Session No.

VoIP
H.323
A PSTN Network

PSTN Network A PBX

Class 5
Class 4 Switch
Switch
Class 4
Switch
Intern
PSTN Network B
ational
GW
Class 5 Intern Class 4 Class 5
Switch ational Switch Switch
GW
Signaling
User to Network Network to Network to User
Signalling
Network Signalling Signalling

User
Network Node User
Network Node
Access Signalling Network Signalling Access Signalling

Two modes of signaling

CAS Channel Associated Signaling
CCS Common Channel Signaling (e.g. SS7)

Signalling protocols
Access Signalling: Digital Subscriber System 1 (DSS1) PRI, QSIG , CAS
Network Signalling: Signalling system 7 ISUP, TUP ,CAS
Signaling in PSTN SS7
 SS7 signaling
SSP
SCP

VOICE Routes voice

Provides calltraffic between
control external to the
SSPsLocally connected subscribers/devices

Coordinates Other switches
actions of multiple
Controlling devices
elements as needed by service
One Multiple
or moredevice types (e.g.,
applications 10 party lines,
generally
ISDN lines)
associated with each SCP
S S Alerting
STP
Displays
SS7
T T Feature processing
Packet router, used only in
Bill generation quasi associated
P P IP
signaling mode
SSP SSP Provides STP routes
a means ofmessages based
interacting with the point code
on Absolute
S S subscriber references
via voice or Global
T T Media Dialogs title address references
Proprietary signaling
P P

IP
SCP
Circuit Switching Vs packet switching
Voice & Data characteristics

Voice calls
Delay sensitive
Long Hold time
Narrow bandwidth requirement
Data Calls
Delay Insensitive
Short Hold time
Wide bandwidth utilization.
Difference between the Networks
Packet Network Telecom Network

Two-party communication model Three-party communication model

(client/server or peer-to-peer) (caller-network-callee)

No direct control by network over the Communication controlled (managed)

communication except access rights by the network

Communicating users agree on which Communication protocols are network

communication protocol to use specific and transparent to users

User-defined QoS constraints provided QoS constraints of a communication

to the network known and guaranteed by the network

Communication can be universal but no Universality of communication service

network interconnect mandatory at through interconnect agreements
service level between sub-network operators
Charging based on flat rate or volume Charging based on usage of the
of transported data communicating service
Converged Networks
What is VoIP

VoIP stands for Voice over Internet Protocol. It is often referred

to as Internet Telephony.

Transmission of digitized voice in packet network (IP)

Enables telephone conversation to be carried over IP network (in

part or end-to-end)

Optimized for data communication

Enables telephony providers to provide cheaper service

What is VOIP ?

Voice is digitized, packetized and transmitted over IP network

instead of PSTN
The difference is concealed mainly in transmitting network and
transmitting format
PSTN Vs VoIP
PSTN Internet Telephony

Underlying Technology TDM circuit Packet switching

switching

QoS Guarantees Yes No

Network resource reserved Yes No

at call setup

Network Elements Class 4, Class 5 Gateways, switches, routers

switching systems

Call Processing Intelligence Mostly integrated in In separate telephony servers

switching system

Bandwidth per call 64 kbps Variable

Signaling CAS, ISDN,SS7 SIP, H.323

Transport TDM ATM, FR, native IP in access, ATM, native IP in core

How reliability acheieved Redundancy within Redundant routes through network

each network
element
Voice Communication Requirements

Telephone quality -- Very few noticeable errors and low delay and
no variation in delay
Packet transmission -- has a larger delay which is extremely
important for voice
Jitter -- the variable delay is important for voice
Small amount of packet lost is tolerable But what is the amount of
tolerance?
 Transport Layer
Provide end-to-end communication services for applications
Two primary transport layer protocols: Transmission Control
Protocol (TCP) [RFC793] and User Datagram Protocol (UDP)
[RFC768]

Source port Destination port

Length Checksum

Data

UDP Packet

TCP Packet
TCP Or UDP ?
Service TCP UDP

Connection setup Takes time, but TCP does this to ensure No connection required.
reliability.

