TELEPHONE EXCHANGE INTRODUCTION

TELEPHONE
• The Telephone use came about from a device
invented by Alexander Graham Bell in 1876 .
• Telephone is devices used for communicating
sound, especially speech.
• Tele- means long distances and
• phono means sound, voice or speech
 TELEPHONE EXCHANGE
• Telephone Exchange is a place where switching between two
or more subscribers is done through either manually or
electronically.
In addition to switching, signalling, controlling and diagnostic
are also done at “Exchange”
• A telephone exchange is a telecommunications system used in
the public switched telephone network or in large enterprises.
• Basics elements of Telephone Exchange
1. Card Frames ,shelf's
2. Mother Board, cards
3. Main Distribution Frame
4. Power Supply Panel & Protective Devices
 EVOLUTION
• PSTN - Public switched telephone network, also
knows as the plain old telephone system(POTS) is
basically the inter-connected telephone system over
which telephone calls are made via copper wires.
• PSTN is based on the principles of circuit switching
• Therefore when a call is made a particular dedicated
circuit activates which eventually deactivates when
the call ends
• Telephone calls transmits as analogue signals across
copper wires
 MANUAL EXCHANGE
• 1877 - The world's first commercial telephone
exchange opened in Friedrichsberg close to Berlin
• 1878 – Worlds’ first US telephone exchange was
established at New-Haven in Connecticut in the
USA
• Telephones were sold in pairs and the customers
were supposed to lay out there own cables
• Connectivity type – point to point connections
• Network structure – mesh topology, star topology
• Switching technique – manual switching
MANUAL EXCHANGES
ELECTRO MECHANICAL EXCHANGE
• 1887 – Almon Brown Strowger invented the
first electromechanical switch, known as the
Strowger switch or step by step switch
• Switch operated according to the train of
pulses generated by the customer premises
telephone
• Pulses were generated by a telegraph key on
the telephone until the dial was introduces
• 1920’s – Rotary dial telephones enters service
ELECTRO MECHANICAL EXCHANGE
• 1935 – Crossbar switches were introduced
• Intersecting bars move to make contact in
order to complete the circuit
• Markers were used to control exchanges
• Takes only 1/10 of a second to complete a call
ELECTRO MECHANICAL EXCHANGE
CROSS BAR SWITCH
 ELECTRONIC EXCHANGE
• 1950 – Time division multiplexing (TDM) is
introduced
• 1968 – Stored program control (SPC) electronic
switching was introduced
• calls are completed within nano seconds
• An electronic switch Upgradable to new versions
since software dependant Call set-up is controlled
by programmed software's
• 1984 – ISDN exchange are introduced.
ELECTRONIC EXCHANGE
 TELEPHONY WORKS
• Pick up the phone
– Wait for a dial-tone
• Dial the destination tel. #
• Remote phone starts ringing
– Caller is alerted of the ringing by ringback tone
• Destination picks up and
– A point-to-point circuit is established
• The circuit carries a digitized version of the voice samples
– E.g., 8 bits/voice sample, one sample at a time, PCM
(pulse code Modulation)
• If Any one whether the caller or called cradles the phone
than the circuit will be disconnected.
CIRCUIT SWITCHING
 CIRCUIT SWITCHING
• In Circuit Switching the resources need to be
reserved during the setup phase, the
resources remain dedicated for the entire
duration of data transfer until the teardown
phase.
 CIRCUIT SWITCHING
1. Waste of channel bandwidth

2. Will make a dedicated path

 TELEPHONY WORKS
• The circuit and BW allocated to the call are
devoted to
– Only one conversation
• During the time between the digitized samples
and the silent periods
– The circuit is idle carrying no information
• All this time is wasted
– 50-70% of the Band Width
TIME SPACE SWITCH
Switching Matrix
 Circuit Switching
• Advantages :
1. Once the circuit has been set up,
communication is fast and without error.
2. It is highly reliable
• Disadvantages:
1. Wastes a lot of bandwidth, especial in speech
whereby a user is sometimes listening, and
not talking.
2. Channel set up may take longer
Limitations of Circuit Switching Exchanges

