on the bitwise comparition, not to forward the received H.245 message to the
MGCF, no MOS requestAck message will be generated. However, the MGCF will request the IM-MGW
to automatically retransmit the MOS requestAck message generated by the MGCF itself.
If the IM-MGW does not support forwarding SPC PDUs or has not been requested by the MGCF to forward these
PDUs, it shall discard received SPC PDUs.
IM MGW MGCF
4 H.248: MOD.req
[C=C1,T=T2 Remove Signal-SPC+ SPC-reception]
NOTE 2: The Context model in figure E.4.2.2.1 is assumed in this call flow.
Figure E.4.2.7.4.4.1: Mn interactions for terminating SPC procedure.
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If the processing of an incoming MONA Preference Message in the MGCF leads to the conclusion that SPC procedures
need to be aborted as a result of the test of the SPC bits, the MGCF shall configure the IM-MGW to terminate SPC
operations by removing the SPC related event SPC-reception and SPCout signal using the Stop SPC procedure. In this
case, H.245 signaling may be started simultaneously.
If the MGCF determines the completion of the SPC procedures according to Clause K.8 of ITU-T H.324 [81] (based on
incoming signal 1 in figure E.4.2.7.4.4.1), the MGCF shall configure the IM-MGW to stop sending and detecting SPCs
(signal 4 in figure E.4.2.7.4.4.1).
E.4.2.7.5 Mn Interactions for fallback from MONA procedures to standard H.324 setup
IM MGW MGCF
1. H.248: ADD.req
[C=?, T=?, stream=1]
2. H.248: ADD.resp Prepare Bearer or
[C=C1, T=T1] Establish Bearer or
Reserve TDM Circuit
3. H.248: ADD.req
[C=C1, T=?, stream=2{Codec=H.263},
stream=3{Codec=AMR}] Reserve IMS
Connection Point or
4. H.248: ADD.resp Reserve IMS
[C=C1, T=T3] Connection Point
and Configure
6. H.248: ADD.req Remote Resources
5. Bearer Establishment
[C=C1, T=?, Mux=H223,T1,
LocalControl{MONA preferences, muxlv=2},
Event=legacyDetected(h245msgout[octet string]), Add
MONA-preference-reception, MONA-preference-
completed, H245msgin, Multiplex
MPC-reception, SBCin ] Termination
8. MONA preferences 7. H.248: ADD.resp
[ack=00] [C=C1, T=T2]
8. MONA preferences
[ack=00]
11. H.248 : Notify.req
,
9. Legacy interworking condition
[C=C1,T=T2
10. octet string (H.245 TCS+MSD)
Event=legacyDetected
Notify detection of
12. H.248 : Notify.resp
legacy interworking
13. standard H.245 call setup procedures continuation, Figure E.4.2.4.1 starting with step 9
NOTE 1: MONA preference messages are repeated several times. One repetition is shown for each such message,
with the same signal number as the first message.
NOTE 2: The Context model in figure E.4.2.2.1 is assumed in this call flow.
Figure E.4.2.7.5.1: Mn signalling interactions for fallback from MONA procedures to standard H.324
setup
When the MGCF requests that the MONA preferences negotiation is started, the MGCF may also initiate standard
H.245 signalling towards the IM-MGW that shall only be sent in case the IM-MGW detects legacy interworking. The
MGCF arms an event to detect legacy interworking with an embedded signal descriptor including an H.245 message out
signal. The embedded H.245 signal is the initial H.245 TCS+MSD signal to send in case fallback to legacy
interworking is detected. The MGCF shall also request to be notified about a H.245 message on the H.223 control
channel. The MGCF shall also provision a multiplexing level which will be advertised by the IM-MGW (Signal 6).
If the IM-MGW detects a legacy interworking condition (signal 9), it shall stop sending MONA preference messages,
including those MONA messages used to encapsulate MPC or SPC. The IM-MGW shall engage in normal H.324
multiplexing level negotiations. In case the MGCF armed a legacy detection event, the embedded H.245 signal shall be
sent by the IM-MGW (signal 10). The legacy detection event is sent to the MGCF (signal 11).
The MGCF shall upon detection of legacy interworking stop MONA procedures and continue with standard H.245 call
set up procedures, as depicted in Figure E.4.2.4.1 starting with step 9.
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If the IM-MGW receives a normal H.245 message (not depicted), it shall also forward this message to the MGCF. If the
MGCF receives such a H.245 message during the MONA call setup, and this H.245 message is a normal Terminal
Capability Set message, the MGCF shall also stop MONA procedures and continue with standard H.245 call set up
procedures, as depicted in Figure E.4.2.4.1 starting with step 9.
To stop MONA related procedures at the IM-MGW, the MGCF shall remove the MONA related signals that were
active (MONA Preferences, SPCout, MPC Mux Code; note that MONA Preference and SPCout causes the MGW to
send the MONA Preference message or the contained H.245 message several times) and events
MonaPreferenceReception, MonaPreferenceCompleted, SPC-reception and MPC-reception, using the Stop MONA
Negotiation procedure.
E.4.2.8.1 Overview
The following sub-clauses describe the interworking procedures between the RTCP messages listed in table E.4.2.8.1.1,
which are used to enhance the quality of the media distribution by MTSI terminals at the IMS side, and the
corresponding H.245 messages in 3G-324M at the CS side listed in Table E.4.2.8.1.1.
As the H.245 protocol is terminated at the MGCF and RTCP is received at the IM-MGW, Mn procedures to support the
transfer of RTCP related information are defined in the following Sub-Clauses to support this interworking.
NOTE 1: It is very unlikely that the MGCF will receive a Flow control command, as the capability of controlling
the video bit rate during a video call was neither implemented nor tested for UEs offering 3G-324M since
R99, due to the lack of such a necessity in WCDMA where a fixed 64 kbps bearer is maintained to
continuously transport 3G-324M PDUs.
