VoIP

Voice Over IP

IP Telephony

Instead of using traditional circuit switch systems for

voice communications, IP Telephony
p
y uses a p
packet
protocol originally designed for data communications.

Circuit Switched - PSTN

Packet Switched Data Network

Definition

IP Telephony
Transmission of voice, fax, and related services over
packet-switched IP- based networks.
networks
Internet Telephony
Specific sub-set of IP Telephony in which the principal
transmission network is the public Internet.
Voice-over-the-Net(VoN) ; Internet Phone ; Net
Telephony
p
y
Voice-over-IP (VoIP)
Specific sub-set of IP Telephony in which the principal
transmission network(s) is (are) private,
private managed IPIP
based network(s).
Voice-over-frame relay ; Voice-over-cable ; Voiveover-DSL
DSL (V
(VoDSL)
DSL)

Models of IP Telephony

Three mains models to use IP Telephony

PC-to-PC
PC
to PC over IP
IP.
PC-to-Phone over IP.
Phone-to-Phone over IP
IP.

PC-to-PC
I t
Internet
t

ISP

ISP

PSTN

PSTN

USER A

Server
Modem

USER B
M d
Modem

PC-to-Phone
IP Telephony
Provider

I t
Internet
t

ISP

IPTP

Gateway
PSTN

PSTN
USER A

USER B
Modem

USER B

Phone-to-Phone (1)
Management IP Network

Gateway

Gateway

Network of IP Telephony
Service Provider

PSTN
USER AUSER B
USER A

PSTN
USER B

USER B

Phone-to-Phone (2)
ISP

Internet

PSTN

ISP

PSTN

Server
USER A

USER A

USER B

IP Telephony: QoS
Packet loss (%)
10

Unacceptable for Voice

or Fax

ITU G.114
Utility Recommendation

Possibly Tolerable for

Voice

Operational Target for

Voice and Fax

0
100

200

300

400

500

Delay (ms)

QoS: Delays
Network Delay

Sender Delay:

Coding delay
Packeting delay
Transmission delay

IP Ne
etwork

Receiver Delay:

Network

DePacketing delay

Inversion

Receiver delay

Loss Packet
T"#T#T

Sender

Decoding delay

Delay Variation :
T#T Jitter
Receiver

QoS Technologies
Reservation
Allocates resources on a per
per-flow
flow basis
Flows include information such as transport protocol,
source address & port, destination address and port
IntServ/RSVP

Prioritization
Traffic flows are aggregated and categorized by "class of
service
service
DiffServ and MPLS.

DiffServ: Differentiated Services

RFCs 2474, 2475
Creates
C t classes
l
off service
i for
f traffic
t ffi fl
flows with
ith
different priorities:
Aggregates
A
t llarge numbers
b
off individual
i di id l flows
fl
att
the edge of the network into small numbers of
aggregated flows through the core of the
network.
Flows are marked at network edge in the IPv4
ToS field (DS field).
Services applied through the core.

IP Telephony Protocols
SIP, H.323 and MGCP
Call Control and Signaling
H.323

Signaling
Si
li
and
d
Gateway Control

M di
Media
Audio/
Video

H.225
H.245

Q.931

RAS

SIP

MGCP

TCP

RTP

RTCP

RTSP

UDP
IP

H.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP.
H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP.
SIP supports TCP and UDP.

SIP: Session Initiation Protocol

Session Initiation Protocol - An

application layer signaling protocol that
defines initiation, modification and
termination of interactive,, multimedia
communication sessions between users.
IETF RFC 2543 Session Initiation Protocol

SIP Distributed Architecture

SIP Components

Location
Server

Redirect
Server

Registrar
Server

PSTN
User Agent

Gateway
Proxy
Server

Proxy
Server

SIP Messages
SIP components communicate by exchanging SIP messages:
SIP Methods:
INVITE Initiates a call by inviting
user to participate in session.
ACK - Confirms that the client has
received
i d a final
fi l response to
t an INVITE
request.
BYE - Indicates termination of the call.
CANCEL - Cancels a pending request.
request
REGISTER Registers the user agent.
OPTIONS Used to query the
capabilities of a server.
server
INFO Used to carry out-of-bound
information, such as DTMF digits.

