TTI Bundling in FDD and TD-LTE

Abstract

TTI Bundling has been introduced in FDD and TD-LTE to improve Uplink coverage. Here are the
salient features of TTI Bundling with respect to FDD and TD-LTE. We will first explore the motivation
for TTI Bundling and then compare it with other techniques. Finally, we shall address the
implementation of TTI bundling in FDD and TD-LTE .

Introduction

TTI bundling is a technique used to send a transport block multiple times in consecutive subframes
without waiting for HARQ ACK/NACK messages. Normally, a transport block is converted to multiple
redundancy versions after coding and the first redundancy version is sent in a subframe. Subsequent
transmissions of the transport block are dependent on the HARQ ACK/NACK which is sent 4
subframe durations later or more after the first transmission. In TTI bundling, the different
redundancy versions can all be sent in consecutive subframes without waiting for the HARQ
ACK/NACK feedback and a combined ACK/NACK can be sent after processing all the transmissions
of a transport block.

The motivation for TTI bundling which is illustrated in Figure 1 is the low transmission power of some
handsets, short TTI length, and the long RTT of the HARQ transmissions. TTI bundling is expected
to improve the UL coverage of applications like VOIP over LTE wherein low power handsets are likely
to be involved. This feature has more relevance for TDD over FDD as coverage issues are likely to
be more challenging in TD-LTE. Simulation results reported in publications indicate a 4 dB gain due
to TTI bundling on the UL.

LTE Coverage Improvement by TTI Bundling

ABSTRACT Compared to WCDMA, the LTE radio access has a significantly shorter Transmission
Time Interval (TTI) in order to reduce end-to-end delays. However, if a User Equipment (UE) at the
cell edge is limited by its available transmission power, it may not be able to transmit an entire VoIP
packet during one TTI, since the instantaneous source data rate is too high. Thus TTI bundling has
been recently introduced as a feature of LTE Rel. 8 to improve the uplink coverage. In TTI bundling, a
VoIP packet is transmitted as a single PDU during a bundle of subsequent TTIs without waiting for
the HARQ feedback. HARQ feedback is only expected for the last transmission of the bundle. This
paper studies TTI bundling and compares it to the conventional RLC segmentation. The si mulation
results indicate that TTI bundling provides a gain of more than 4 dB in terms of the sustainable path
loss.
Alternative Approach

The alternative to TTI bundling is RLC segmentation wherein a VOIP payload is split into smaller size
RLC PDUs as shown in Figure 2. The smaller RLC PDUs will result in smaller transport blocks which
can be decoded with better accuracy. One drawback of this method is the potential overhead
increase due to RLC segmentation due to multiple RLC headers needed. For a typical VOIP
payload, it has been shown that as we increase the segmentation factor from 1 to 8, the overhead
increases from 14% to 55%. Each RLC PDU which is mapped into a transport block will need a
separate PDCCH assignment message which will contribute to control signal overhead for such a
scheme. There might be retransmissions of each of those transport blocks which will also potentially
increase the control signaling overhead. In addition, since we are transmitting many small transport
blocks, the chances of interpreting a NACK as a ACK also increases proportionately with the increase
in the RLC segmentation size. Hence, RLC segmentation has many disadvantages when we
consider the transmission of a VOIP like payload from a power limited terminal.
Overview of TTI Bundling

TTI bundling is used to achieve successful transmissions from power limited terminals. The process
as shown in Figure 3 is typically triggered by UE informing the eNB about its power limitations at the
present state. This could for example happen at the edge of a cell when the terminal has to send
high power but is limited by the power capability of the terminal. This triggers the eNB to transmit the
various redundancy versions of the same transport block in consecutive subframes or TTIs giving rise
to the name TTI bundling. A single PDCCH allocation is sufficient for the multiple transmissions thus
saving control overhead as compared to the RLC segmentation approach. A single HARQ
ACK/NACK for the combined transmissions is generated after processing the TTI bundle which can
reduce the error rate of the transport block as compared with processing a single redundancy
version. This approach can also reduce the delay in the HARQ process compared to transmissions
of the redundancy versions separated in time using the normal approach.
TTI Bundling Operation

As shown in Figure 4, TTI bundling enables up to 4 redundancy versions of the same transport block
to be sent in 4 consecutive subframes. In TD-LTE systems, the TDD configurations standardized
allow only a maximum of 3 consecutive UL subframes. A single RLC PDU is transmitted as multiple
redundancy versions in consecutive subframes using a single common allocation. The channel
coding used in LTE enables easy generation of the multiple redundancy versions from which the
transmissions in the TTI bundle are generated. A common RLC header is shared across the TTI
bundle and the same HARQ process identity is used for multiple transmissions in the TTI
bundle. Combined processing of the redundant transmissions over multiple subframes leads to a
better probability of detection of the transport block. Thus, with limited power, the UE has a better
chance of a successful transmission with lesser latency using the TTI bundling method.