NOTE 2: Terminals are expected to support receiving the FlowControlCommand, as this is a mandatory feature in
clause 6.4.3 in H.324. But most terminals would not really follow the received FlowControlCommand,
but rather ignore it. However, for older 3G-324M terminal lock-up or system crash was not uncommon
during IOTs. Therefore safe handling of this command is not guaranteed absolutely
The interworking shown in Table E.4.2.8.1.1 is only applicable if video is relayed without transcoding.
If transcoding is applied, the IM-MGW should adopt its transcoder accordingly if receiving the RTCP messages listed
in Table E.4.2.8.1.1, and the MGCF should configure the transcoder in the IM-MGW accordingly if it receives the
H.245 messages in Table E.4.2.8.1.1.
For an MGCF and IM-MGW that support video interworking as specified in the present Annex E, the support of the
procedures in the present Clause E.4.2.8.2 and E.4.2.8.3 is optional.
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AVPF Temporary Maximum Media Bit-rate Request (TMMBR) packet is detected for interworking, the MGCF needs
to configure a combination of PT=205 (transport layer feedback message) and FMT=3 (see IETF RFC 5104 [110]).
A IM-MGW configured in this way shall check after receiving an incoming RTCP Packet if it is of a desired type, as
expressed by a bit pattern received from the MGCF. An incoming RTCP message can be of compound format and
contain several RTCP packets. The IM-MGW shall then perform the check separately for each packet. If the IM-MGW
determines that the received RTCP packet is of a desired type, the IM-MGW shall send information derived from the
RTCP message as ObservedEventsDescriptor Parameter(s) of event "RTCP-interworking" in an H.248 "Notify"
message to the MGCF.
The information derived from the RTCP message to be sent to the MGCF is listed in table E.4.2.8.2.1. Information
elements in table E.4.2.8.2.1 shall only be sent if the corresponding RTCP message has been received.
Figure E.4.2.8.2.2.1 shows examples of interactions between RTCP and H.245 for IM CN subsystem originated
feedback on the quality of the media distribution.
MGCF
IMS IM MGW
CS Network
2. H.248: MOD.resp
3. receive RTCP feedback messages
6. H.248: Notify.resp
Figure E.4.2.8.2.2.1: Interactions between RTCP and H.245 for IM CN subsystem originated RTCP
feedback on the quality of the media distribution
In signal 1, the MGCF requests the IM-MGW to detect received RTCP feedback message from the IMS side and notify
the feedback information elements to the MGCF when the interworking is required. To request the IM-MGW to detect
and notify these feedback information elements, the MGCF shall send the "RTCP-interworking" H.248 event to the IM-
MGW. The event may be indicated through an H.248 MOD command.
In signal 3, the IM-MGW receives a RTCP feedback message from the IMS side.
In signal5, the IM-MGW notifies the feedback information elements from RTCP message to the MGCF.
In signal7, the MGCF send the corresponding H.245 message to the CS side to request for the media adaption.
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MGCF
IMS IM MGW
CS Network
4. H.248: MOD.resp
Figure E.4.2.8.3.2.1: Interactions between H.245 and RTCP for CS network originated H.245 feedback
on the quality of the media distribution
In signal 1, the MGCF receives an H.245 feedback message from the CS side.
In signal3, the MGCF requests the IM_MGW to send an RTCP message provisioning the signal "H.245 interworking"
with parameters identifying information to be transported within this RTCP message.
In signal5, the IM-MGW send the corresponding RTCP message to the IMS side to request for the media adaption.
To request the IM-MGW to send an RTCP message, the MGCF shall send the corresponding information elements
listed in table E.4.2.8.3.2.1 as parameters of the "H.245-interworking signal" over the Mn interface.
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E.4.3.14 Signal-H.245-Interworking
This procedure is used by the MGCF to indicate the IM-MGW the feedback information elements when a H.245
feedback message on the quality of the media distribution requiring interworking is received.
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Annex F (normative):
PSTN XML Scheme
F.1 Scope
This section defines the PSTN XML Schema to be used for providing the BearerCapability, Low Layer Compatibility,
High Layer Compatibility and Progress indicator embedded as body in SIP messages.
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If the XML scheme is embedded in SIP messages as body, the Content-Type header shall be set to
"application/vnd.etsi.pstn+xml" and the Content-Disposition shall be set to "signal" with the "handling" parameter set to
"optional".
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<xs:complexType name="BCOctet4-1Type">
<xs:sequence>
<xs:element name="RateMultiplier" type="SevenBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="BCOctet5Type">
<xs:sequence>
<xs:element name="Layer1Identification" type="TwoBitType"/>
<xs:element name="UserInfoLayer1Protocol" type="FiveBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="BCOctet5aType">
<xs:sequence>
<xs:element name="SynchronousAsynchronous" type="OneBitType"/>
<xs:element name="Negotiation" type="OneBitType"/>
<xs:element name="UserRate" type="FiveBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="BCOctet5bV110Type">
<xs:sequence>
<xs:element name="IntermediateRate" type="TwoBitType"/>
<xs:element name="NIConTX" type="OneBitType"/>
<xs:element name="NIConRX" type="OneBitType"/>
<xs:element name="FlowControlOnTX" type="OneBitType"/>
<xs:element name="FlowControlOnRX" type="OneBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="BCOctet5bV120Type">
<xs:sequence>
<xs:element name="RateAdaptionHeader" type="OneBitType"/>
<xs:element name="MultipleFrameEstablishmentSupport" type="OneBitType"/>
<xs:element name="ModeOfOperation" type="OneBitType"/>
<xs:element name="LogicalLinkIdentifier" type="OneBitType"/>
<xs:element name="Assignor" type="OneBitType"/>
<xs:element name="InbandOutbandNegotiation" type="OneBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="BCOctet5cType">
<xs:sequence>
<xs:element name="NumberOfStopBits" type="TwoBitType"/>
<xs:element name="NumberOfDataBits" type="TwoBitType"/>
