IP Telephony (VoIP)

CSI4118
Fall 2005

Introduction (1)
A recent application of Internet technology Voice over
IP (VoIP): Transmission of voice over Internet
How VoIP works

Continuously sample audio

Convert each sample to digital form
Send digitized stream across Internet in packets
Convert the stream back to analog for playback

Why VoIP
IP telephony is economic; High costs for traditional telephone
switching equipments.
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Introduction (2)
Challenge
Voice transmission delay
Call setup: call establishment, call termination, etc.
Backward compatibility with existing PSTN (Public
Switched Telephone Network)

IP Telephony Standards:
ITU (International Telecommunication Union) controls
telephony standards.
IETF (Internet Engineering Task Force) controls
TCP/IP standards.
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Encoding, Transmission, &

Playback (1)
Both groups agree on the basics for encoding
and transmission of audio:
Audio is encoded using a well-known standard such
as Pulse Code Modulation (PCM).
Audio is transferred using the Real-time Transport
Protocol (RTP).
RTP message is encapsulated in a UDP datagram
that is further encapsulated in an IP datagram for
transmission.
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Encoding, Transmission, &

Playback (2)
UDP is used for transport because
lower overhead: audio must be played as it arrives.
Playback cannot be stopped to wait for a
retransmitted packet.

Two independent RTP sessions exist, because

an IP phone call involves transfer in two
directions
IP phone acts as sender for outgoing data, and
IP phone acts as receiver for incoming data.
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Signaling Systems & Protocols

Main complexity of VoIP: Call setup and call
management.
The process of establishing and terminating a call is
called Signaling.
In traditional telephone system, signaling protocol is SS7
(signaling System 7).
In VoIP, signaling protocols are:
SIP (Session Initiation Protocol), by IETF
H.323, by ITU
Megaco & MGCP, jointly by IETF and ITU.

VoIP signaling protocols should be able to interact with SS7.

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A Basic IP Telephone System

The simplest IP telephone system uses two basic components:

- IP telephone: end device allowing humans to place and receive

calls.
- Media Gateway Controller: providing overall control and coordination
between IP phones; allowing a caller to locate a callee (e.g. call
forwarding)
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Interconnection with Others (1)

IP telephone system needs to interoperate
with PSTN or another IP telephone
system.
Two additional components needed for
such interconnection:
Media Gateway
Signaling Gateway
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Interconnection with Others (2)

Media gateway: translates audio between IP

network and PSTN.
Signaling Gateway: translates signaling
operations.

Signaling Protocols
Two major protocols: H.323, SIP
H.323, invented by ITU, defines four elements
that comprising a signaling system:
Terminal: IP phone
Gatekeeper: provides location and signaling
functions; coordinates operation of Gateway.
Gateway: used to interconnect IP telephone system
with PSTN, handling both signaling and media
translation.
Multipoint Control Unit: provides services such as
multipoint conferencing.
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Signaling Protocols
SIP: Session Initiation Protocol. Invented by
IETF.
SIP defines three main elements that comprise a
signaling system:
User Agent: IP phone or applications
Location servers: stores information about users
location or IP address
Support servers:
Proxy Server: forwards requests from user agents to another
location.
Redirect Server: provides an alternate called partys location
for the user agent to contact.
Registrar Server: receives users registration requests and
updates the database that location server consults.
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H.323 Characteristics
H.323 consists of a set of protocols that work
together to handle all aspects of communication,
including:
Transmission of a digital audio phone call
Signaling to set up and manage phone call
Allows transmission of video and data while a phone
call is in progress
Sends binary message
Incorporates protocols for security
Uses a special hardware Multipoint Control Unit for
conferencing calls
Defines servers for address resolution, authentication,
accounting, features, etc.

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H.323 Layering
H.323 uses both UDP and TCP over IP.
Audio travels over UDP
Data travels over TCP

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SIP Characteristics
Operates at the application layer.
Encompasses all aspects of signaling, e.g. location of
called party, ringing a phone, accepting a call, and
terminating a call.
Provides services such as call forwarding.
Relies on multicast for conference calls.
Allows two sides to negotiate capabilities and choose the
media and parameters to be used.
SIP URI is similar to email address. (with prefix sip:)
E.g.
sip:bob@somewhere.com

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SIP Methods
Six basic message types, known as
methods:

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An Example SIP Session

User agent A contacts DNS

server to map domain name in
SIP request to IP address.

User agent A sends a INVITE

message to proxy server that
uses location server to find the
location of user agent B.

Call is established between A

and B. Then media session
begins.

Finally, B terminates the call

by sending a BYE request.
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Telephone Number Mapping &

Routing (1)
How should users be named?
PSTN follows ITU standard E.164 for phone numbers. E.g. 1613-123-4567
SIP uses IP addresses. E.g. sip:smith@uottawa.ca

In an integrated network (PSTN + IP), two problems

defined:
Locate a user
Find a efficient route to the user

IETF proposed two protocols:

ENUM: E.164 NUMbers
TRIP: Telephone Routing over IP
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Telephone Number Mapping &

Routing (2)
ENUM
Converting E.164 phone number into a Uniform
Resource Identifier (URI)
Using Domain Name System to store mapping
A phone number is converted into a special domain
name: e164.arpa
E.g. 1-800-555-1234 4.3.2.1.5.5.5.0.0.8.e164.arpa
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Telephone Number Mapping &

Routing (3)
TRIP
Finding a user in an integrated network
Used by location server or other NEs to advertise
routes
Independent of signaling protocols
Dividing the world into a set of IP Telephone
Administrative Domains (ITADs)
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IP Telephones and Electrical

Power
Analog telephone system continues to work when
electrical power are unavailable
The wires that connect a telephone to the central office supply
the power

Currently, IP telephones have to depend on an external

source of power
IP phones must have both network connection and power
connection.
Several mechanism proposed to integrate power with network
connections.
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Summary (1)

IP telephony or VoIP refers to the transmission of voice telephone

calls over IP networks.

Hot area both in research and market because of low cost

Challenge in backward compatibility with PSTN

The complexity of IP telephony is on signaling. Both ITU and IETF

propose signaling standards.
H.323, by IUT
SIP, by IETF, offering similar functions to H.323, but simpler than H.323.
Both are competing to be recognized as #1 signaling protocol

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Summary (2)
H.323 uses a set of protocols for call setup and
management
SIP uses a set of servers to handle various aspects of
signaling
ENUM maps an E.164 telephone number into a URI
(usually SIP URI)
TRIP provides routing among IP telephone
administrative domains
IP telephones depends on external power, while analog
phones dont.
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