Summary: TTI bundling is a useful technique for improving coverage of VOIP handsets in LTE
systems. It is applicable to both FDD and TD-LTE deployments and can improve the link budget by
up to 4 dB. Differences in implementation exist between FDD and TD-LTE systems. TTI bundling
helps achieve good latency performance for VOIP even at the edges of cells.

Adaptive transmission of VoIP packets using TTI bundling in LTE uplink

ABSTRACT In long term evolution (LTE) uplink, transmission time interval (TTI) bundling technique is
employed for the voice over internet protocol (VoIP) service to improve the cell coverage
performance. In frequency-division duplex (FDD) mode, LTE uses 4 TTIs or subframes as a bundle
for transmission with a round trip time (RTT) 16ms. Thus the maximum number of transmission
bundles is only three if the delay budget is strictly limited to 50ms in VoIP service. Recently, a
coverage-enhanced scheme can transmit up to five bundles, but lacks time for feedback processing.
Once the unnecessary retransmissions occur, the user equipment (UE) transmit power can be
wasted for extra transmission. To overcome this problem, we propose an adaptive scheme using
block error rate (BLER) versus signal-to-noise ratio (SNR) mapping table, which enables UE
predetermine the maximum number of transmission bundles appropriately. Instead of using five
transmission bundles, our proposed scheme requires no more than five transmission bundles
according to SNR variation scale of the channel. The conducted simulation results show that our
proposed scheme can achieve higher throughput than coverage-enhanced scheme by reducing
unnecessary retransmissions, while it still satisfies the BLER requirements by employing a tolerance

Performance of TTI Bundling for VoIP In EUTRAN TDD Mode

ABSTRACT The long term evolution (LTE) of 3GPP radio-access technology aims to develop a
framework towards a high-data-rate, low-latency and packet-optimized radio access technology:
Evolved Universal Terrestrial Radio Access Networks (E-UTRAN). However, low terminal
transmission power, short TTI length and long HARQ RTT give a critical problem on LTE TDD UL
performance in a coverage-limited scenario. To solve coverage problem, this paper presents an
effective coverage enhancement mechanism called TTI bundling to enhance uplink VoIP
performance in LTE TDD mode. HARQ design for different TDD configurations is discussed with TTI
bundling. Performance evaluation for TTI bundling with VoIP traffic is carried out by system
simulation, and the impact from different bundling options with different bundle size and packet delay
budget is investigated. The simulation analysis proves that with proper bundle size and suitable
packet delay budget setting, TTI bundling can enhance the coverage performance for LTE TDD
effectively.

Main Feature
ActivationService::isTTIBundlingForVoIPEnable d
CellL1L2ControlChannelsConf::extraDCI0powerOffsetForTTIBundling
CellL1ULConf::betaOffsetACKIndexForTTIbundling
CellL2ULConf::maxHARQtxTTIbundling
CellL2ULConf::maxNbrOfTTIbundlingUsers
CellL2ULConf::maxULVoIPdataRateForTTIBundling
CellL2ULConf::tTIBundlingFixedMCSvalue
CellL2ULConf::tTIbundlingNotificationRepetitionTimer
CellL2ULConf::ttiBundlingPUSCHPRBZoneSize
CellL2ULConf::ttiBundlingPUSCHPRBzoneStart
CellL2ULConf::uplinkLinkBudgetAlarmClearanceThreshold
CellL2ULConf::uplinkLinkBudgetAlarmTimeToClear
CellL2ULConf::uplinkLinkBudgetAlarmTimeToTrigger
CellL2ULConf::uplinkLinkBudgetAlarmTriggerThreshold
CellRadioConf::minHARQtxWithoutMGcollisionForTTIbundling
RrcMeasurementConf::cpichEcn0OffsetForTtiBVoIPCalls
RrcMeasurementConf::cpichRscpOffsetForTtiBVoIPCalls
RrcMeasurementConf::rsrpOffsetOnNeighborCellForTtiBVoIPCalls
RrcMeasurementConf::rsrpOffsetOnServingCellForTtiBVoIPCalls
RrcMeasurementConf::rsrqOffsetOnNeighborCellForTtiBVoIPCalls
RrcMeasurementConf::rsrqOffsetOnServingCellForTtiBVoIPCalls
New Feature Status
& Definition for TDD & FDD sites.xlsx
TTI Bundling and VoIP Performance in LTE - Part I