SIP Headers
SIP borrows much of the syntax and semantics from
HTTP.
A SIP messages looks like an HTTP message message
formatting, header and MIME support
The SIP address is identified by a SIP URL, in the
format: user@host.

SIP: Communication Establishment

Establishing communication using SIP usually occurs in
six steps:
1.
2.

3.

4.
5.
6
6.

Registering, initiating and locating the user.

Determine the media to use involves delivering a description
of the session that the user is invited to.
Determine the willingness of the called party to communicate
the called party must send a response message to indicate
willingness to communicate accept or reject.
Call setup.
Call modification or handling (eg call transfer (optional)).
Call termination
termination.

SIP: Registering
Each time a user turns on the SIP
user client (SIP IP Phone, PC, or
other
th SIP device),
d i ) the
th client
li t
registers with the proxy/registration
server.
Registration can also occur when
the SIP user client needs to inform
the proxy/registration server of its
location.
The registration information is
periodically refreshed and each user
client must re-register with the
proxy/registration server
server.
Typically the proxy/registration
server will forward this information
to be saved in the location/redirect
/
server.

Proxy/
Location/
Registration
Redirect
Server
Server
REGISTER
REGISTER

SIP Phone
User

200

200

SIP Messages:
REGISTER Registers the address
listed in the To header field.
200 OK.
OK

Simplified SIP Call Setup

Proxy Server

Use Agent
User
INVITE

Location/Redirect Server
INVITE
302
(Moved Temporarily)

User Agent

Proxy Server

ACK
INVITE
INVITE
302
(Moved Temporarily)
ACK

Call
Setup

180 (Ringing)
200 (OK)
ACK

Media
Path
Call
Termination

180 (Ringing)
200 (OK)
ACK

INVITE
180 (Ringing)
200 (OK)
ACK

RTP MEDIA PATH

BYE

BYE

BYE

200 ((OK))

200 ((OK))

200 ((OK))

IP Telephony Signaling Protocols:

H.323

Describes terminals and other entities that

provide multimedia communications services
over Packet Based Networks (PBN) which
may not provide a guaranteed Quality of
Service.
H.323 entities may provide real-time audio,
/ data communications.
video and/or
ITU-T Recommendation H.323 Version 4

H.323 Components

Gatekeeper

Multipoint
Control Unit

Circuit
Switched
Networks

Packet Based
Network

Terminal

Gateway

H.323 : Communication Establishment

Establishing communication using H.323 may
occur in five steps:
1. Call setup.
2. Initial communication and capabilities
exchange.
exchange
3. Audio/video communication establishment.
4. Call services.
5. Call termination.

Simplified H.323 Call Setup

Both endpoints have previously registered
with the gatekeeper.
Terminal A initiate the call to the
T
Terminal
i lB
Terminal A
gatekeeper. (RAS messages are
Gatekeeper
exchanged).
1. ARQ
2. ACF
The gatekeeper provides information for
Terminal A to contact Terminal B.
3. SETUP
4. Call Proceeding
Terminal A sends a SETUP message to
5. ARQ
Terminal B.
6. ACF
Terminal B responds with a Call Proceeding
7.Alerting
message and also contacts the gatekeeper
8.Connect
for permission.
H.245 Messages
Terminal B sends a Alerting and Connect
RTP Media Path
message.
Terminal B and A exchange H.245
RAS messages
messages to determine master slave,
Call Signaling Messages
terminal capabilities, and open logical
Note: This diagram only illustrates a simple
channels.
point-to-point
i tt
i t call
ll setup
t
where
h
call
ll signaling
i
li
is
i
The two terminals establish RTP media
not routed to the gatekeeper. Refer to the H.323
recommendation for more call setup scenarios.
paths.