• In 1990s, things changed, Internet emerged as a

game changer.
• In the 2000s IP started becoming the driving factor
for communication.
• Data became important than voice.
• The architecture built for voice is not flexible
enough to carry data.
• Expensive upgrades and operating expenses.
• Limited migration strategy to new tech .
VOIP EXCHANGES
VOIP-Voice Conversion
 VOIP EXCHANGES
• VOIP (Voice over Internet Protocol) is a
technology that allows telephone calls to be
made over computer networks like the Internet.
• Packetization and real-time transport of classic
telephone system audio over an IP network.
• Allows 2-way voice transmission over broadband
connection.
• Also called IP Telephony, Internet telephony,
broad-band telephony etc
 VOIP EXCHANGES
• Traditional Digital Telephony involves signalling,
channel setup, digitization of the analog voice
signals, and encoding. And are being transmitted
over a circuit-switched network.
• In VOIP the digital information is packetized, and
transmission occurs as IP packets over a packet-
switched network and transport audio streams
using special media delivery protocols that encode
audio and video with audio codecs, and video
codecs
Circuit Switched & Packet Networks
Converged Network
 VoIP – System Components
• IP Exchange (Server)
• VoIP Software
• IP Telephone
• VoIP Gateway (Analog or Digital)
• Network Switches (with PoE)
 IP SERVER(EXCHANGE)
• This is a common server on which a VoIP
software is installed and configured to provide
voice telephony.
• Some vendors sell propriety servers as well in
the name of features.
• Board has issued a policy guideline to use
open standard telephony.
 VoIP SOFTWARE
• Open Source
– Asterisk, Freeswitch, SipXecs, yate etc.
• Propriety
– Cisco, Tadiran, Microsoft etc.

Open Source Telephony software :

• Open source software provides users the freedom of choice in
programming according to user requirements.
• It eliminates the vendor lock-in as well as promotes openness and
standardization such as (license for server software, license for IP
phone, license for gate ways , license for concurrent calls, etc)
 IP TELEPHONES
• Looks like the ordinary traditional phone.
• Works like a modern mobile phone.
• Dial tone is local. Dial the number and then
send the same to the exchange.
IP Phones
Connectivity of IP Phones
 VoIP GATEWAYS
• VoIP gateway is a device that converts
telephony traffic into a IP for transmission
over a data (IP) network and vice-verse.
• Can be Analog or Digital.
• Analog VoIP gateways are used to connect
ordinary Push-Button telephone to a VoIP
system.
• Analog gateways – FXS, FXO
GATE WAYS
 GATEWAYS
• FXS used to connect the PBT (station) to the
gateway.
• FXO used to connect CO Line (office) to the
gateway.
• Digital gateways used to connect a PRI line to
theVoIP system.
 Network Switches
• These are normal switches but support Power
over Ethernet (PoE) on each port.
• IEEE 802.3af-2003

• IEEE 802.3at-2009
– 30W of DC power/Power available – 25.5W
 AUDIO CODECS in VOIP EXCHANGES

• Popular audio codecs

• μ-law and a-law versions of G.711, G.722,
• a popular open source voice codec known as
iLBC,
• a codec that only uses 8 kbit/s each way called
G.729
VOICE CODECS
 VOICE CODECS
• ADPCM: Adaptive Differential Pulse Code
Modulation,
• LD-CELP: Low Delay Code Excited Linear
Predictive
• CS-ACELP: Conjugates Structure Algebraic
Code Excited Linear Predictive,
• MP-MLQ: Multi Pulse, MultiLevel Quantization
• MOS :Mean opinion score
 VOIP CODECS

• A codec transforms analog signals into digital format.