One of the benefits of LTE is its superlative performance in the amount of payload that can be delivered in a
short period of time. This is a convoluted way to say: “fantastic throughput”. Yours truly has been testing this
fact, countless times by running a speed test app every night before I go to sleep! (59Mbps in DL is the
maximum I have seen. I would reveal my carrier’s name if it were not for fear of the repercussions …. ;-)

The minimum allocation in LTE, spans 180KHz and 1ms in the frequency and time domains respectively (this
is exactly one resource block in frequency and two resource blocks in time). This resource should be sufficient
to carry a VoIP packet using coded narrow band AMR ( @ 12.2 kbps = 244 bits in 20ms). Since a VoIP packet
carries 20ms worth of speech and our transmission time interval (TTI) is only 1ms, a UE that is engaged in
continuous voice is only active for 5% of the time!

The efficiency of LTE can be exploited to increase the voice capacity of the cell by time-multiplexing many
more users that could be scheduled in the intervening time before the first batch of users have to be scheduled
again. Disregarding many realistic and important constraints, and perfect RF conditions, the theoretical
capacity of voice calls per cell can be estimated as the number of users that can be scheduled in a TTI
multiplied by 20. In 20MHz band there are 100 Resource Blocks and presumably one user can be scheduled per
resource block in a TTI (this is far from the truth due to PDCCH limitations), then the theoretical peak capacity
of LTE is around 2000 calls per cell! Even with a 50% efficiency, 1000 calls per cell is impressive.

There is another way to use the 5% activity factor in LTE that is about improving the Uplink coverage at the
cost of reduced capacity. This technique is known in the 3GPP specs as TTI Bundling. In TTI bundling, the
20ms worth of speech packet is repeated in consecutive frames. Up to four TTIs can be used to send copies of
the same VoIP packet over the air. But why does the uplink coverage improve when copies of a voice packet
are bundled together?
To understand the impact of TTI bundling for uplink coverage, we need to remind ourselves of the uplink link
budget in LTE. In fact we only need to consider how much the eNodeB sensitivity is improved when TTI
bundling is used. TTI-Bundling allows for efficient decoding since it implies a four-fold redundancy in
transmission without any need for retransmissions! This should decrease the required signal to noise ratio at the
cell edge, without appreciable increase in latency. According to RAN1#54 report R1-081856, there is a 4dB
gain in uplink coverage in when 4 TTIs are bundled together (an extra twist to this result is that it is calculated
for 2 RBs which is what is needed for Wide-Band AMR transmissions).

TTI Bundling is activated in the network using Layer 3 signaling. One way to implement the activation or
deactivation of TTI bundling is to consider the UE power head room. This is a strong indicator of how much the
UE is struggling to close the uplink and be heard by the eNodeB at the appropriate Signal to Noise Ratio. So a
simple implementation for TTI bundling algorithm could depend on thresholds for the UEs available power at
any given moment in time.

In part two of this blog, I will look at the effects of delay and retransmission in TTI bundling.
Quality of Service (QoS) in LTE

Background: Why we need QoS ?

There are premium subscribers who always want to have better user experience on their 4G LTE device. These users are
willing to pay more for high bandwidth and better network access on their devices. Not only the subscribers but some
services itself need better priority handling in the network (e.g. VoIP call). To be able to full fill this, QOS plays the ke y
role. QOS defines priorities for certain customers / services during the time of high congestion in the network

3GPP definition for QoS

In LTE Network QoS is implemented between UE and PDN Gateway and is applied to a set of bearers. 'Bearer' is
basically a virtual concept and is a set of network configuration to provide special treatment to set of traffic e.g. VoIP
packets are prioritized by network compared to web browser traffic.
In LTE, QoS is applied on Radio bearer, S1 bearer and S5/S8 bearer, collectively called as EPS bearer as shown in figure
below.

