Session No.9

VoIP H.323

A PSTN Network

PSTN Network A

Class 4 Switch

Class 5 Switch

Class 5 SwitchClass 4

Switch

PSTN Network B

Class 4 Switch

International

GWClass 5 Switch

International

GW

PBX

Network NodeNetwork NodeUser User

Network to Network Signalling

User to Network Signalling

Network to User Signalling

Access Signalling Access SignallingNetwork Signalling

Signalling protocols❖Access Signalling: Digital Subscriber System 1 (DSS1) – PRI, QSIG , CAS❖Network Signalling: Signalling system 7 – ISUP, TUP ,CAS

Two modes of signaling•CAS – Channel Associated Signaling•CCS – Common Channel Signaling (e.g. SS7)

Signaling

Signaling in PSTN – SS7

IP

VOICE

STP

STP

STP

STP

SCP

SSP

SS7

SSP

STP

Packet router, used only in quasi associated signaling mode

STP routes messages based on Absolute point code references or Global title address references

SCP

Provides call control external to the SSPs

Coordinates actions of multiple elements as needed by service

One or more applications generally associated with each SCP

IP

Provides a means of interacting with the subscriber via voice

Media DialogsProprietary signaling

SSP

Routes voice traffic between• Locally connected subscribers/devices• Other switches

Controlling devicesMultiple device types (e.g., 10 party lines,

ISDN lines)AlertingDisplays

Feature processingBill generation

SS7 signaling

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.

Universality of communication servicethrough interconnect agreements between sub-network operators

Communication can be universal but no network interconnect mandatory at service level

Charging based on usage of the communicating service

Charging based on flat rate or volume of transported data

QoS constraints of a communication known and guaranteed by the network

User-defined QoS constraints provided to the network

Communication protocols are network specific and transparent to users

Communicating users agree on which communication protocol to use

Communication controlled (managed) by the network

No direct control by network over the communication except access rights

Three-party communication model (caller-network-callee)

Two-party communication model (client/server or peer-to-peer)

Telecom NetworkPacket Network

Difference between the Networks

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

Redundant routes through networkRedundancy within each network element

How reliability acheieved

ATM, FR, native IP in access, ATM, native IP in coreTDMTransport

SIP, H.323CAS, ISDN,SS7Signaling

Variable64 kbpsBandwidth per call

In separate telephony serversMostly integrated in switching system

Call Processing Intelligence

Gateways, switches, routersClass 4, Class 5 switching systems

Network Elements

NoYesNetwork resource reserved at call setup

NoYesQoS Guarantees

Packet switchingTDM circuit switching

Underlying Technology

Internet TelephonyPSTN

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]

TCP Packet

UDP Packet

Data

ChecksumLength

Destination portSource port

TCP Or UDP ?

Without ACKs, the network cannot signal congestion to the sender.

Network devices can take advantage of TCP ACKs to control the behavior of senders.

Congestion controls

ACKs, which are used in TCP to control packet flow, are not returned.

The receiver can signal the sender to slow down.

Flow controls

UDP does not insert sequence numbers. The packets are expected to arrive as a continuous stream or they are dropped.

Sequentially numbers packets.Packet sequencing (provide information about the correct order of packets)

Since UDP does not return ACKs, the receiver cannot signal that packets have been successfully delivered. Lost packets are not retransmitted.

Returns ACKs (acknowledgments).Guaranteed message delivery

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

Connection setup

UDPTCPService

VoIP - History

1995: Vocaltec, Inc.Internet Phone v1.0

1996: - H323 v1- SIP Draft

1999: SIP RFC 2543 2002 June:

SIP RFC 3261

2000 Nov: H323 v41998 Jan:

H323 v2

2000 Nov: MEGACO/H.248

v1

1998 Oct: MGCP v1

2003: H323 v5

2004 July: TGCP I08

VoIP Network Architecture

Softswitch

App Server

MediaServer

Softswitch

MG

SGSTP

SSP

PSTN Carrier Network

VoIP Enterprise Network

PBX

MG

PSTN Enterprise Network

IP Carrier NetworkTransport Layer

Call Control Layer

App Layer

SIGTRAN

MGCP, Megaco MGCP,

Megaco

Parlay, JAIN

SIP, MGCP

SIP –T,

BICC

RTPMedia

SS7

Transport Layer

• Routers, Repeaters etc

Call Control Layer

• Phones• Media Gateway• Signaling Gateway• Softswitch

Application layer

• App Server• Media Server

VoIP Network - Phones

Softswitch

App Server

MediaServer

Softswitch

MG

SGSTP

SSP

PSTN Carrier Network

VoIP Enterprise Network

PBX

MG

PSTN Enterprise Network

IP Carrier NetworkTransport Layer

Call Control Layer

App Layer

SIGTRAN

MGCP, Megaco MGCP,

Megaco

Parlay, JAIN

SIP, MGCP

SIP –T,

BICC

RTPMedia

SS7

Phones (End User Terminal)• Intelligent or dumb

• Capable of hosting applications

• SIP, MGCP phones available

VoIP Network – Media Gateway

Softswitch

App Server

MediaServer

Softswitch

MG

SGSTP

SSP

PSTN Carrier Network

VoIP Enterprise Network

PBX

MG

PSTN Enterprise Network

IP Carrier NetworkTransport Layer

Call Control Layer

App Layer

SIGTRAN

MGCP, Megaco MGCP,

Megaco

Parlay, JAIN

SIP, MGCP

SIP -T

RTP

Media Gateway• Terminates different types of

media interfaces (TDM, IP, ATM)

• Converts one media format to another format e.g, G.711 to G723

• Controlled by Softswitch via gateway control protocols such as MGCP, Megaco

Media

SS7

VoIP Network – Signaling Gateway

Softswitch

App Server

MediaServer

Softswitch

MG

SGSTP

SSP

PSTN Carrier Network

VoIP Enterprise Network

PBX

MG

PSTN Enterprise Network

IP Carrier NetworkTransport Layer

Call Control Layer

App Layer

SIGTRAN

MGCP, Megaco MGCP,

Megaco

Parlay, JAIN

SIP, MGCP

SIP -T

RTPMedia

SS7

Signaling Gateway• Terminates different types of

signaling interfaces

• Transparently communicates signaling from one interface type to another (SIGTRAN)

• Might convert signaling in one format to another format

VoIP Network - Softswitch

Softswitch

App Server

MediaServer

Softswitch

MG

SGSTP

SSP

PSTN Carrier Network

VoIP Enterprise Network

PBX

MG

PSTN Enterprise Network

IP Carrier NetworkTransport Layer

Call Control Layer

App Layer

SIGTRAN

MGCP, Megaco MGCP,

Megaco

Parlay, JAIN

SIP, MGCP

SIP -T

RTPMedia

SS7

Softswitch/MGC/Call Agent

• Controls Media Gateways & Signaling Gateways

• Routes VoIP sessions/calls to other softswitches, phones etc

• Completely software based

VoIP Network – Application Server

Softswitch

App Server

MediaServer

Softswitch

MG

SGSTP

SSP

PSTN Carrier Network

VoIP Enterprise Network

PBX

MG

PSTN Enterprise Network

IP Carrier NetworkTransport Layer

Call Control Layer

App Layer

SIGTRAN

MGCP, Megaco MGCP,

Megaco

Parlay, JAIN

SIP, MGCP

SIP -T

RTPMedia

SS7

Application Server• Service Logic Execution

• Hosts applications

• Caters to multiple softswitches

• Can be hosted even outside the carrier network

• SIP/Parlay/JAIN/CPL/proprietary

VoIP Network – Media Server

Softswitch

App Server

MediaServer

Softswitch

MG

SGSTP

SSP

PSTN Carrier Network

VoIP Enterprise Network

PBX

MG

PSTN Enterprise Network

IP Carrier NetworkTransport Layer

Call Control Layer

App Layer

SIGTRAN

MGCP, Megaco MGCP,

Megaco

Parlay, JAIN

SIP, MGCP

SIP -T

RTPMedia

SS7

Media Server• Provides media capabilities

needed for applications

• Announcements, Voice Mail, IVR, 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: LAN’s

– 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 Reference for key feature summary

H.323 Version 3 http://www.packetizer.com/iptel/h323/whatsnew_v3.html

Date

H.323 Version 1 New release. Refer to the specification.http://www.packetizer.com/iptel/h323/