• Different codecs have different bandwidth
requirements:
• G.711: The G.711 codec uses the most bandwidth. It encodes each
of the 8000 samples that are taken each second into 8 bits, resulting
in a 64-kbps codec bandwidth.
• G.722: The G.722 wideband codec splits the input signal into two
sub-bands and uses a modified version of adaptive differential pulse
code modulation (ADPCM) (including adaptive prediction) for each
band. The bandwidth of G.722 is 64, 56, or 48 kbps.
• G.726: The G.726 ADPCM coding schemes uses less bandwidth.
These coding schemes encode each of the 8000 samples that are
taken each second using 4, 3, or 2 bits, resulting in bandwidths of 32,
24, or 16 kbps.
 VOIP CODECS
• G.728: The G.728 low-delay code excited linear prediction (LDCELP)
coding scheme compresses pulse code modulation (PCM) samples
using a codebook. Waveshapes of five samples are represented by a
10-bit codeword, which identifies the best matching pattern of the
codebook. Because of the compression of five samples (worth 40
bits in PCM) to 10 bits, the bandwidth of LDCELP is 16 kbps.
• G.729: The G.729 conjugate structure algebraic code excited linear
prediction (CS-ACELP) coding scheme also offers codebook-based
compression. Waveshapes of 10 bits are represented by a 10-bit
codeword, reducing the bandwidth to 8 kbps.
• The bandwidth codec, however, indicates only the
bandwidth that is required for the digitized voice itself. It
does not include any packetization overhead.
VOIP PROTOCOLS and the OSI Model
 The Standard VOIP Packet Format
VoIP
Packet
Link IP UDP RTP Voice
Header Header Header Header Payload

X Bytes 20 Bytes 8 Bytes 12 Bytes X Bytes

G.711 Standard has a 160 Byte Voice Payload

G.729 Standard has a 20 Byte Voice Payload
G.723.1 Standard has a 24 Byte Voice Payload
VOIP PACKET
 VOIP PACKET
• https://planetcalc.com/3144/
RTP HEADER
 VOIP BANDWIDTH
• VOIP traffic bandwidth includes useful audio
data payload and protocol stack overhead
(RTP, UDP, IP, and network L2, L1 overhead).
bandwidth value in kilobits per second (Kbps).
The algorithm is quite simple:

• Where
Lvoip = Audio data encoded by codec
Opacket = Protocol stack overhead
C second = Number of Packets
 BANDWIDTH CALCULATION
• Example calculation for G.729 (8Kbps) audio codec (up
to network L2 layer):
• VOIP data size: 20(codec sample size)
1(samples per packet) = 20 bytes
• Protocol stack overhead
RTP-L2:12(RTP)+8(UDP)+20(IP)+18(L2)=58 bytes
• Packets per second: 1000(milliseconds in a
second)/20(packet duration in ms) = 50 packets

• Bandwidth:(20+58)50*8/1000=31.2 Kilobits per second

 45 msec 64 msec < 100 msec

Sample Encode Packetize Transmit

Speaking

Network
45 msec 64 msec
Hearing
Output Decode Jitter Reconstruct Receive
Buffer

Total of Less Than 250 msecs of Delay is Tolerable

But Delay of Less than 150 msec is the Standard
 QoS Requirements for Voice

• Voice calls, either one-to-one or on a conference connection

capability, require the following:
• ≤ 150 ms of one-way latency from mouth to ear (per the ITU
G.114 standard)
• ≤ 30 ms jitter
• ≤ 1 percent packet loss
• 17 to 106 kbps of guaranteed priority bandwidth per call
(depending on the sampling rate, codec, and Layer 2
overhead)
• 150 bps (plus Layer 2 overhead) per phone of guaranteed
bandwidth for voice control traffic
 ADVANTAGES of VOIP
• Portability :convenience of being able to make
and receive calls from any location using the
same phone number VoIP takes the lead.
• Scalability: VoIP network is perfect for small
and large business communities can be
expended at any time just by increasing the
license at server, and connecting additional
voip phones to already established lan
network.
 ADVANTAGES of VOIP
• Flexibility: More flexible for implementation of the VoIP
network. there is no need for additional hardware for
expansion ( such as expansion of shelfs, cards, license,
outdoor network, etc) since the only device required for
VoIP service performance is the VoIP phone system.
• COST: VoIP calls carried over the Internet are cheaper and
can save a lot of money especially for large enterprises that
have to handle a huge number of calls on daily basis
• Multi functionality: Call forwarding, call waiting, paging,
group calls, speed dialling and lots of other features deliver
more enhanced call processing opportunities that can bring
to higher productivity.
 Disadvantages of VOIP