In order to comprehend the concept of QoS , we must understand the bearer types and properties associated with each
bearer through hierarchical chart as shown below. First there are two types of Bearer, i.e. Dedicated bearer and Default
bearer. There is at-least one default bearer established when UE is attached to LTE network while dedicated bearer is
always established when there is need to provide QoS to specific service (like VoIP, video etc). Please go through the
article Default and Dedicated Bearer which hopefully will help to explain the concept in more detail.
Dedicated bearer can be subdivided into Non-GBR and GBR types.

GBR provides guaranteed bit rate and is associated with parameters like GBR and MBR

- GBR: The minimum guaranteed bit rate per EPS bearer. Specified independently for uplink and downlink

- MBR: The maximum guaranteed bit rate per EPS bearer. Specified independently for uplink and downlink

On the other hand, Non-GBR bearer does not provide guaranteed bit rate and has parameter like A- AMBR and UE-
AMBR

- A-AMBR: APN Aggregate maximum bit rate is the maximum allowed total non-GBR throughput to specific APN. It is
specified interdependently for uplink an downlink

- UE -AMBR: UE Aggregate maximum bit rate is the maximum allowed total non-GBR throughput among all APN to a
specific UE

As you can see, the default bearer can only be non-GBR type. Some other important terms associated with each bearer
type are discussed below:

- ARP: Allocation and retention priority is basically used for deciding whether new bearer modification or establishment
request should be accepted considering the current resource situation.
- TFT: Traffic flow template is always associated with dedicated bearer and while default bearer may or may not have
TFT. As mentioned earlier, dedicated bearer provides QoS to special service or application and TFT defines rules so that
UE and Network knows which IP packet should be sent on particular dedicated bearer. It usually has rules on the basis of
IP packet destination/source or protocol used.

L-EBI: It stands for Linked EPS bearer ID. As I discussed in previous article about dedicated and default bearer, we
know that each dedicated bearer is always linked to one of default bearers. L-EBI tells Dedicated bearer which default
bearer it is attached to.

IP Address/ PDN: Each default bearer is attached to some PDN network and has its own IP address while dedicated
bearer does not need this since it is linked to default bearer.

You can also see one other parameter associated with all bearers i.e. QoS class of identifier (QCI).This parameter
basically defines IP level packets characteristics as shown below

EXAMPLE

Let me try to explain here again with the same example I gave in Default and Dedicated Bearer section

Usually LTE networks with VoLTE implementations have two default and one dedicated bearer

Default bearer 1: Used for signaling messages (sip signaling) related to IMS network. It uses qci 5
Dedicated bearer: Used for VoLTE VoIP traffic. It uses qci 1 and is linked to default bearer 1
Default bearer 2: Used for all other smartphone traffic (video, chat, email, browser etc), assuming qci 9 is used here
This means that Default bearer 1 is associated with IMS PDN and has specific IP address. It has throughput limitations
defined in terms of A-AMBR and UE-AMBR. Since it has qci 5 which means that its IP packets has the highest priority
over other IP packets and maximum delay as 100ms between UE and PGW with packet loss percentage up to 10-6

Default bearer 2 is associated with internet PDN and has specific IP. It has throughput limitations defined in terms of A-
AMBR and UE-AMBR as well. Since it has qci 9 which means that its IP packets has the lowest priority over other IP
packets and maximum delay possible as 300ms between UE and PGW with packet loss percentage up to 10-6

Dedicated bearer will be linked to Default bearer 1 with L-EBI and it also has TFT which basically defines which IP
packets should be allowed to travel on this bearer. It has throughput limitations defined in terms of MBR and GBR. Since
it is using QCI 1, the IP packets traveling on this bearer have the second highest priority. The maximum delay possible to
IP packets on this bearer is 100 ms and the percentage of packet loss will be under 10-2

AKA Digest authentication scheme for VoLTE (IMS)

When a VoLTE client needs to connect to IMS network, it has to authenticate the network while network also
needs to make sure that only the correct user is registered to its network. AKA Digest is one of the scheme to
authenticate VoLTE client to the IMS server

AKA
AKA stands for "Authentication and key agreement". This scheme comes from the legacy 3gpp networks and
has been widely used in LTE, 3G, CDMA and WiMAX technologies. In this mechanism, a secret key is already
known to both user device (USIM, iSIM) and authentication servers (HSS, HLR).

The server will challenge the end user using AKA algorithms and shared key and sends RAND, AUTN values
towards UE. UE will authenticate network and prepares result (RES for network to authenticate UE) with the
help of shared key in UICC and parameters sent by Server.

