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Agenda
 Evolution Of Telephony
 Modern Telephone Networks
 Limitations of Voice Networks
 Evolution of Packet Switching Technology
 Advancement in data networks
 Why packet based telephony
 VoIP signaling protocols
 H.323
 SIP

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 Digital Signaling—
Better and Cheaper

Distorted Analog Clear Digital

Communications Communications

2 x 4 KHz = 8 KHz
300–3400 Hz  4 KHz
8 KHz x 8 Bits = 64 kbps
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The Dawn of Digital Telephony
 Programmable Call routing logic
– Enables more sophisticated logic
– Customer can create new applications and is
provided greater control
 Digital trunks
– Better Voice Quality (especially long distance)
– Combine many calls onto single trunk line
 Digital Signaling
– More efficient and sophisticated signaling
– Fast and accurate call setup

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The Dawn of Digital Telephony
30 Channel PCM
CO Switch
Voice & Signaling

CO Switch CO Switch

Analogue
Loop Start Access loop
Signaling E1 Leased Line
2 Wire
PBX

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 Limitations of Voice Network
 64 K reserved end to end bandwidth to every
user
– Voice network has been optimized for voice
– A user requiring more bandwidth to transmit
information has to slow down for want of
bandwidth as one user can get only 64K
– Another user at that moment might be sending
data at rate lower than 64 K thus wasting the
bandwidth
– Inefficient utilization of resources
 Due to monolithic nature of voice technology it
is always difficult to deploy new services
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 Limitations of Voice Network
 Voice networks were not designed to carry data
– Voice is delay sensitive where as data is error
sensitive
– Voice traffic is smooth where as data traffic is
bursty in nature
– STDM technology has been chosen for data
where as voice networks are TDM technology
– Packet switching is more suitable for data as it
can absorb bursts and bandwidth can be
allocated on demand
– Packet switching utilizes bandwidth more
efficiently

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 Evolution of Data Networks
 Most data applications by nature require a
variable amount of bandwidth
 Packet switching is conducive to data transport
as it treats the transmission line as shared
medium and bandwidth is allocated on demand
 It does not allocate each user a fixed sized
channel through the network
 If a given user does not has data to send for a
period of time , it simply uses the bandwidth to
send someone else’s data
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 Evolution of Data Networks
 With the growing nos of PC population data
communication became a reality
 Various data communication technology
evolved in 60s and 70s
– 1960s – TCP/IP (ARPANET)
– 1970s – X.25
– 1980s – Frame Relay
– 1990s – ATM
 All these data communication technologies
depend on packet switching technology
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 The Insurgent Internet
 In 1960s ARPANET started connecting
different universities and research institutions
in USA and Europe
 ARPANET grew in size and complexity
 Eventually in 1990s Internet became a reality
for civilian use
 Internet is a ubiquitous network covering every
part of the globe
 Cheapest means of communication and has
penetrated the life of common man in west

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 The Insurgent Internet
 Internet is a reality and it is going to stay
 Due to its global reach and seamless nature,
Internet/IP network is providing varieties of
new services to the users which is not possible
by any other means of technology
 Due to its layered model approach the IP
network has not been tailor-made for any
specific application. It can carry data from any
application i.e voice, video, data etc
 Creation/deployment of new services is easy
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 The Insurgent Internet
 Modern data networks now have the capacity,
speed and reliability to support advanced
applications
– Interactive voice
– Broadband streaming video and audio
– Interactive gaming applications with advanced
3D graphics

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The Multi-service IP Network

IP Network

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Voice and Data Network
Growth Factors

Data
Network Traffic

Voice

1999 2003 Time

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Voice Vs Data Traffic Growth
YEAR Voice Traffic Volume % Data Traffic Volume %
1996 98 2
1997 96 4
1998 93 7
1999 87 13
2000 78 22
2001 65 35
2002 51 49
2003 41 59

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 Divergent Network

PSTN Internet

Voice Network Data Network

Cable TV
Network

TV Network

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 One Common Digital Network
Digital Services
–Data
–Voice
–Digital Audio
–HDTV
–Video On Demand
–Online Games