• No service during a power outage :

• Reliability :your VOIP service will be affected by the quality
and reliability of your Network . Poor internet network and
congestion can result in garbled or distorted voice quality
• Security :Security is a main concern with VoIP, as it is with
other Internet technologies. The most prominent security
issues over VoIP are identity and service theft, viruses and
malware, denial of service, spamming, call tampering and
phishing attacks
 VOIP PROTOCOL STANDARDS
• ITU: International Telecommunication Union
H.323 -ITU recommends for “ Packet based Multimedia communication
systems”.
- Most common VoIP protocol
- Distributed Architecture

• IETF: Internet Engineering Task Force

SIP: Session Initiation Protocol
-IETF RFC 2543 ,3265
-Distributed Architecture
Real-time Transport Protocol – RTP
- A transport protocol for real-time application
- IETF RFC 1889 ,3550
- Provides transport for audio/media of VoIP communication
-Used by All of VoIP signaling protocols
 VOIP PROTOCOL STANDARDS
• MGCP Media Gateway Control Protocol
- IETF RFC 2075 ,3435
- Centralized Architecture for Multimedia
applications such as VoIP

• H.248 Gateway Control Protocol

Collaboration between ITU & IETF referred to
as IETF RFC 2885,3015 (MEGACO)
 PROTOCOL STANDARDS
Three major VOIP Signaling Protocols:
–ITU-T H.323 –a collection of protocols
▪Specifies “Packet Based Multimedia Communications System”
▪Currently most mature and is popular in enterprise networks
⌐Supported by many vendors
□Microsoft Netmeeting (MM conferencing) is based on H.323
–IETF MEGACO = ITU-T H.248
▪Telephony signaling protocol based on existing PSTN
▪Upgrade of earlier MGCP
⌐Currently supported by many vendors
▪Used by VOIP telephony service providers
–IETF SIP
▪Client-Server protocol for telephony applications over IP networks
▪Moves application control to the endpoints
▪Supported by some vendors
⌐Microsoft Windows Messenger (Instant Messaging) is based on SIP
▪Used by some ISPs
▪May be more common in future
 Centralized Architecture
• Mostly used in older networks
• Worked well for basic telephony services
• Trade off between easy management and
endpoint innovation
• Associated with MGCP and H.248
• Intelligence focused in centralized Gateway
unit (media agent)
• Endpoints are relatively or completely dumb.
 Distributed Architecture
• Associated with H.323 and SIP protocol
• Allows Network Intelligence to be distributed between endpoints
and control devices.
• Intelligence:
Call state
Calling features
Calling routines
any aspect of call handling
• More flexible
• VoIP is treated like any other distributed IP application
• Well understood by engineers who design and run IP data networks
• More complex than the Centralized Architecture
VOIP PROTOCOLS
H.323 Network Architecture and
Components
 H.323 protocol
• H.323 terminal handles RTP for MM
• Gateway connects one or more PSTN, N-ISDN, B-ISDN to LAN
–Provides audio/video transcoding
• Gatekeeper does address translation and admission control
• Multi-point Control Unit controls multimedia conferencing
• H.323 Zone is the collection of all terminals, gateways, and MCU managed by a single
Gatekeeper

• ISDN = Integrated Switched Digital Network

• MCU = Multi-point Control Unit
• MM= Multimedia
• POTS = Plain Old Telephone Service
• RTP = Real Time Protocol
H.323 Components And Protocols
 H.323 Components
H.323 elements include terminals, gateways, gatekeepers and Multipoint
Control Units (MCUs).

Terminals
Also known as endpoints, terminals provide point-to-point and multipoint
conferencing for audio, video and data
Gateways
Gateways are used to connect between Switched Circuit Network (SCN)
endpoints and H.323 endpoints. Gateways are only needed when an H.323
endpoint needs to interconnect to a different network
Gatekeeper
Gatekeepers provides pre-call and call-level control services to H.323
endpoints.
 H.323 Components
Multipoint Controller (MC)
A Multipoint Controller supports conferencing
between three or more endpoints. A Multipoint
Processor (MP) receives audio, video and data
streams, and then redistributes those streams to
the endpoints in a multipoint conference
H.323 Protocol Stack
Call Exchange (direct mode)
 H.323 Call-Signaling Process
There are five general steps in the H.323 signaling process:
setup/teardown, capabilities negotiation, open media channel,
perform call, and release.
 