<xs:element name="Parity" type="ThreeBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="BCOctet5dType">
<xs:sequence>
<xs:element name="DuplexMode" type="OneBitType"/>
<xs:element name="ModemType" type="SixBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="BCOctet6Type">
<xs:sequence>
<xs:element name="Layer2Identification" type="TwoBitType"/>
<xs:element name="UserInfoLayer2Protocol" type="FiveBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="BCOctet7Type">
<xs:sequence>
<xs:element name="Layer3Identification" type="TwoBitType"/>
<xs:element name="UserInfoLayer3Protocol" type="FiveBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="BCOctet7aType">
<xs:sequence>
<xs:element name="AdditionalLayer3Info" type="FourBitType"/>
</xs:sequence>
</xs:complexType>
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<xs:complexType name="BCOctet7bType">
<xs:sequence>
<xs:element name="AdditionalLayer3Info" type="FourBitType"/>
</xs:sequence>
</xs:complexType>
<!--Definition of High Layer Compatibility Octets-->
<xs:complexType name="HLOctet3Type">
<xs:sequence>
<xs:element name="CodingStandard" type="TwoBitType"/>
<xs:element name="Interpretation" type="ThreeBitType"/>
<xs:element name="PresentationMethod" type="TwoBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="HLOctet4Type">
<xs:sequence>
<xs:element name="HighLayerCharacteristics" type="SevenBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="HLOctet4aMaintenanceType">
<xs:sequence>
<xs:element name="HighLayerCharacteristics" type="SevenBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="HLOctet4aAudioType">
<xs:sequence>
<xs:element name="VideoTelephonyCharacteristics" type="SevenBitType"/>
</xs:sequence>
</xs:complexType>
<!--Definition of Low Layer Compatibility Octets-->
<xs:complexType name="LLOctet3Type">
<xs:sequence>
<xs:element name="CodingStandard" type="TwoBitType"/>
<xs:element name="InformationTransferCapability" type="FiveBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="LLOctet3aType">
<xs:sequence>
<xs:element name="NegotiationIndicator" type="OneBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="LLOctet4Type">
<xs:sequence>
<xs:element name="TransferMode" type="TwoBitType"/>
<xs:element name="InformationTransferRate" type="FiveBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="LLOctet4-1Type">
<xs:sequence>
<xs:element name="RateMultiplier" type="SevenBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="LLOctet5Type">
<xs:sequence>
<xs:element name="Layer1Identification" type="TwoBitType"/>
<xs:element name="UserInfoLayer1Protocol" type="FiveBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="LLOctet5aType">
<xs:sequence>
<xs:element name="SynchronousAsynchronous" type="OneBitType"/>
<xs:element name="Negotiation" type="OneBitType"/>
<xs:element name="UserRate" type="FiveBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="LLOctet5bV110Type">
<xs:sequence>
<xs:element name="IntermediateRate" type="TwoBitType"/>
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<xs:sequence>
<xs:element name="DefaultPacketSize" type="FourBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="LLOctet7cType">
<xs:sequence>
<xs:element name="PacketWindowSize" type="SevenBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="LLOctet7aTR9577Type">
<xs:sequence>
<xs:element name="AdditionalLayer3Info" type="FourBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="LLOctet7bTR9577Type">
<xs:sequence>
<xs:element name="AdditionalLayer3Info" type="FourBitType"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="DispOctet3Type">
<xs:sequence>
<xs:element name="DisplayInformation" type="SevenBitType"/>
</xs:sequence>
</xs:complexType>
<!--Definition of the information elements-->
<xs:complexType name="BearerCapabilityType">
<xs:sequence>
<xs:element name="BCoctet3" type="BCOctet3Type"/>
<xs:element name="BCoctet4" type="BCOctet4Type"/>
<xs:element name="BCoctet4-1" type="BCOctet4-1Type" minOccurs="0"/>
<xs:element name="BCoctet5" type="BCOctet5Type" minOccurs="0"/>
<xs:element name="BCoctet5a" type="BCOctet5aType" minOccurs="0"/>
<xs:element name="BCoctet5bV110" type="BCOctet5bV110Type" minOccurs="0"/>
<xs:element name="BCoctet5bV120" type="BCOctet5bV120Type" minOccurs="0"/>
<xs:element name="BCoctet5c" type="BCOctet5cType" minOccurs="0"/>
<xs:element name="BCoctet5d" type="BCOctet5dType" minOccurs="0"/>
<xs:element name="BCoctet6" type="BCOctet6Type" minOccurs="0"/>
<xs:element name="BCoctet7" type="BCOctet7Type" minOccurs="0"/>
<xs:element name="BCoctet7a" type="BCOctet7aType" minOccurs="0"/>
<xs:element name="BCoctet7b" type="BCOctet7bType" minOccurs="0"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="HighLayerCompatibilityType">
<xs:sequence>
<xs:element name="HLOctet3" type="HLOctet3Type"/>
<xs:element name="HLOctet4" type="HLOctet4Type"/>
<xs:element name="HLOctet4aMaintenance" type="HLOctet4aMaintenanceType" minOccurs="0"/>
<xs:element name="HLOctet4Audio" type="HLOctet4aAudioType" minOccurs="0"/>
</xs:sequence>
</xs:complexType>
<xs:complexType name="LowLayerCompatibilityType">
<xs:sequence>
<xs:element name="LLOctet3" type="LLOctet3Type"/>
<xs:element name="LLOctet3a" type="LLOctet3aType" minOccurs="0"/>
<xs:element name="LLOctet4" type="LLOctet4Type"/>
<xs:element name="LLOctet4-1" type="LLOctet4-1Type" minOccurs="0"/>
<xs:element name="LLOctet5" type="LLOctet5Type" minOccurs="0"/>
<xs:element name="LLOctet5a" type="LLOctet5aType" minOccurs="0"/>
<xs:element name="LLOctet5bV110" type="LLOctet5bV110Type" minOccurs="0"/>
<xs:element name="LLOctet5bV120" type="LLOctet5bV120Type" minOccurs="0"/>
<xs:element name="LLOctet5c" type="LLOctet5cType" minOccurs="0"/>
<xs:element name="LLOctet5d" type="LLOctet5dType" minOccurs="0"/>
<xs:element name="LLOctet6" type="LLOctet6Type" minOccurs="0"/>
<xs:element name="LLOctet6aHDLC" type="LLOctet6aHDLCType" minOccurs="0"/>
<xs:element name="LLOctet6aUserSpecific" type="LLOctet6aUserSpecificType" minOccurs="0"/>
<xs:element name="LLOctet6b" type="LLOctet6bType" minOccurs="0"/>
<xs:element name="LLOctet7" type="LLOctet7Type" minOccurs="0"/>
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Annex G (normative):
Overlap digit message body
G.1 Scope
This section defines a message body that shall be used for sending additional digits, which have not previously been
sent, in SIP INFO messages when the in-dialog method is used for overlap dialling.