VFS

IP
Digital Broadcasting
Head-End

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 Why Packet Based Telephony
 Growth Potential
– Data Networks has unanimously been
recognized as growth Industry
 Voice Networks are still earning the major
portion of the revenue for the operators
– Voice revenue is still much higher than data
revenue
 Reduced Cost
– One network is better than managing two
different networks thus reduced cost of
operation

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 Why Packet Based Telephony
 Packet based voice systems are efficient
– It benefits from the variable nature of modern
voice coding techniques and employ voice
compression and silence suppression
 Deployment of New services
– Deployment of new applications difficult with
conventional voice technology due to its
monolithic nature
– In packet based architecture new services can
be deployed easily due to its open architecture
and separation of different layers

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Reduced Operating Cost

PSTN
Circuit-Switched
Dedicated links – Constant bit-rate

IP

Voice Packet – Switched

Compression Shared bandwidth – variable bit rate Voice Activity
Detection
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 VoIP Packet

RTP – Real Time control Protocol is used

To provide complete transport functionality
In conjunction with UDP. It provides the pay
Load type identification, Time stamping and
IP Header Sequence numbering of packets. These
20 Bytes
Functionality are not there in UDP.
UDP Header
8 Bytes RTCP- Real Time control protocol packets
RTP Header Are also exchanged periodically by end points
12 Bytes To provide the quality of feedback about the
RTP streams exchanged between them.
Voice Sample
Size of sample is dependent
on type of Coder used

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 VoIP QoS Issues
 Delay
– One way end to end delay should not increase
150 ms

 Jitter
– Jitter Buffer should not exceed 50 ms

 Packet Loss
– Shall not exceed 5% over a period of time

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 Delay
ITU’S DELAY RECOMMENDATIONS
ONE WAY DELAY DESCRIPTION
0-150 MSEC ACCEPTABLE FOR MOST USER APPLICATION (G.114)
150-400 MSEC ACCEPTABLE PROVIDED ADMINISTRATION ARE
AWARE OF THE QUALITY OF USER APPLICATION
400 + MSEC UNACCEPTABLE FOR GENERAL NETWORK
PLANNING PURPOSE

TIME (MSEC)

SATELLITE
SATELLITE QUALITY
QUALITY
HIGH
HIGH QUALITY
QUALITY FAX
FAX , BROADCAST

0 100 200 300 400 500 600 700 800

DELAY TARGET (for VOICE COMMUNICATION)

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 Packet Loss

MISSING PACKET
VOCODER ALGORITHM

PACKET LOSS IS A MAIN CULPRIT OF POOR VOICE QUALITY IN VoIP

NETWORKS
 UP TO 5% PACKET LOSS DISTRIBUTED OVER TIME IS STILL OK
 VOICE CODEC SHOULD BE TOLERANT TO OCCASIONAL PACKET
LOSS.

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Many Faces of….

IP Network

PC-to-PC
Internet Telephony

IP Network PSTN

PC-to-Phone
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 Many Faces of……

IP Network PSTN
PSTN

Phone-to-Phone
Carrier bypass

PSTN

IP Network

POTS/ IAD
IP Device
Local loop replacement

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 So Many Flavors …
 Enterprise
– LAN and intranet groupware applications
– IP PBX
– Unified messaging
– Call Centers
 Carrier
– Carrier bypass
– Local loop replacement
 Consumer
– Voice Chat
– Click-to-dial
– Internet Telephony

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 The Importance of Features

Revenue

Value-added
Services

Basic Voice
Services

Click-to-Dial Time
Video Telephony
Collaborative Web Browsing
And Many More

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 Protocols & Standards
 H.323 from ITU-T
– User to user signaling protocol
– V.1 released in 1996, currently version 4 is being used
– Widely deployed in enterprise networks
– Considered as heavy protocols due to more no of steps
and more nos of protocols inside it
 SIP (Session Initiation Protocol) from IETF
– User to user signaling protocol
– Released in 1999
– Considered more efficient due to less steps and less
nos of protocols inside it.
– Gradually penetrating in carrier class networks