HTTP Digest
Http Digest is the popular authentication scheme used for authenticating users to access web servers and
other applications which requires security and data integrity. This scheme is much secure than the basic
authentication as it applies hash function to the password before sending it [RFC2617].

HTTP Digest is username / password based authentication procedure. The authentication server provides one
time created " nonce " value to the client. The client uses the nonce value and creates a secure response that
contains the password, username and other parameters to the server. The password which is known both to
server and client, is always fixed

Now For IMS

Now since IMS is a part of 3GPP and on the contrary SIP signaling defines http digest for authentication
[RFC3261]. Therefore in order to use 3GPP AKA with IMS, the parameters from AKA are mapped onto http
digest [RFC3310]. In simple words the parameters / headers used to transport http digest information, will
transport AKA information in identical format

With 3GPP AKA digest, the "nonce" now contains RAND, AUTN. The password now contains the one time
RESPONSE generated by client with help of UICC (USIM, ISIM). Thus the method is even more secure.

Authentication in IMS networks

1) VoLTE Client sends SIP register request to IMS Server. The user is not authenticated at this point.
The SIP register request contains IMS related identities (private identity, public identity, URI, etc)
2) The IMS server (S-CSCF) obtains authentication vector and SQN from HSS that contains a random
challenge RAND, authentication token AUTN, expected authentication result XRES, a session key
for integrity check IK, and a session key for encryption CK
3) The server creates an authentication request, which contains the random challenge RAND, and the
network authenticator token AUTN
4) The authentication request is delivered to the client with "401 UNAUTHORIZED" message
5) The client verifies the AUTN with the ISIM. If the verification is successful, the network has been
authenticated. The client then produces an authentication response RES, using the shared secret K
and the random challenge RAND
 3GPP AKA Operation in IMS

I. The authentication response RES is delivered to the server using new regiser sip message
II. The server compares the authentication response RES with the expected response. If the two match,
the user has been successfully authenticated
III. Session keys IK and CK can be used for protecting further communications between the client and the
server
IV. Server sends "200 OK" message to inform the VoLTE client about successful registration

Semi persistent scheduling

Every VoIP packet is received / sent every 20ms when the user is talking whereas in silence period, discontinuous
transmission (DTX) is used to reduce the transmission rate. Also, in order to sustain voice quality, silent insertion
descriptor (SID) packet arrives every 160ms. The frequent arrival/transmission of VoIP packet means large control
overhead for lower layers (L1/L2) in the radio protocol stack. To deal with this issue, semi persistent scheduling plays an
important role.
Scheduling is a mechanism where UE requests eNB for the resource allocation during each transmission time interval
(TTI). If UE has some data that it needs to transmit continuously, it will request eNB every TTI for the resource allocation.
This scheduling type is dynamic scheduling. The advantage of dynamic scheduling is flexibility and diversity of resource
allocation but as mentioned, this results in huge L1/L2 load which in turn means inefficient use of scarce radio resources.

In case of semi persistent scheduling, eNB can assign predefined chunk of radio resources for VoIP users with interval of
20ms. Therefore, UE is not required to request resources each TTI, saving control plan overhead. This scheduling is semi -
persistent in the sense that eNB can change the resource allocation type or location if required for link adaptation or
other factors.

TTI Bundling

With all the hype created around IMS and LTE, operators have started questioning network vendors if they are
supporting RAN specific features for VoLTE. TTI bundling is one of the features among many others that can
help VoIP (VoLTE) calls in LTE.

TTI Bundling is LTE feature to improve coverage at cell edge or in poor radio conditions. UE has limited power in uplink
(only 23dBm for LTE) which can result in many re transmissions at cell edge (poor radio). Re transmission means delay
and control plan overhead which may not be acceptable for certain services like VoIP. To understand TTI bundling one
need to have the basic idea of Hybrid Automatic Repeat Request (HARQ) and Transmission Time interval (TTI).