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If the message body is embedded in SIP INFO messages, the Content-Type header shall be set to "application/x-
session-info" and the Content-Disposition header shall be set to "signal" with the handling parameter set to "optional".
G.3 ABNF
x-session-info = SubsequentDigit
Annex H (normative):
Interworking of Originating Line Information (OLI) parameter
(network option)
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Annex I (informative):
Change history
Change history
Date TSG # TSG Doc. CR Rev Subject/Comment Old New
2003-09 NP#21 NP-030326 Approved at NP#21 and placed under change control 2.0.0 6.0.0
2003-12 NP#22 NP-030569 001 1 Use of response code 500 instead of 503 6.0.0 6.1.0
2003-12 NP#22 NP-030569 002 1 Autonomous Release at I MGCF on T7 expiry 6.0.0 6.1.0
2003-12 NP#22 NP-030569 003 1 Clarification of 487 mapping to 127 6.0.0 6.1.0
2003-12 NP#22 NP-030569 004 2 Table 12 modifications 6.0.0 6.1.0
2003-12 NP#22 NP-030569 008 Correction of clause titles 6.0.0 6.1.0
2003-12 NP#22 NP-030570 009 1 Failure handling in MGCF 6.0.0 6.1.0
2003-12 NP#22 NP-030569 010 1 Interworking of user plane 6.0.0 6.1.0
2003-12 NP#22 NP-030569 011 2 Alignment between subclause 7.2.3 and 7.3.3 in TS 29.163 6.0.0 6.1.0
2003-12 NP#22 NP-030570 012 5 Corrections to clause 9 of TS 29.163 6.0.0 6.1.0
2003-12 NP#22 NP-030569 013 1 Interworking (overlap to en-bloc conversion) timer corrections 6.0.0 6.1.0
2003-12 NP#22 NP-030570 014 2 IM-MGW initiated release 6.0.0 6.1.0
2003-12 NP#22 NP-030569 015 1 Alignment of TS 29.163 with the ITU-T Q.1912.5 recommendation 6.0.0 6.1.0
2003-12 NP#22 NP-030570 016 1 Corrections to table 29 and 30 of TS 29.163 6.0.0 6.1.0
2003-12 NP#22 NP-030569 018 1 Mapping of unknown cause code values 6.0.0 6.1.0
2003-12 NP#22 NP-030569 021 2 Addition of References 6.0.0 6.1.0
2003-12 NP#22 NP-030569 022 3 Handling of closed used group supplementary service 6.0.0 6.1.0
2003-12 NP#22 NP-030570 023 2 Corrections on Clause 9.2.8 Handling of RTP telephony events 6.0.0 6.1.0
2003-12 NP#22 NP-030570 024 Wrong Mn Procedure in Figure 36 6.0.0 6.1.0
2003-12 NP#22 NP-030569 025 1 Interworking of Hold/Resume from the CS Network 6.0.0 6.1.0
2004-03 NP#23 NP-040083 030 2 Reason Headers 6.1.0 6.2.0
2004-03 NP#23 NP-040083 031 2 Informative annex for missalignments with Q.1912.5 6.1.0 6.2.0
2004-03 NP#23 NP-040083 032 2 Criteria for sending UPDATE in BICC 6.1.0 6.2.0
2004-03 NP#23 NP-040084 033 2 Impact of Forking on Mn procedures 6.1.0 6.2.0
2004-03 NP#23 NP-040083 034 1 Impact of Forking on Incoming call interworking 6.1.0 6.2.0
2004-03 NP#23 NP-040083 035 2 Impact of Forking on Outgoing call interworking 6.1.0 6.2.0
2004-03 NP#23 NP-040083 036 1 Impact of Forking on COLP supplementary service 6.1.0 6.2.0
2004-06 NP#24 NP-040241 037 1 Message sequence implies that CS side „ACM‟ message is sent only 6.2.0 6.3.0
after 200 OK to PRACK is received
2004-06 NP#24 NP-040241 038 1 Originated/terminated correction 6.2.0 6.3.0
2004-06 NP#24 NP-040242 039 1 Interworking with Nb user plane procedures 6.2.0 6.3.0
2004-06 NP#24 NP-040242 040 1 Codec Negotiation between BICC CS networks and the IM CN 6.2.0 6.3.0
subsystem
2004-06 NP#24 NP-040242 041 1 Codec negotiation incoming call interworking 6.2.0 6.3.0
2004-06 NP#24 NP-040242 042 2 Codec negotiation – Mid call interworking 6.2.0 6.3.0
2004-06 NP#24 NP-040242 043 1 Codec parameter translation – IM CN subsystem to BICN 6.2.0 6.3.0
2004-06 NP#24 NP-040242 044 2 MGCF IM-MGW interactions 6.2.0 6.3.0
2004-06 NP#24 NP-040241 045 Notify IMS RTP Tel Event (same as „Report DTMF‟) message 6.2.0 6.3.0
sequence shows IEs that are not used with this procedure
2004-06 NP#24 NP-040241 046 Correction of sub-clause 7.2.3.2.5.1 Backward call indicators 6.2.0 6.3.0
2004-09 NP#25 NP-040334 050 3 Corrections to AMR codec parameter translations 6.3.0 6.4.0
2004-09 NP#25 NP-040346 048 2 Non call-related Mc procedures 6.3.0 6.4.0
2004-12 NP#26 NP-040582 059 Editorial mistake in Table 12 6.4.0 6.5.0
2004-12 NP#26 NP-040582 056 1 Corrections to EFR codec parameters 6.4.0 6.5.0