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 What is H.323
 H.323 standard has been developed by the ITU-
T for equipment manufacturers and vendors
who provide voice over IP service.
 It provides technical recommendations for
voice communication over packet based
network which includes IP networks also.
 Originally developed for multimedia
conferencing on LANs, but was later extended
to Voice over IP.
 First version released in 1996. Presently version
4 is in operation

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 H.323 Architecture
Components
–Terminals
Gateway
–Gatekeepers
MCU
Internet –Gateways
Router –MCUs
Gatekeeper
PSTN

Terminal

Phone
ISDN
Terminal Video Phone
Gateway

H.323 Architecture

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 Terminals
 These are LAN endpoints that provide real-time, two-
way communications.
 All H.323 terminals are required to support H.245,
H.225, Q931, Registration Admission and status(RAS)
and Real Time Transport (RTP) protocols.
 RTP is used as a media transport protocol that carries
the voice traffic.
 H.323 terminals may also use T.120 data conferencing
protocols, video codecs.
 An H.323 terminal can communicate with either
another H.323 terminal, a H.323 gateway or a MCU

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 Terminals

H.323 H.323 H.323 H.323

Router
LAN A LAN B
H.323 H.323

 Multimedia communications services

over packet-based networks
 Real-time audio, video and/or
data communication
 Point-to-point, multipoint, or broadcast
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 Gateways
 A gateway connects two dissimilar networks
 A H.323 gateway provides connectivity between
a H.323 network and non H.323 network
 Translate protocols for call setup and release,
Converts media formats between different
networks and transfers information between
the networks connected by the gateway
 A gateway is not required however for
communication between two terminals on
H.323 network
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 Gateways

H.323 H.323 Circuit Format

Gateways H.320
(Over ISDN)

LAN A Telephone H.324

Network (Over POTS)

Speech Only
Packet Format
(Telephone)

 Appropriate translation between transmission formats

 Translation between
communication procedures
 Call setup and clearing on both sides
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 Gatekeepers
 It is an important component of the H.323
architecture and functions as its manager. It is
the central point for all calls within its zone and
provides services to registered endpoints.
 A zone is the aggregation of the gatekeeper and
the registered endpoints.
 A gatekeeper performs functions as address
translation, admissions control, call signaling,
call authorization, call management and
bandwidth management.

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Multipoint Control Units (MCU)
 The MCU acts as an endpoint on the network
for providing capability for three or more
terminals and gateways to participate in a
multipoint conference
 The MCU consists of a mandatory multipoint
controller (MC) and an optional Multipoint
Processor(MP)
 MC determine the common capabilities of the
terminals and MP is responsible for
multiplexing the media streams from different
terminals.

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 H.323 Protocol Stack

AUDIO VIDEO TERMINAL CONTROL AND MANAGMENT DATA

G.711
G.722 H.225
H.261 H.225 H.245
G.723.1
H.263 CALL T.124
G.728 CONTROL
RTCP RAS SIGNALLING
CHANNEL
G.729.a CHANNEL
RTP T.125

UNRELIABLE TRANSPORT RELIABLE TRANSPORT

PROTOCOL (UDP) PROTOCOL (TCP)

NETWORK LAYER (IP)

LINK LAYER IEEE 802.3 T.123

PHYSICAL LAYER

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 H.323 Protocol Stack
 H.225 RAS – Used for registration of endpoints
with gatekeeper and subsequently for
admission of the call (UDP)
 H.225 Call Signaling – Used for sending call
setup messages to gatekeeper/endpoints. Call
setup can be direct or gatekeeper routed (TCP)
 H.245 Control Signaling – Used for capabilities
exchange of the terminals, determination of
master and slave and then for opening and
closing of logical channels. (TCP)

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 Call Establishment in H.323
Transport Address of
T2 for Call Signaling
T-1 GK T-2
ARQ(1)
H.225
RAS
Message
ACF(2) Call Signaling
s SETUP(3)

CALL PROCEEDING(4)