HARQ
HARQ is a process where data at mac layer is protected against noisy wireless channels through error correction
mechanism. There are couple of different versions of HARQ but in LTE we have a type known as
'Incremental Redundancy Hybrid ARQ'. When receiver detects erroneous data, it doesn't discard it. On the other hand,
sender will send the same data again but this time, with different set of coded bits. The reciever will combine the
previously recieved erroneous data with newly attempted data by the sender. This way the chances of successfully
decoding the bits improve every time. This will repeat as long as the receiver is not able to decode the data. The
advantage of this method is that with each re-transmission, the coding rate is lowered. Whereas in other types of
HARQ, it might use the same coding rate in every re-transmission

TTI
TTI is LTE smallest unit of time in which eNB is capable of scheduling any user for uplink or downlink transmission. If a
user is receiving downlink data, then during each 1ms, eNB will assign resources and inform user where to look for its
downlink data through PDCCH channel. Check the following figure to understand the concep t of TTI
Now coming to TTI Bundling ...
HARQ is a process where receiver combines the new transmission every time with previous erroneous data. There is one
drawback however, that it can result in delay and too much control overhead in case of poor radio conditions if the
sender has to attempt many transmissions. For services like VoIP this means bad end user experience. Well, there is
another way- Instead of re-transmitting the erroneous data with new set of coded bits, why not send few versions
(redundancy versions) of the same set of bits in consecutive TTI and eNB sends back Ack when it successfully decodes
the bits. I hope the figure below will make it clear. This way we are avoiding delay and reducing control plane overhead
at mac layer

Voice solutions in LTE

The original idea behind LTE is that it would provide only wireless internet services. However, major revenue for cellular
operators comes from voice calls and SMS and therefore Voice in LTE has become a hot topic. Recently, I got an
opportunity to work with various voice solutions-the experience which I believe would be useful to share here.

LTE does not have a 'circuit switch core' which means that we cannot have voice calls as it is in 2G and 3G technologies.
In the initial LTE deployment cases however, operators are using their legacy networks along with their 4G network for
voice services.

So far we have heard of the following available voice solutions which I will discuss briefly.

 Circuit Switched Fall Back (CSFB)

 Simultaneous Voice and LTE (SV-LTE)
 Voice over LTE (VoLTE)
 Voice over LTE via Generic Access (VoLGA)
 Over the top (OTT)

Circuit Switched Fall Back

An operator who deployed LTE network, already owning a 3G or 2G network can take benefit from the feature called
"Circuit switched fall back'. The main idea is that 4G smartphones are going to have a radio capabilities for 3G/2G
networks as well. Such handsets can connect at a time either to LTE or 2G/3G . The shortcoming is that someone on
voice call will not be able to use LTE network for browsing or chatting etc.

CSFB for operator means very little investment since only few modifications are required in the network. Additional
interface between MME and MSC is required (SGs). CSFB solution has also been standardized by 3GPP and has gained
large industrial support.
Simultaneous Voice and LTE
SV-LTE is handset specific in which handset is capabile of usi ng two radios (LTE and WCDMA/GSM/CDMA) at one time.
So a user can use packet services from LTE while voice call can be made on other networks simultaneously unlike CSFB.
The shortfall here is high battery utilization due to dual radio operation.

For CDMA and LTE pair, the SV-LTE is the standard solution and being widely adopted. There are already SV-LTE
smartphones available in the market. I came across a few available for LGU+ in Korea and Verizon in USA. Both operate
LTE networks as an overlay to their old CDMA networks.

SV-LTE is the cheapest option for operators as no new modification is required to the network. Nevertheless, as
mentioned earlier it is at the cost of high battery utilization
Voice over LTE (IMS)

I believe this is going to be the most popular and widely adopted future voice solution for LTE. Instead of using legacy
networks, VoLTE utilizes IP Multimedia Subsystem (IMS) and provides voice services using the application layer on LTE.
IMS is a group of core network entities responsible for providing rich multimedia services over IP network. VoIP call,
SMS, MMS, LIVE TV are a few such services. IMS has been in the communication industry for long but with the
emergence of 4G networks, it is gaining popularity again.

Voice over LTE via Generic Access

I think, operators will accept VoLGA as a last option for voice capability. This solution uses CS core only from legacy
networks and also require new network elements. Therefore LTE handsets do not need 3G/2G radio capabilities since
radio part won't be used from legacy networks. Good thing about this solution however is that unlike CSFB, LTE handset
will be able to use voice and data simultaneously.
Over the top VoIP application
OTT is actually not LTE specific but a generic solution that we already have been using on 3G/WiFi networks. OTT
application is completely transparent to network and also out of operators' control. I am talking about
generic VoIP clients like Viber, skype, Tango etc. They do not give the real taste of voice flexibility as in other 3GPP
networks and also lack the QoS for voice. Nonetheless, these will be widely used by the consumers as an alternative,
because of the fact that it gives them full flexibility to choose their own service.