2004-12 NP#26 NP-040582 057 2 DTMF towards IM CN subsystem 6.4.0 6.5.0
2004-12 NP#26 NP-040582 054 3 Mapping of continuity signal 6.4.0 6.5.0
2004-12 NP#26 NP-040583 058 2 Clarifications for Mn procedures for call hold 6.4.0 6.5.0
2005-03 NP#27 NP-050105 060 1 Corrections to AMR codec parameters 6.5.0 6.6.0
2005-06 CP#28 CP-050038 064 1 Call Hold corrections 6.6.0 6.7.0
2005-09 CP#29 CP-050451 073 4 Coding of Called Party Number 6.7.0 7.0.0
2005-09 CP#29 CP-050451 074 1 Mapping of Hop Counter 6.7.0 7.0.0
2005-09 CP#29 CP-050451 077 3 mapping of Called Party Number 6.7.0 7.0.0
2005-12 CP#30 CP-050515 070 2 Mapping of codecs 7.0.0 7.1.0
2005-12 CP#30 CP-050521 080 2 Clean-up of hanging contexts and terminations 7.0.0 7.1.0
2005-12 CP#30 CP-050515 081 3 Interworking of 3PTY and CONF 7.0.0 7.1.0
2005-12 CP#30 CP-050515 082 2 Interworking of ACR 7.0.0 7.1.0
2005-12 CP#30 CP-050513 086 2 Support of Tel and SIP URI 7.0.0 7.1.0
2005-12 CP#30 CP-050515 087 1 Support of Tel and SIP URImapping of "restriction by the network" 7.0.0 7.1.0
2005-12 CP#30 CP-050514 088 2 IMS Terminating Callflows without preconditions 7.0.0 7.1.0
2005-12 CP#30 CP-050514 089 2 IMS Originating Callflows without preconditions 7.0.0 7.1.0
3GPP
Release 8 283 3GPP TS 29.163 V8.16.0 (2011-09)
2005-12 CP#30 CP-050514 090 2 IMS Terminating Procedures without preconditions 7.0.0 7.1.0
2005-12 CP#30 CP-050514 091 3 IMS Originating Procedures without preconditions 7.0.0 7.1.0
2005-12 CP#30 CP-050512 093 3 Handling of Overlap signalling 7.0.0 7.1.0
2005-12 CP#30 CP-050516 094 1 Incorporating of TR 24.819 fixed broadband access impacts into TS 7.0.0 7.1.0
29.163
2005-12 CP#30 CP-050659 095 1 Interworking of FCI and BCI 7.0.0 7.1.0
2006-03 CP#31 CP-060056 096 Clarfication of IAM to SIP Invite message mapping 7.1.0 7.2.0
2006-03 CP#31 CP-060056 098 1 SCTP changes 7.1.0 7.2.0
2006-03 CP#31 CP-060046 100 Bearer Released use with TDM Circuit 7.1.0 7.2.0
2006-03 CP#31 CP-060046 105 488 status code 7.1.0 7.2.0
2006-03 CP#31 CP-060047 109 4 Interworking RTP timestamps and IuFP frame numbers 7.1.0 7.2.0
2006-03 CP#31 CP-060129 110 Status Code 433 for ACR 7.1.0 7.2.0
2006-06 CP#32 CP-060223 111 3 Removal of editor´s notes on open points for Mn Procedures for non- 7.2.0 7.3.0
preconditions Callflows
2006-06 CP#32 CP-060223 112 4 Add related references to T.38 7.2.0 7.3.0
2006-06 CP#32 CP-060220 116 1 Bearer Released use with IMS terminations 7.2.0 7.3.0
2006-06 CP#32 CP-060223 117 1 Reference to the correct value of Anonymous URI 7.2.0 7.3.0
2006-09 CP#33 CP-060429 119 3 Interworking of REFER 7.3.0 7.4.0
2006-09 CP#33 CP-060429 120 2 Interworking of Nature of connection indicators 7.3.0 7.4.0
2006-09 CP#33 CP-060429 121 3 Interworking of CPC 7.3.0 7.4.0
2006-09 CP#33 CP-060429 122 2 MGCF Procedures for non-preconditions Callflows 7.3.0 7.4.0
2006-09 CP#33 CP-060429 123 1 Suitable references for Status Code 433 for ACR 7.3.0 7.4.0
2006-09 CP#33 CP-060424 125 2 Echo Control Device indication in ACM/CPG 7.3.0 7.4.0
2006-09 CP#33 CP-060437 126 Missing description of CUG service 7.3.0 7.4.0
2006-09 CP#33 CP-060425 128 2 Missing procedures toward IMS Terminations 7.3.0 7.4.0
2006-09 CP#33 CP-060471 129 1 Changes due to non-precondition setup 7.3.0 7.4.0
2006-12 CP#34 CP-060626 130 4 Interworking of USI 7.4.0 7.5.0
2006-12 CP#34 CP-060734 131 1 Handling of emergency call in MGCF 7.4.0 7.5.0
2006-12 CP#34 CP-060632 132 Clarifications on Supplementary service handling 7.4.0 7.5.0
2006-12 CP#34 CP-060633 133 1 Unknown User Identity 7.4.0 7.5.0
2007-03 CP#35 CP-070095 136 1 Scope update for Multimedia interworking 7.5.0 7.6.0
2007-03 CP#35 CP-070095 137 4 Multimedia interworking 7.5.0 7.6.0
2007-03 CP#35 CP-070103 138 1 Adding CS data call interworking to interworking capabilities 7.5.0 7.6.0
overview table 1
2007-06 CP#36 CP-070412 140 3 Media oriented negotiation acceleration 7.6.0 7.7.0
2007-06 CP#36 CP-070412 141 6 Mn Procedures of Multimedia Interworking 7.6.0 7.7.0
2007-06 CP#36 CP-070413 142 2 Change Table 11 7.6.0 7.7.0
2007-06 CP#36 CP-070413 143 1 correction of cpc interworking 7.6.0 7.7.0