Guaranteed message delivery Returns ACKs (acknowledgments). Since UDP does not return ACKs, the
receiver cannot signal that packets have
been successfully delivered. Lost packets
are not retransmitted.

Packet sequencing (provide Sequentially numbers packets. UDP does not insert sequence numbers.
information about the correct The packets are expected to arrive as a
order of packets) continuous stream or they are dropped.

Flow controls The receiver can signal the sender to ACKs, which are used in TCP to control
slow down. packet flow, are not returned.

Congestion controls Network devices can take advantage of Without ACKs, the network cannot signal
TCP ACKs to control the behavior of congestion to the sender.
senders.
VoIP - History

1995: Vocaltec, Inc. 1998 Jan: 1998 Oct: 2000 Nov: 2003:
Internet Phone v1.0 H323 v2 MGCP v1 H323 v4 H323 v5

1996: 1999: 2000 Nov:

- H323 v1 MEGACO/H.248 2002 June: 2004 July:
SIP RFC 2543
- SIP Draft v1 SIP RFC 3261 TGCP I08
VoIP Network Architecture
Application layer
Media App
Server Server App Server
Media Server
App Layer

Parlay, JAIN

Media
SS MG RTP MG Call Control Layer
P MGCP,
Megaco PBX
MGCP, Phones
Softswitch Megaco
MediaPSTN
Gateway
Enterprise
STP SS7 SG SIGTRAN
SIP T, SignalingNetwork
Gateway
BICC Softswitch Softswitch
PSTN Carrier
Network
SIP, MGCP
Call Control Layer
VoIP Enterprise
Network
Transport Layer
Transport Layer Routers, Repeaters
IP Carrier Network etc
VoIP Network - Phones
Media App Phones (End User Terminal)
Server Server
Intelligent or dumb
App Layer Capable of hosting applications
Parlay, JAIN
SIP, MGCP phones available

Media
SS MG RTP MG
P MGCP,
Megaco PBX
MGCP,
Softswitch Megaco
PSTN Enterprise
SG SIGTRAN Network
STP SS7 SIP T,
BICC Softswitch
PSTN Carrier
Network
SIP, MGCP
Call Control Layer
VoIP Enterprise
Network

Transport Layer
IP Carrier Network
VoIP Network Media Gateway
Media Gateway
Media App
Server Server Terminates different types of
media interfaces (TDM, IP, ATM)
App Layer
Converts one media format to
Parlay, JAIN another format e.g, G.711 to
G723

SS
Media
RTP
Controlled by Softswitch via
MG MG
gateway control protocols such as
P MGCP,
Megaco
MGCP, Megaco PBX
MGCP,
Softswitch Megaco
PSTN Enterprise
SG SIGTRAN Network
STP SS7 SIP -T
Softswitch
PSTN Carrier
Network
SIP, MGCP
Call Control Layer
VoIP Enterprise
Network

Transport Layer
IP Carrier Network
VoIP Network Signaling Gateway
Media App Signaling Gateway
Server Server Terminates different types of
signaling interfaces
App Layer
Transparently communicates
Parlay, JAIN
signaling from one interface type
to another (SIGTRAN)
Media
SS MG RTP Might
MG convert signaling in one
P MGCP,
Megaco format to another PBX
format
MGCP,
Softswitch Megaco
PSTN Enterprise
SG SIGTRAN Network
STP SS7 SIP -T
Softswitch
PSTN Carrier
Network
SIP, MGCP
Call Control Layer
VoIP Enterprise
Network

Transport Layer
IP Carrier Network
VoIP Network - Softswitch
Softswitch/MGC/Call Agent
Media App
Server Server Controls Media Gateways &
Signaling Gateways
App Layer