May 1996

H.323 Version 2 http://www.packetizer.com/iptel/h323/whatsnew_v2.html

January 1998

September 1999

H.323 Version 4 November 2000

http://www.packetizer.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

RTCP

RTP

IP

MGCP

Call Control and Signaling Signaling and Gateway Control

Media

Q.931 RAS

UDP

SIPH.245

Audio/Video

RTSP

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– H.245 - capabilities exchange

carried unreliably 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

Request for status information from gatekeeper to terminal

Information Request(IRQ)

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

Disengage Request(DRQ)

Request for changed bandwidth allocation, from terminal to gatekeeper.Gatekeeper either confirms(BCF) or rejects(BRJ)

Bandwidth Request(BRQ)

Request for access to packet network from terminal to gatekeeper. Gatekeeper either confirms(ACF) or rejects(ARJ)

Admission Request(ARQ)

Request from terminal or gateway to register with a gatekeeper. Gatekeeper either confirms(RCF) or rejects(RRJ)

Registration Request(RRQ)

FunctionMessage

Important H.323 messages:RAS

Recommended default time values for response to RAS messages and subsequent retry counts if response is not received

RAS timers and Request in progress(RIP)

Response to IRQ. May be sent unsolicited by terminal to gatekeeper at predetermined intervals

Info Request Response(IRR)

FunctionMessage

Important H.323 messages: H.225

Indicates release of call if H.225.0(Q.931)call signaling channel is open.Sent by H.323 terminal

Release complete

Indicates desire of calling entity to setup a connection to the called entity

Setup

Acceptance of call by called entity.Sent from called entity to calling entity

Connect

Requested call establishment has been initiated. Sent by called user

Call proceeding

Called user has been alerted - “Phone is ringing”. Sent by called user

Alerting

FunctionMessage

Important H.323 messages: H.225

Requests call status.Sent by gatekeeper or endpoint to another endpoint

Status enquiry

Responds to an unknown call signaling message or Status enquiry message.Provides call state information

Status

FunctionMessage

Important H.323 messages: H.245

Used by receive terminal to request particular modes of transmission from a transmit terminal.Mode types:Audio mode, Video mode, Data mode, Encryption mode

Request mode

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

Close logical channel

Acceptance of call by called entity.Sent from called entity to calling entity. Possible replies:Acknowledge, Reject, Confirm

Open logical channel

Contains information about a terminals capability to transmit and receive multimedia streams. Possible replies: Acknowledge, Reject, Release

Terminal capability set

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

Master Slave Determination

FunctionMessage

Important H.323 messages: H.245

Indicates end of H.245 session.Terminal will not send any more H.245 messages

End session command

Commands far-end terminal to send its transmit/receive capabilities

Send terminal capability set

FunctionMessage

A high-level communication exchange between two endpoints (EP) and two gatekeepers (GK)

SIP vs. H.323

SIP

ITU.IETF.

Peer-to-Peer. Peer-to-Peer.

Telephony based. Borrows call signaling protocol from ISDN Q.SIG.

Internet based and web centric. Borrows syntax and messages from HTTP.

Intelligent H.323 terminals.Intelligent user agents.

H.323 Gatekeeper.SIP proxy, redirect, location, and registration servers.

Widespread.Interoperability testing between various vendor’s products is ongoing at SIP bakeoffs.SIP is gaining interest.

Information

Standards BodyRelationship

Origins

Client

Core servers

Current Deployment

Interoperability IMTC sponsors interoperability events among SIP, H.323, and MGCP. For more information, visit: http://www.imtc.org/

H.323

SIP vs. H.323

Information H.323SIP

Capabilities Exchange

Supported by H.245 protocol. H.245 provides structure for detailed and precise information on terminal capabilities.

SIP uses SDP protocol for capabilities exchange. SIP does not provide as extensive capabilities exchange as H.323.

Control Channel Encoding Type

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

Server Processing

Version 1 or 2 – Stateful.Version 3 or 4 – Stateless or stateful.

Stateless or stateful.

Quality of Service

Bandwidth management/control and admission control is managed by the H.323 gatekeeper.The H323 specification recommends using RSVP for resource reservation.

SIP relies on other protocols such as RSVP, COPS, OSP to implement or enforce quality of service.

SIP vs. H.323

Information H.323SIP

Security 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.

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 Location and Call Routing

Uses E.164 or H323ID alias and a address mapping mechanism if gatekeepers are present in the H.323 system.Gatekeeper provides routing information.

Uses SIP URL for addressing.Redirect or location servers provide routing information.

That’s all for today!

Any questions?

Thank you!