2007-06 CP#36 CP-070413 144 1 The interworking of the PSTN ECT service 7.6.0 7.7.0
2007-06 CP#36 CP-070413 145 2 Mapping of HLC 7.6.0 7.7.0
2007-06 CP#36 CP-070413 146 1 Mistake in the handling of sending ringing tone 7.6.0 7.7.0
2007-06 CP#36 CP-070413 147 1 Support of all types of TMR 7.6.0 7.7.0
2007-06 CP#36 CP-070483 148 5 IMS communication service identifier 7.6.0 7.7.0
2007-06 CP#36 CP-070412 149 1 Editorial Corrections 7.6.0 7.7.0
2007-06 CP#36 CP-070415 151 2 Taking P-Early-Media header into account in 29.163 7.6.0 7.7.0
2007-06 CP#36 CP-070416 153 2 IP realm connection indication 7.6.0 7.7.0
2007-09 CP#37 CP-070562 155 2 Essential corrections to P-Early-Media header procedures 7.7.0 7.8.0
2007-09 CP#37 CP-070562 157 2 Action of requesting the absent CLI 7.7.0 7.8.0
2007-09 CP#37 CP-070561 160 2 7 Khz Mapping 7.7.0 7.8.0
2007-09 CP#37 CP-070551 162 2 Correction to Mn procedures 7.7.0 7.8.0
2007-09 CP#37 CP-070553 163 2 Maximum Multiplex Level for H.223 negotiation 7.7.0 7.8.0
2007-09 CP#37 CP-070553 164 Multiplex tables 7.7.0 7.8.0
2007-09 CP#37 CP-070553 165 Flow correction: removal of demux- and connection properties 7.7.0 7.8.0
2007-09 CP#37 CP-070562 166 2 Interworking of SIP History-Info header 7.7.0 7.8.0
2007-09 CP#37 CP-070563 173 2 Mn Procedures to support P-early-media SIP header 7.7.0 7.8.0
2007-09 CP#37 CP-070553 174 3 Corrections to Multimedia Mn Procedures 7.7.0 7.8.0
2007-09 CP#37 CP-070686 168 3 Corrections to Multimedia Mn Procedures 7.8.0 8.0.0
2007-09 CP#37 CP-070686 169 3 Add support for equal Carrier Access procedures 7.8.0 8.0.0
2007-12 CP#38 CP-070721 178 1 Inactivity timout procedures – Alignment to Mc profile 8.0.0
CP-070723 179 1 Termination heartbeat – Alignment to Mc profile
CP-070722 181 1 Update P-Early-Media Reference
CP-070724 182 1 Addition of interworking for Sub-adressing
CP-070725 185 1 Add support for ISUP to SIP interworking for carrier-based routeing 8.1.0
2008-03 CP#39 CP-080047 196 3 SIP XML transit specific element interworking 8.1.0 8.2.0
2008-03 CP#39 CP-080047 197 3 Progress Indicator mapping 8.1.0 8.2.0
2008-03 CP#39 CP-080047 198 4 Procedure for Fall back interworking 8.1.0 8.2.0
2008-03 CP#39 CP-080045 199 3 Addition of UUS Interworking description 8.1.0 8.2.0
2008-03 CP#39 CP-080041 201 Reason Header in Responses 8.1.0 8.2.0
2008-03 CP#39 CP-080042 203 1 Corrections for facsimile interworking 8.1.0 8.2.0
2008-03 CP#39 CP-080039 205 2 Correction to Call setup if multimedia call can not be recognized in 8.1.0 8.2.0
3GPP
Release 8 284 3GPP TS 29.163 V8.16.0 (2011-09)
an unambiguous manner
2008-05 CP#40 CP-080297 207 1 Additions of subclause for TISPAN CDIV supplemetary service 8.2.0 8.3.0
interworking
2008-05 CP#40 CP-080297 208 1 Additions of subclause for TISPAN CONFsupplemetary service 8.2.0 8.3.0
interworking
2008-05 CP#40 CP-080297 209 2 Additions of subclause for TISPAN TIP/TIR supplemetary service 8.2.0 8.3.0
interworking
2008-05 CP#40 CP-080297 210 1 Additions of subclause for CUG simulation service for TISPAN 8.2.0 8.3.0
supplemetary
2008-05 CP#40 CP-080297 211 2 Additions of subclausefor MCID simulation service for TISPAN 8.2.0 8.3.0
supplemetary
2008-05 CP#40 CP-080297 212 4 Inclusion of common procedure in TS 29.163 8.2.0 8.3.0
2008-05 CP#40 CP-080297 215 4 TMR and Fallback mapping 8.2.0 8.3.0
2008-05 CP#40 CP-080294 217 3 Interworking of Terminal Portability 8.2.0 8.3.0
2008-05 CP#40 CP-080294 218 2 Satellite indicator mapping 8.2.0 8.3.0
2008-05 CP#40 CP-080291 219 1 MONA Mn Procedures 8.2.0 8.3.0
2008-05 CP#40 CP-080290 221 2 DTMF Encoding 8.2.0 8.3.0
2008-05 CP#40 CP-080290 223 1 DTMF Mn Procedures 8.2.0 8.3.0
2008-09 CP#41 CP-080556 226 Correction to the communication diversion service 8.3.0 8.4.0
2008-09 CP#41 CP-080552 228 3 Correction to supplementary service sections in TS 29.163 8.3.0 8.4.0
2008-09 CP#41 CP-080563 231 1 Mapping of TMU 8.3.0 8.4.0
2008-09 CP#41 CP-080563 233 1 Progress Indicator mapping 8.3.0 8.4.0
2008-09 CP#41 CP-080563 235 1 Editorial changes 8.3.0 8.4.0
2008-09 CP#41 CP-080554 241 2 CCBS interworking 8.3.0 8.4.0
2008-09 CP#41 CP-080559 242 1 Improvements to MTSI and 3G324M interworking 8.3.0 8.4.0
