VoIP

Voice Over IP

Why An Organization Would Use IPT

IP Telephony
Instead of using traditional circuit switch systems for voice
communications, IP Telephony uses a 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 packetswitched IP- based 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
Voice-over-IP (VoIP)
Specific sub-set of IP Telephony in which the principal
transmission network(s) is (are) private, managed IP-based
network(s).
Voice-over-frame relay ; Voice-over-cable ; Voive-overDSL (VoDSL)

PC-to-PC
Internet

ISP

ISP

PSTN

PSTN

USER A

Server
Modem

USER B
Modem

PC-to-Phone
IP Telephony
Provider

Internet

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)

RSVP

QoS: Delays
Network Delay

Sender Delay:

Coding delay
Packeting delay
Transmission delay

IP Network

Receiver Delay:

Sender

Network

DePacketing delay
Receiver delay

Inversion

Loss Packet

T"#T#T

Decoding delay

Delay Variation :
T#T Jitter
Receiver

Delay

QoS Technologies
Reservation
Allocates resources on a per-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
DiffServ and MPLS.

L2 marking

Understanding L3 Marking

DSCP

IP Telephony Protocols
SIP, H.323 and MGCP

Call Control and Signaling

H.323

Signaling and
Gateway Control

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.

Packet Encapsulation
RTP datagram
Version,
flags & CC

Payload
Type

Sequence
Number

Synchronization
Source ID

Timestamp

CSRC ID
(if any)

Codec Data

0-60

0-1460

UDP datagram
Source
Port Number

Destination
Port Number

Version &
header length

UDP length
2

Total
Length

Packet
ID

Ethernet Frame
Inter-frame
gap

Preamble

12

Flags &
TTL
Frag Offset

Header
Checksum

Source
Address

Destination
Address

Start of frame
delimiter

0-1472

Options
(if any)

Data

0-40

0-1480

Length or
Ethertype

Destination
Address
1

Data

Protocol

IP packet

TOS

UDP checksum

Source
Address
6

Data
2

0-1500

Pad

Checksum
0-46

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 components
UAC: user-agent client (caller application)
UAS: user-agent server: accept, redirect, refuse
call
redirect server: redirect requests
proxy server: server + client
registrar: track user locations
user agent = UAC + UAS
often combine registrar + (proxy or redirect
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 a final response to an INVITE
request.
BYE - Indicates termination of the call.
CANCEL - Cancels a pending request.
REGISTER Registers the user agent.
OPTIONS Used to query the
capabilities of a 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.

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.

SIP: Registering
Each time a user turns on the SIP
user client (SIP IP Phone, PC, or
other SIP device), the client
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.
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.

Simplified SIP Call Setup

Proxy Server

User Agent
INVITE

Location/Redirect Server
INVITE

User Agent

Proxy Server

302
(Moved Temporarily)
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)

SIP Call Signaling

Assumes Endpoints(Clients)
know each others IP addresses
SIP
Endpoint
Signaling
Plane

SIP
Gateway
Invite
180 Ringing
200 OK

SIP + SDP
(TCP or UDP)

Ack

Bearer
Plane

RTP Stream
RTP Stream
RTCP Stream

Media (UDP)

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,
video and/or data communications.
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.

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
Terminal B
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 call setup where call signaling is
The two terminals establish RTP media
not routed to the gatekeeper. Refer to the H.323
recommendation for more call setup scenarios.
paths.

H.323 Call Signaling

Assumes Endpoints(Clients)
know each others IP addresses
H.323
Endpoint

Setup
Alerting

H.323
H.225 (TCP) Gateway
(Q.931)

Connect

Terminal Capability Set

Signaling
Plane

Terminal Capability Set & Acknowledge

Terminal Capability Set Acknowledge
Open Logical Channel

H.245 (TCP)

Open Logical Channel & Acknowledge

Open Logical Channel Acknowledge

Bearer
Plane

RTP Stream
RTP Stream
RTCP Stream

H.323v1 (5/96) - 7 or 8 Round Trips

H.323v2 Fast Start (2/98) - 2 Round Trips

Media (UDP)

stateless
Having no information about what occurred previously. Most modern applications maintain
state, which means that they remember what you were doing last time you ran the
application, and they remember all your configuration settings. This is extremely useful
because it means you can mold the application to your working habits.
The World Wide Web, on the other hand, is intrinsically stateless because each request for a
new Web page is processed without any knowledge of previous pages requested. This is one
of the chief drawbacks to the HTTP protocol. Because maintaining state is extremely useful,
programmers have developed a number of techniques to add state to the World Wide Web.
These include server APIs, such as NSAPI and ISAPI, and the use of cookies.
Having no information about what occurred previously. Most modern applications maintain
state, which means that they remember what you were doing last time you ran the
application, and they remember all your configuration settings. This is extremely useful
because it means you can mold the application to your working habits.
The World Wide Web, on the other hand, is intrinsically stateless because each request for a
new Web page is processed without any knowledge of previous pages requested. This is one
of the chief drawbacks to the HTTP protocol. Because maintaining state is extremely useful,
programmers have developed a number of techniques to add state to the World Wide Web.
These include server APIs, such as NSAPI and ISAPI, and the use of cookies.
stateful
Having the capability to maintain state. Most common applications are inherently stateful.

How Your Voice Becomes A Packet

How to Turn Voice Into Bits

The Nyquist Theorem

Quantization Process

Quantization Techniques

CODECs

MOS

DSPs