Semi persistent scheduling

Every VoIP packet is received / sent every 20ms when the user is talking whereas in silence period, discontinuous
transmission (DTX) is used to reduce the transmission rate. Also, in order to sustain voice quality, silent insertion
descriptor (SID) packet arrives every 160ms. The frequent arrival/transmission of VoIP packet means large control
overhead for lower layers (L1/L2) in the radio protocol stack. To deal wi th this issue, semi persistent scheduling plays an
important role.

Scheduling is a mechanism where UE requests eNB for the resource allocation during each transmission time interval
(TTI). If UE has some data that it needs to transmit continuously, it wil l request eNB every TTI for the resource allocation.
This scheduling type is dynamic scheduling. The advantage of dynamic scheduling is flexibility and diversity of resource
allocation but as mentioned, this results in huge L1/L2 load which in turn means i nefficient use of scarce radio resources.

In case of semi persistent scheduling, eNB can assign predefined chunk of radio resources for VoIP users with interval of
20ms. Therefore, UE is not required to request resources each TTI, saving control plan overhead. This scheduling is semi-
persistent in the sense that eNB can change the resource allocation type or location if required for link adaptation or
other factors.
TTI Bundling
With all the hype created around IMS and LTE, operators have started questioning network vendors if they are
supporting RAN specific features for VoLTE. TTI bundling is one of the features among many others that can
help VoIP (VoLTE) calls in LTE.

TTI Bundling is LTE feature to improve coverage at cell edge or in poor radio condi tions. UE has limited power in uplink
(only 23dBm for LTE) which can result in many re transmissions at cell edge (poor radio). Re transmission means delay
and control plan overhead which may not be acceptable for certain services like VoIP. To understand TTI bundling one
need to have the basic idea of Hybrid Automatic Repeat Request (HARQ) and Transmission Time interval (TTI).

HARQ
HARQ is a process where data at mac layer is protected against noisy wireless channels through error correction
mechanism. There are couple of different versions of HARQ but in LTE we have a type known as
'Incremental Redundancy Hybrid ARQ'. When receiver detects erroneous data, it doesn't discard it. On the other hand,
sender will send the same data again but this time, with different set of coded bits. The reciever will combine the
previously recieved erroneous data with newly attempted data by the sender. This way the chances of successfully
decoding the bits improve every time. This will repeat as long as the receiver is not able to decode the data. The
advantage of this method is that with each re-transmission, the coding rate is lowered. Whereas in other types of
HARQ, it might use the same coding rate in every re-transmission

TTI
TTI is LTE smallest unit of time in which eNB is capable of scheduling any user for uplink or downlink transmission. If a
user is receiving downlink data, then during each 1ms, eNB will assign resources and inform user where to look for its
downlink data through PDCCH channel. Check the following figure to understand the concept of TTI

Now coming to TTI Bundling ...

HARQ is a process where receiver combines the new transmission every time with previous erroneous data. There is one
drawback however, that it can result in delay and too much control overhead in case of poor radio conditions if the
sender has to attempt many transmissions. For services like VoIP this means bad end user experience. Well, there is
another way- Instead of re-transmitting the erroneous data with new set of coded bits, why not send few versions
(redundancy versions) of the same set of bits in consecutive TTI and eNB sends back Ack when it successfully decodes
the bits. I hope the figure below will make it clear. This way we are avoiding delay and reducing control plane ove rhead
at mac layer

Voice solutions in LTE

The original idea behind LTE is that it would provide only wireless internet services. However, major revenue for cellular
operators comes from voice calls and SMS and therefore Voice in LTE has become a hot topic. Recently, I got an
opportunity to work with various voice solutions-the experience which I believe would be useful to share here.

LTE does not have a 'circuit switch core' which means that we cannot have voice calls as it is in 2G and 3G technologies.
In the initial LTE deployment cases however, operators are using their legacy networks along with their 4G network for
voice services.
So far we have heard of the following available voice solutions which I will discuss briefly.

 Circuit Switched Fall Back (CSFB)

 Simultaneous Voice and LTE (SV-LTE)
 Voice over LTE (VoLTE)
 Voice over LTE via Generic Access (VoLGA)
 Over the top (OTT)

Circuit Switched Fall Back

An operator who deployed LTE network, already owning a 3G or 2G network can take benefit from the feature called
"Circuit switched fall back'. The main idea is that 4G smartphones are going to have a radio capabilities for 3G/2G
networks as well. Such handsets can connect at a time either to LTE or 2G/3G . The shortcoming is that someone on
voice call will not be able to use LTE network for browsing or chatting etc.