Setup/Teardown
To initiate an H.323 call, H.225 is required for the setup process.
The following are the most commonly used signaling messages :
Setup: A forward message sent by a calling entity in an attempt to
establish a connection with the called entity
Proceeding: A backward message sent from the called entity to the
calling entity to inform that call establishment procedures were
initiated
 H.323 Call-Signaling Process
Alerting: A backward message sent from the
called entity to inform that called party ringing
was initiated
Connect: A backward message sent from the
called entity to the calling entity that the called
party answered the call. The connect message can
contain the transport UDP/IP address for H.245
control signaling
Release: sent by endpoint initiating disconnect
 SIP
• Proposed Standard described in IETF RFC 2543
• Application-layer control protocol
• A signaling protocol for initiating, managing and
terminating voice and audio session across packet
networks with one or more participants .
• Text-based protocol with highly extensible
• Session can be
1. Call between two simple telephone
2. Collaborative multi-media conference session, etc.
 SIP Functionality
• User location
• User availability
• User Capabilities
• Session setup
• Session Management
• Session termination
 FUNCTIONALITIES
• SIP serves 4 major functionalities
1. It allows to locate the user ( i.e translating user’s name
to their current network address)
2. Inviting the user for session
-negotiation so that all of the participants in a session
can agree on the features to be supported among them
3. Delivering the session description
- call management such as adding ,dropping or
transferring participants.
4. Terminate the session
SIP ARCHITECTURE
 Application Services

SIP Servers
SIP Servers SIP Proxy, Locate,
Register, Redirect
Processes

SIP

SIP
SIP
PSTN

SIP

RTP/UDP
PBX

IP Device w ith
SIP Agents
 SIP entities
• User Agent (UA)
– User agent client (UAC)
– User agent server (UAS)
• Proxy server
– Stateless proxy server
– Stateful proxy server
• Redirect server
• Registrar server
 USER AGENT(UA)
• User Agent Client (UAC)
–Application which originates SIP requests
• User Agent Server (UAS)
–Application which contacts user upon receiving SIP request, and–Returns
user’s response on his behalf
- Accepts, rejects or redirects
• User Agent (UA)
–Application which contains both UAC & UAS and exchange
request/response messages

• UA is a piece of software that can be placed in a computer or a laptop

Therefore, SIP can offer –Various telephony services,
e.g., ▪Internet phones-to-Internet phones
▪Internet phones-to-PSTN phones
▪PC phones-to-PC phones
 SIP SERVERS
• proxy server: The Proxy Servers are application layer
routers that forward SIP request & responses
• Redirect server: A redirect server is a server that
accepts a SIP Requests & then return the location of
another SIP user agent & server where the user
might be found.
• Registrar server: A registrar is a server that accepts
REGISTER requests. A registrar is typically co-located
with a proxy or redirect server and offer location
services
 SIP operation

1.invitation 2.invitation

4. OK 3. OK

5.Acknowledge 6.Acknowledge
USER USER
AGENT 1 AGENT2

7. Audio/Video data 7. Audio/Video data

From: Thomas Doumas

Next Generation Telephony: A Look at Session Initiation Protocol White Paper
SIP Protocol Structure
Client
 INVITE Transaction
Client
 ACK
 Sending
Non INVITE Transaction
Request
 Receiving
MatchingResponse
Requests to
Client Transactions
Server
Server
Receiving Request
 Sending
INVITE Response
Transaction
 Non INVITE Transaction
 Matching Requests to
Framing
Server Transactions
Error
ErrorHandling
Handling