2008-09 CP#41 CP-080552 244 2 Coding of the b=line in SDP information 8.3.0 8.4.0
2008-09 CP#41 CP-080555 245 2 CW interworking 8.3.0 8.4.0
2008-09 CP#41 CP-080556 246 1 Mapping of Call Rejected 8.3.0 8.4.0
2008-12 CP#42 CP-080761 248 6 Message body to transfer digits 8.4.0 8.5.0
2008-12 CP#42 CP-080761 249 2 I-MGCF: Overlap signalling procedures 8.4.0 8.5.0
2008-12 CP#42 CP-080761 250 4 O-MGCF: Overlap signalling additions 8.4.0 8.5.0
2008-12 CP#42 CP-080753 251 1 Satellite indicator 8.4.0 8.5.0
2008-12 CP#42 CP-080769 252 1 Revision Annex F3 8.4.0 8.5.0
2008-12 CP#42 CP-080771 255 1 Update reference for DAI Parameter for the "tel" URI 8.4.0 8.5.0
2008-12 CP#42 CP-080771 256 2 Editorial Corrections 8.4.0 8.5.0
2008-12 CP#42 CP-080771 262 3 Correction of the mapping tables for interwoking call forwarding 8.4.0 8.5.0
CDIV
2008-12 CP#42 CP-080771 263 1 Addition of interworking for Sub-addressing 8.4.0 8.5.0
2008-12 CP#42 CP-080753 265 1 Update to reference for ACR 8.4.0 8.5.0
2008-12 CP#42 CP-080771 266 1 Corrections to References 8.4.0 8.5.0
2008-12 CP#42 CP-080771 267 2 Miscellaneous corrections 8.4.0 8.5.0
2008-12 CP#42 CP-080766 268 3 Clarification of RTCP messages usage in the inter-working gateways 8.4.0 8.5.0
2008-12 CP#42 CP-080762 272 1 CCBS Interworking 8.4.0 8.5.0
2008-12 CP#42 CP-080763 273 4 CDIV alignment with 3GPPP2 8.4.0 8.5.0
2008-12 CP#42 CP-080763 274 1 Modification of Ti/w2 timer values 8.4.0 8.5.0
2008-12 CP#42 CP-080763 275 2 Receipt of INVITE with no SDP (offer) 8.4.0 8.5.0
2008-12 CP#42 CP-080770 276 MONA corrections 8.4.0 8.5.0
2008-12 CP#42 CP-080771 278 2 Addition of IP interface type 8.4.0 8.5.0
2009-03 CP#43 CP-090095 281 1 Handle the SIP URI in History-Info 8.5.0 8.6.0
2009-03 CP#43 CP-090228 282 1 Miscellaneous corrections in History-Info mapping tables 8.5.0 8.6.0
2009-03 CP#43 CP-090095 283 1 Map priv-value of session and header in History-Info 8.5.0 8.6.0
2009-03 CP#43 CP-090094 285 1 Progress Indicator mapping 8.5.0 8.6.0
2009-03 CP#43 CP-090094 286 2 Supplementary service reference and naming correction 8.5.0 8.6.0
2009-03 CP#43 CP-090078 290 2 Corrections to Tables 12 and 16 8.5.0 8.6.0
2009-03 CP#43 CP-090217 291 4 Clarification of CDIV mapping 8.5.0 8.6.0
2009-05 CP#44 CP-090349 292 2 Missing MONA procedures in stage 2 8.6.0 8.7.0
2009-05 CP#44 CP-090348 293 Correction on CDIV mapping 8.6.0 8.7.0
2009-05 CP#44 CP-090348 295 2 Correction of ACM and CPG sending procedures 8.6.0 8.7.0
2009-05 CP#44 CP-090348 296 1 Renumbering of duplicated table 17c 8.6.0 8.7.0
2009-05 CP#44 CP-090487 297 2 I-MGCF procedures for MCID mapping 8.6.0 8.7.0
2009-05 CP#44 CP-090332 300 2 Correction of the procedure for setting of Continuity Indicator in 8.6.0 8.7.0
subclause 7.3.3.1.2.2
2009-05 CP#44 CP-090344 302 1 Mapping between GSM HR codec and SDP parameters 8.6.0 8.7.0
2009-09 CP#45 CP-090568 306 1 Correcting references to H.324 regarding MONA 8.7.0 8.8.0
2009-09 CP#45 CP-090579 307 Correction of the mapping of PSTN XML body with ISUP parameters 8.7.0 8.8.0
to ACM, REL and CON
2009-12 CP#46 CP-090835 313 5 Correction of CPC parameter mapping 8.8.0 8.9.0
2009-12 CP#46 CP-090848 314 CDIV redirection parameters mapping 8.8.0 8.9.0
2009-12 CP#46 CP-090848 319 5 Support interworking of Call Forwarding information 8.8.0 8.9.0
2009-12 CP#46 CP-090834 322 3 Reference to Reason Header in Responses 8.8.0 8.9.0
2009-12 CP#46 CP-090835 324 2 Interworking ISUP OLI parameter 8.8.0 8.9.0
3GPP
Release 8 285 3GPP TS 29.163 V8.16.0 (2011-09)
2009-12 CP#46 CP-090835 326 1 Mapping for Communication Barring Service 8.8.0 8.9.0
2009-12 CP#46 CP-090834 330 2 Mapping of From header at O-MGCF 8.8.0 8.9.0
2010-03 CP#47 CP-100071 334 1 Handling of SDP bandwidth attribute 8.9.0 8.10.0
2010-03 CP#47 CP-100081 336 1 MCID Interworking 8.9.0 8.10.0
2010-03 CP#47 CP-100079 338 2 Interworking of RTCP and H.245 messages 8.9.0 8.10.0
2010-03 CP#47 CP-100081 341 2 Correction of mapping between ISUP CSI and dai parameter 8.9.0 8.10.0
2010-03 CP#47 CP-100081 343 Duplicate table number 8.9.0 8.10.0
2010-03 CP#47 CP-100071 346 Corrections to Release Procedures 8.9.0 8.10.0