CSFB for operator means very little investment since only few modifications are required in the network. Additional
interface between MME and MSC is required (SGs). CSFB solution has also been standardized by 3GPP and has gained
large industrial support.
Simultaneous Voice and LTE
SV-LTE is handset specific in which handset is capabile of using two radios (LTE and WCDMA/GSM/CDMA) at one time.
So a user can use packet services from LTE while voice call can be made on other networks simultaneously unlike CSFB.
The shortfall here is high battery utilization due to dual radio operation.

For CDMA and LTE pair, the SV-LTE is the standard solution and being widely adopted. There are already SV-LTE
smartphones available in the market. I came across a few available for LGU+ in Korea and Verizon in USA. Both operate
LTE networks as an overlay to their old CDMA networks.

SV-LTE is the cheapest option for operators as no new modification is required to the network. Nevertheless, as
mentioned earlier it is at the cost of high battery utilization
Voice over LTE (IMS)

I believe this is going to be the most popular and widely adopted future voice solution for LTE. Instead of using legacy
networks, VoLTE utilizes IP Multimedia Subsystem (IMS) and provides voice services using the application layer on LTE.
IMS is a group of core network entities responsible for providing rich multimedia services over IP network. VoIP call,
SMS, MMS, LIVE TV are a few such services. IMS has been in the communication industry for long but with the
emergence of 4G networks, it is gaining popularity again.

Voice over LTE via Generic Access

I think, operators will accept VoLGA as a last option for voice capability. This solution uses CS core only from legacy
networks and also require new network elements. Therefore LTE handsets do not need 3G/2G radio capabilities since
radio part won't be used from legacy networks. Good thing about this solution however is that unlike CSFB, LTE handset
will be able to use voice and data simultaneously.
Over the top VoIP application
OTT is actually not LTE specific but a generic solution that we already have been using on 3G/WiFi networks. OTT
application is completely transparent to network and also out of operators' control. I am talking about
generic VoIP clients like Viber, skype, Tango etc. They do not give the real taste of voice flexibility as in other 3GPP
networks and also lack the QoS for voice. Nonetheless, these will be widely used by the consumers as an alternative,
because of the fact that it gives them full flexibility to choose their own service.

Parameters
This section describes the parameters introduced by the TTI Bundling feature and
parameters affected by activating the feature.
Introduced Parameters
The following table describes the parameters introduced by the TTI Bundling feature:

Parameter Description

Activates or deactivates the licensed feature TTI Bundling. The value of the
featureStateTtiBundling attribute is irrelevant when no valid license key is installed for the feature.

keyIdTtiBundling The license key ID for the TTI Bundling feature.

The license status of feature TTI Bundling, ENABLED or DISABLED. The

licenseStateTtiBundling value is ENABLED when a license key is installed.

serviceStateTtiBundling Indicates if the feature TTI Bundling is operable or inoperable.

TtiBundlingId The value component of the RDN (Relative Distinguished Name).

Affected Parameters

The following table describes the parameters affected by the TTI Bundling feature:

Parameter Description

QciProfileOperatorDefined: Indicates the service that the bearer is used for VoIP. It shall be set to
ServiceType or VOIP for one bearer if the UE shall use TTI Bundling.
QciProfilePredefined:
ServiceType

The TTI Bundling feature can affect DRX: the DRX parameter
DRX parameter: OnDurationTimer. If the value of OnDurationTimer is less than 4 in the
OnDurationTimer MOM it will be set to 4 for UEs that are configured to use TTI Bundling.

New Counter Validation

The following table lists the important counters associated with the TTI Bundling
feature:

Counter Description

Sum of all sample values recorded for number of UEs using TTI
pmTtiBundlingUeSum Bundling.

pmTtiBundlingUeSamp Counts the number of times the corresponding pmTtiBundlingUeSum has

 been incremented.

Maximum sample value of number of UEs using TTI Bundling.

pmTtiBundlingUeMax

pmVoipQualityUeUlOk Number of UE that are satisfied with their VoIP quality.

pmVoipQualityUeUlNok Number of UE that are not satisfied with their VoIP quality

Number of VoIP quality measurements with low number of samples while

pmVoipQualityUeUlLowSampl the VoIP satisfaction of a UE is between 50% and 99%