Parlay, JAIN Routes VoIP sessions/calls to

other softswitches, phones etc
Media
SS RTP
P
MG
MGCP,
MG
Completely software based
Megaco PBX
MGCP,
Softswitch Megaco
PSTN Enterprise
SG SIGTRAN Network
STP SS7 SIP -T
Softswitch
PSTN Carrier
Network
SIP, MGCP
Call Control Layer
VoIP Enterprise
Network

Transport Layer
IP Carrier Network
VoIP Network Application Server
Media App
Server Server

App Layer

Parlay, JAIN

Media
SS MG RTP MG
P MGCP,
Megaco PBX
MGCP,
Softswitch Megaco
PSTN Enterprise
SG SIGTRAN Network
STP SS7 SIP -T
Softswitch
Application Server
PSTN Carrier
Network Service Logic Execution

Call Control Layer

SIP,
Hosts
MGCP applications
VoIP Enterprise
Caters to multiple softswitches
Network
Can be hosted even outside the
carrier network
Transport Layer
IP Carrier Network SIP/Parlay/JAIN/CPL/proprietary
VoIP Network Media Server
Media App
Server Server

App Layer

Parlay, JAIN

Media
SS MG RTP MG
P MGCP,
Megaco PBX
MGCP,
Softswitch Megaco
PSTN Enterprise
SG SIGTRAN Network
STP SS7 SIP -T
Softswitch
PSTN Carrier
Network
SIP, MGCP
Call Control Layer
Media Server VoIP Enterprise
Network
Provides media capabilities
needed for applications
Transport Layer Announcements, Voice Mail, IVR,
IP Carrier Network
conference capabilities
VoIP Signaling
Course Overview

What is H.323 ?
H.323 entities
Protocols in H.323
Important H.323 messages
SIP vs.H.323
What is H.323 ?

A technology for the transmission of real-time audio, video and data over packet-based networks

Packet-based networks include;

IP-based Networks: the Internet
IPX-based Networks: LANs
Enterprise Networks
Metropolitan Area Networks
Wide Area Networks
What is H.323 ?

Can be applied in a wide variety of mechanisms such as:

Audio only(IP telephony)
Audio & Video(Video Telephony)
Audio & Data
Audio, Video & Data
H.323 versions

Version Date Reference for key feature summary

H.323 Version 1 May 1996 New release. Refer to the specification.

http://www.packetizer.com/iptel/h323/

H.323 Version 2 January 1998 http://www.packetizer.

com/iptel/h323/whatsnew_v2.html

H.323 Version 3 September http://www.packetizer.

1999 com/iptel/h323/whatsnew_v3.html

H.323 Version 4 November http://www.packetizer.

2000 com/iptel/h323/whatsnew_v4.html
H.323 The primary goal

Interoperability with other multimedia-services networks

Achieved through use of a gateway

Gateway performs signaling translation required for interoperability

H.323 Components
Entities Protocols
- Terminals - Parts of H.225.0 RAS, Q.931
- Gateways - H.245
- Gatekeepers - RTP/RTCP
- MCUs - Audio/video codecs
H.323 Pictorial Overview

Call Control and Signaling Signaling and Media

Gateway Control Audio/
Video

H. Q. RT RTC RTS
RAS SIP MGCP
245 931 P P P

UDP

IP
TCP
H.323 Architecture

H.323 Network Architecture and Components

H.323 Entities:Terminals
Used for real-time,bi-directional multimedia communications
Can either be a PC or stand-alone device running the H.323 stack and
multimedia applications
Support audio communications and can optionally support video or data
are compatible with H.320,H.321,H.322 and H.324 terminals
Must support:

Voice - audio codecs

Signaling and setup - Q.931, H.245, RAS
H.323 Entities: Terminals (cont.)
Comparison of audio codes
H.323 Entities: Gateways

Connect and provide communication between an H.323 and non-H.323

network
Connectivity is achieved by:

Translating protocols for call-setup and release

Converting media formats between different networks
Transferring information between the two networks
Is not required for communication between 2 terminals in the same H.323
network
H.323 Entities: Gateways (cont.)
H.323 Entities: Gatekeepers

An entity considered as the brain of the H.323 network -is the focal
point for all calls within the network

Typically a software application, implemented on a PC,but can be

integrated in a gateway or terminal

Usually one gatekeeper per zone; alternate gatekeeper might

exist for backup and load balancing
 H.323 Entities:Gatekeepers(contd.)