ARQ(5)
RAS
Transport Address T2 Message
for Control Signaling ACF(6) s

ALERTING(7)

CONNECT(8)

H.225
Call Signaling
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 H.323 Control Signaling Flow
T-1 GK T-2
Terminal Capabilities set (9)

Terminal Capabilities set Ack (10)

Terminal Capabilities set (11)

Terminal Capabilities set Ack (12)

Open Logical Channel (13)

Open Logical Channel Ack (14)

Transport Address of T1
For RTP and RTCP
Open Logical Channel (15) streams

Transport Address of T2
For RTP and RTCP Open Logical Channel Ack (16)
streams

H.245 messages
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H.323 Media Stream and Media
Control Flow
T-1 GK T-2

RTP Media Stream (17)

RTP Media Stream (18)

RTCP Messages (19)

RTCP Messages (20)

RTP Media stream and RTCP messages

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 H.323 Call Release
T-1 GK T-2

End Session Command (21)

End Session Command (22)

Release Complete(23)

DRQ(24) DRQ(24)

DCF(25)
DCF(25)

H.225 Message

H.245 Signaling Message RAS Message

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Session Initiation Protocol (SIP)
 Session Initiation Protocol (SIP) is the IETF
standard for voice or Multimedia session
establishment over the Internet.
 SIP RFC-2543 released in Feb 1999
 SIP also works in client server architecture.
Request is initiated by client and response is
given by servers. The request message, together
with the associated response messages makes a
SIP transaction.
 SIP is a text based signaling protocol
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Session Initiation Protocol (SIP)
 SIP depends on Session Description Protocol (SDP) for
negotiation of session parameters such as codec
identification and media
 SIP supports user mobility through proxy servers and
redirecting requests to the users currently registered
location
 Major SIP Features
– Call setup
– Renegotiate call parameters
– User location
– User availability
– User capability
– Call handling

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Session Initiation Protocol (SIP)
 SIP is an application layer protocol that can
establish, modify and terminate multimedia
sessions on Internet telephony calls
 Being a text based based protocols it resembles
the HTTP and SMTP. Like these protocols SIP
is also a textual protocol based on the client
server model
 SIP addressing resembles email addressing.

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 SIP Architecture
SIP Proxy & Redirect
Servers

SIP

SIP User SIP SIP

Agents (UA)
SIP
Gateway
PSTN
SIP Phone RTP

Legacy PBX
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 SIP Components
 User Agents – A SIP User Agent is an end
system acting on behalf of the user. It consists
of two parts :
– User Agent Client (UAC) : This is the user
client portion, which is used to initiate a SIP
request to the SIP servers or the UAS
– User Agent Server (UAS) : This is the server
portion that listens and responses to SIP
requests
– User=UAC+UAS

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 SIP Components
 The SIP architecture describes the following types of
network servers to help in the SIP call setup and
services
– Registration servers – This server receives registration
requests from SIP users and updates their current
location with itself
– Proxy server – This server receives SIP requests and
forwards them to the next hop server, which has more
information of the called party.
– Redirect Server – This server on receipt of the SIP
request, determines the next hop server and returns the
address of the next hop server to the client instead of
forwarding the request to the next hop server itself.

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Call Establishment through
Proxy Server
Invite IP-Based
Network
Client Client
Invite

Server Server

User Agents Client Server User Agents

Proxy Redirect

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 Call Establishment through
Proxy Server
Invite IP-Based
Network
Client Client
Invite

Server Server

User Agents Server Client User Agents

Proxy Redirect

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Call Establishment through
Proxy Server
Ack IP-Based
Network
Client RTP Client
Ack

Server Server

User Agents Client Server User Agents

Proxy Redirect

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 Call Release Through a
Proxy Server
Bye IP-Based
Network
Client RTP Client
Bye

Server Server

User Agents Server Client User Agents

Proxy Redirect

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 Call Release Through a
Proxy Server
Ack IP-Based
Network
Client Client
Ack

Server Server

User Agents Client Server User Agents

Proxy Redirect

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