From: http://docs.sun.com/app/docs/doc/821-0203/6nl988v7d
SIP LAYERS
 SIP Protocols
SIP provides basic elements of telephony
• SIP: Call setup and termination
• RTCP: data stream management
• DNS = Domain Name System
• PPP = Point-to-Pont Protocol
• RSTP = Real-time Streaming Protocol (controls video
streams, like a VCR)
• RSVP = Resource Reservation Protocol
• RTP = Real-Time Transfer Protocol
• SDP = Session Description Protocol
 SIP REQUESTS
•INVITE
–Request initiation of a session
Most common and important
•ACK
–Confirm that a session has been initiated
•BYE
–Request termination of a session
•OPTIONS
–Query a host about its capabilities
•CANCEL
–Cancel a pending request
•REGISTER
–Inform a redirection server about the user’s current location
 SIP RESPONSES

•1xx-Provisional (Informational)
–Request received, continued to process request, e.g., 180 = Ringing
•2xx–Success
–Action was successfully received, understood, and accepted, e.g., 200 =
OK
•3xx –Redirection
–Further action must be taken to complete the request, e.g., 305 = Use
Proxy
•4xx-Client Error
–The request contains bad syntax or cannot be fulfilled at the server, e.g.,
484 = Address Incomplete
•5xx -Server Error
–The server failed to fulfill an apparently valid request, e.g., 500 = Internal
Server error
•6xx-Global Failure
–The request is invalid on any server, e.g., 600 = Busy
 Elements of VOIP
HARDWARE ELEMENTS
1. SERVER : A standard robust server should be
used for this purpose
2. GATEWAYS
a. PRI GATEWAYS:- for connecting the existing
TDM Exchanges
b. ANALOG GATEWAYS :-for connecting the
normal analog subscribers.
 Elements of VOIP
3.POE SWITCHES :- for connecting and powering
the IP PHONEs, IP VIDEO PHONEs along the
network.
4.SIP PHONES: IP audio phone , IP audio cum
videophone
5. LAN NETWORK
GATE WAYS
SIP Phones
 VOIP SOFTWAREs
Open Source Telephony software :
• Open source software provides users the freedom of choice in
programming according to user requirements.
• It eliminates the vendor lock-in as well as promotes openness and
standardization such as (license for server software, license for IP
phone, license for gate ways , license for concurrent calls, etc)
• Various open source telephony are available in market some of the
are enlisted below
• 1. Asterisk
• 2. Open SIPS
• 3. Free Switch
• 4. YATE
 ASTERISK
• Asterisk is an open source software for implementation of
IP-PBX developed by DIGIUM corporation ,USA in 1999
• It is designed for the linux OS and can be installed on
either PC servers or compatible embedded hardware.
• It is free downloadable GPL (Gnu Public License) open
source software, can be down loaded from
WWW.asterisk.org site.
• The software includes source code and can be modified
according to user requirements under the terms of the
GPLv2 license
 FEATURES OF ASTERISK
• Various modern features available in asterisk
1. Call features: automated attendant, blind transfer, cdr ,
call forward, Conference bridging, sms messaging,
streaming media access, talk detection, voice mail , etc.
2. Computer –telephony Integration
3. Allows Scalability
4. Audio Codec Support (in standard Distribution)
5. Traditional telephony protocols support (with add-on
hardware
6. ISDN protocols support (with add-on hardware)
 Server and OS requirements
• Server for 100 subscribers for handling 50 concurrent calls
• Server shall be suitable for 24X7 operations (Dell, IBM ,HP,
etc)
• Server should be installed in 1+1 redundancy for critical
uses.
• The processor should be intel dual core minimum1.8 GHz
processor with 2 GB RAM,256 MB cache , RAID1 Dual hard
disk 250GB minimum, with dual Ethernet ports to enable
redundant connection to LAN
• OS shall be recent standard linux distribution.
 Server and OS requirements
• Server for 1000 to 1200 subscribers for handling 500
concurrent calls
• Server shall be suitable for 24X7 operations (Dell, IBM ,HP, etc)
• Server should be installed in 1+1 redundancy for critical
uses .the second server may be provided at geographically
different location.
• The processor should be intel quad core minimum 1.8 Ghz
processor with 8 GB RAM,512 MB cache , RAID5 Dual hard disk
500GB minimum, with dual Ethernet ports to enable
redundant connection to LAN
• OS shall be recent standard linux distribution.