Addressing resolution
Admission control
Bandwidth control
Accounting and Billing
Managing a zone (a collection of H.323 devices)
H.323 Entities: MCUs

Endpoints that support conferences between 3 or more endpoints

Manage conference resources, determine which codec to use and

handle the media stream

Gatekeepers, gateways and MCUs are logically separate components

but can be combined as a single physical device
H.323 Zone
A collection of terminals, gateways and MCUs managed by a single
gatekeeper

Includes at least one terminal and may include gateways and MCUs

Is independent of network topology and may be comprised of multiple

network segments connected using routers
H.323 zone
H.323 Protocol Stack
Audio codecs (G.711, G.723.1, G.728, etc.) and video codecs (H.261, H.263)
compress and decompress media streams
Media streams transported on RTP/RTCP
RTP carries actual media
RTCP carries status and control information
Signaling is transported reliably over TCP
RAS - registration, admission, status
H.225 - call setup and termination carried unreliably
H.245 - capabilities exchange on UDP
H.323 Protocol Stack
H.323 in relation with OSI model
Typical H.323 Network Deployment
H.323 Terminal Characteristics

H.323 terminals must support:

RAS for registration, admission and status control with
a gatekeeper
H.225 for call-signaling and call-setup
H.245 for exchanging terminal capabilities and creation
of media channels
RTP/RTCP for sequencing and carrying media packets
G.711 audio codec & H.261 video codec(optional)
H.225 RAS

This is the protocol between endpoints and gatekeepers

Used to perform registration, admission control, bandwidth changes, status

exchange and disengage procedures between endpoints and gatekeepers

A RAS channel is provided for exchanging RAS messages - is opened prior to

establishment of any other channels
H.225 Call Signaling

Used to establish and terminate a connection between two H.323 endpoints

by exchanging H.225 protocol messages on the call-signaling channel

Call-signaling channel is opened between two endpoints(direct-call

signaling) or between an endpoint and the gatekeeper,if one exists
(gatekeeper-routed call signaling)
H.245 Control Signaling

Used to exchange end-end control messages governing operation of the end-

points

These messages carry information related to:

Capabilities exchange
Flow-control messages
General commands and indications
Opening and closing of channels used to carry media
streams
Important H.323 messages: RAS

Message Function

Registration Request(RRQ) Request from terminal or gateway to register with a

gatekeeper. Gatekeeper either confirms(RCF) or rejects
(RRJ)
Admission Request(ARQ) Request for access to packet network from terminal to
gatekeeper. Gatekeeper either confirms(ACF) or rejects
(ARJ)
Bandwidth Request(BRQ) Request for changed bandwidth allocation, from
terminal to gatekeeper.Gatekeeper either confirms(BCF)
or rejects(BRJ)
Disengage Request(DRQ) Sent from endpoint to gatekeeper.Informs gatekeeper
that endpoint is being dropped. Gatekeeper either
confirms(DCF) or rejects(DRJ).If sent from gatekeeper
to endpoint, DRQ forces call to be dropped. Endpoint
must respond with DCF
Information Request(IRQ) Request for status information from gatekeeper to
terminal
Important H.323 messages:RAS

Message Function
Info Request Response(IRR) Response to IRQ. May be sent unsolicited by terminal to
gatekeeper at predetermined intervals
RAS timers and Request in Recommended default time values for response to RAS
progress(RIP) messages and subsequent retry counts if response is not
received
 Important H.323 messages: H.225