2010-03 CP#47 CP-100081 348 Corrections to Release Procedures when MGCF supports PSTN 8.9.0 8.10.0
XML body
2010-03 CP#47 CP-100071 351 Corrections to Table 6 8.9.0 8.10.0
2010-03 CP#47 CP-100071 354 1 Corrections to through-connection procedures 8.9.0 8.10.0
2010-03 CP#47 CP-100082 356 MONA alignments to H.324 8.9.0 8.10.0
2010-03 CP#47 CP-100081 358 2 Clarification of CCBS/CCNR service interwork 8.9.0 8.10.0
2010-03 CP#47 CP-100081 360 2 Correction of Alert-URN for call-waiting service 8.9.0 8.10.0
2010-03 CP#47 CP-100071 364 Correction for Cause Mapping 8.9.0 8.10.0
2010-06 CP#48 CP-100307 374 1 Correction of Cause Code mapping 8.10.0 8.11.0
2010-06 CP#48 CP-100307 377 1 Addition of ISUP Cause mapping 8.10.0 8.11.0
2010-06 CP#48 CP-100307 380 1 Addition of Response Code mapping (422, 430, 439, 440) 8.10.0 8.11.0
2010-06 CP#48 CP-100414 382 1 Addition of Response Code mapping (417) 8.10.0 8.11.0
2010-06 CP#48 CP-100414 385 MCID interworking corrections 8.10.0 8.11.0
2010-06 CP#48 CP-100414 387 2 Omissions from ISDN Fallback Procedures 8.10.0 8.11.0
2010-06 CP#48 CP-100414 389 1 Use of PSTN XML SendingCompleteIndicator 8.10.0 8.11.0
2010-09 CP#49 CP-100541 397 1 Addition of Cause Value mapping 8.11.0 8.12.0
2010-09 CP#49 CP-100546 399 Alignment to changes in 24.642 8.11.0 8.12.0
2010-09 CP#49 CP-100547 402 3 Corrections to CDIV interworking 8.11.0 8.12.0
2010-09 CP#49 CP-100547 404 3 Call Forwarding History-Info interworking corrections 8.11.0 8.12.0
2010-09 CP#49 CP-100541 409 Correcting unspecific external reference 8.11.0 8.12.0
2010-12 CP#50 CP-100772 414 2 Mapping of ISUP Cause Value 34 8.12.0 8.13.0
2010-12 CP#50 CP-100772 420 2 Mapping of ISUP Cause Value 102 to Response 504 and vice versa 8.12.0 8.13.0
2010-12 CP#50 CP-100780 422 2 XML Octet 7 8.12.0 8.13.0
2010-12 CP#50 CP-100772 425 Changes to ECT 8.12.0 8.13.0
2010-12 CP#50 CP-100776 427 1 Changes to CCBS/CCNR 8.12.0 8.13.0
2010-12 CP#50 CP-100772 430 Reception of 580 final response to UPDATE request 8.12.0 8.13.0
2010-12 CP#50 CP-100777 432 Reference update: IETF draft-liess-dispatch-alert-info-urns 8.12.0 8.13.0
2010-12 CP#50 CP-100780 438 3 Support of Emergency Call CPC interworking 8.12.0 8.13.0
2010-12 CP#50 CP-100780 441 1 CDIV interworking corrections 8.12.0 8.13.0
2011-03 CP#51 CP-110105 462 1 Change of Cause code mapping 8.13.0 8.14.0
2011-03 CP#51 CP-110105 444 2 Clarification of I-MGCF behaviour when SIP preconditions are used 8.13.0 8.14.0
2011-03 CP#51 CP-110105 448 1 Clarification of O-MGCF behaviour when SIP preconditions are used 8.13.0 8.14.0
2011-03 CP#51 CP-110105 484 1 correction BCI mapping 8.13.0 8.14.0
2011-03 CP#51 CP-110105 469 1 Mapping of ISUP Cause Value 88 to Response 606 and vice versa 8.13.0 8.14.0
2011-03 CP#51 CP-110105 474 1 Mapping of SIP Response 604 to ISUP Cause Value 2 8.13.0 8.14.0
2011-03 CP#51 CP-110105 458 1 New Cause Code mappings of CV 23,38- 47and 65-79 8.13.0 8.14.0
2011-03 CP#51 CP-110108 480 2 CC timer expiry correction 8.13.0 8.14.0
2011-03 CP#51 CP-110108 454 2 CCBS/CCNR correction of SUBSCRIBE request 8.13.0 8.14.0
2011-03 CP#51 CP-110109 451 IETF Reference Update: RFC 5993 8.13.0 8.14.0
2011-03 CP#51 CP-110110 369 7 Handling of non-E.164 format numbers on NNI 8.13.0 8.14.0
2011-03 CP#51 CP-110110 477 1 IETF Reference Update 8.13.0 8.14.0
2011-03 CP#51 CP-110110 465 2 Missing Timer definition for TIR 8.13.0 8.14.0
2011-03 CP#51 CP-110110 487 1 rn parameter mapping 8.13.0 8.14.0
2011-03 CP#51 Editorial changes made by MCC 8.14.0 8.14.1
2011-06 CP#52 CP-110395 507 Removal of reference IETF draft-patel-dispatch-cpc-oli-parameter 8.14.1 8.15.0
2011-06 CP#52 CP-110399 520 1 Removal of dial around indicator 8.14.1 8.15.0
2011-06 CP#52 Editorial improvements made by MCC 8.15.0 8.15.1
2011-09 CP#53 CP-110604 525 2 INVITE with multiple m-lines 8.15.1 8.16.0
2011-09 CP#53 CP-110603 529 2 Interactions of " ISDN user part/BICC preference indicator" in the 8.15.1 8.16.0
Forward Call Indicator with Supplementary services.
2011-09 CP#53 CP-110720 532 2 PSTN XML handling 8.15.1 8.16.0
2011-09 CP#53 CP-110607 535 CCBS correction PSTN CLIR 8.15.1 8.16.0
3GPP