Message Function

Alerting Called user has been alerted - Phone is ringing. Sent

by called user
Call proceeding Requested call establishment has been initiated. Sent by
called user
Connect Acceptance of call by called entity.Sent from called
entity to calling entity
Setup Indicates desire of calling entity to setup a connection to
the called entity
Release complete Indicates release of call if H.225.0(Q.931)call signaling
channel is open.Sent by H.323 terminal
 Important H.323 messages: H.225

Message Function
Status Responds to an unknown call signaling message or
Status enquiry message.Provides call state information
Status enquiry Requests call status.Sent by gatekeeper or endpoint to
another endpoint
 Important H.323 messages: H.245

Message Function

Master Slave Determination Determines which terminal is master and which is slave.
Possible replies:Acknowledge, Reject, Release

Terminal capability set Contains information about a terminals capability to transmit

and receive multimedia streams. Possible replies:
Acknowledge, Reject, Release
Open logical channel Acceptance of call by called entity.Sent from called entity to
calling entity. Possible replies:Acknowledge, Reject, Confirm

Close logical channel Open a logical channel for transfer of A/V and data
information. Possible replies:Acknowledge

Request mode Used by receive terminal to request particular modes of

transmission from a transmit terminal.Mode types:Audio
mode, Video mode, Data mode, Encryption mode
 Important H.323 messages: H.245

Message Function

Send terminal capability set Commands far-end terminal to send its transmit/receive
capabilities
End session command Indicates end of H.245 session.Terminal will not send
any more H.245 messages
A high-level communication exchange between two endpoints (EP)
and two gatekeepers (GK)
SIP vs. H.323

Information SIP H.323

Standards IETF. ITU.
Body
Relationship Peer-to-Peer. Peer-to-Peer.
Origins Internet based and web Telephony based. Borrows call
centric. Borrows syntax and signaling protocol from ISDN
messages from HTTP. Q.SIG.
Client Intelligent user agents. Intelligent H.323 terminals.
Core servers SIP proxy, redirect, location, H.323 Gatekeeper.
and registration servers.
Current Interoperability testing Widespread.
Deployment between various vendors
products is ongoing at SIP
bakeoffs.
SIP is sponsors
Interoperability IMTC gaining interest.
interoperability events among SIP, H.323, and
MGCP. For more information, visit: http://www.imtc.org/
 SIP vs. H.323

Information SIP H.323

Capabilities SIP uses SDP protocol for Supported by H.245 protocol. H.

Exchange capabilities exchange. SIP does 245 provides structure for
not provide as extensive detailed and precise information
capabilities exchange as H.323. on terminal capabilities.

Control Text based UTF-8 encoding. Binary ASN.1 PER encoding.

Channel
Encoding Type
Server Stateless or stateful. Version 1 or 2 Stateful.
Processing Version 3 or 4 Stateless or
stateful.
Quality of SIP relies on other protocols such Bandwidth management/control
Service as RSVP, COPS, OSP to and admission control is
implement or enforce quality of managed by the H.323
service. gatekeeper.
The H323 specification
recommends using RSVP for
resource reservation.
SIP vs. H.323
Information SIP
Security Registration - User agent
registers with a proxy server.
Authentication - User agent
authentication uses HTTP
digest or basic authentication.
Encryption - The SIP RFC
defines three methods of
encryption for data privacy.
Endpoint Uses SIP URL for addressing.
Location and Redirect or location servers
Call Routing provide routing information.
H.323
Registration - If a gatekeeper is
present, endpoints register and
request admission with the
gatekeeper.
Authentication and Encryption -
H.235 provides recommendations
for authentication and encryption
in H.323 systems.
Uses E.164 or H323ID alias and a
address mapping mechanism if
gatekeepers are present in the H.
323 system.
Gatekeeper provides routing
information.
Thats all for today!

Any questions?

Thank you!