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Multimedia ProtocolsInstructor: Hamid R. Rabiee

Spring 2012

Outline

Multimedia Protocol Stack (Overview)

Signaling Protocols

H.323 (Multimedia Conferencing Protocol)

Session Initiation Protocol (SIP)

Real-time Protocols

Real-time transport protocol (RTP)

RTP control protocol (RTCP)

Real-time streaming protocol (RTSP)

Digital Media Lab - Sharif University of Technology2

Multimedia Protocol Stack


Signaling Protocols Terminology

Call Establishment and Teardown

Call Control and Supplementary Services

Call waiting, Call hold, Call transfer

Capability Exchange

Admission Control

Protocol Encoding (ASN1, HTTP)

4Digital Media Lab - Sharif University of Technology

H.323 – ITU Standard

H.323 is a multimedia conferencing protocol, which includes voice, video, and data conferencing, for use over packet-switched networks

Devices: Terminals, Gateways, Gatekeepers and Multipoint Control Units (MCUs)

Codecs: Video: H.261, H.263, H.264

Audio: G.711, G.722, G.723.1

Signaling: H.225, H.245

Transport Mechanisms: TCP, UDP, RTP and RTCP

Data collaboration: T.120

Applications

- Application Sharing - Unified Messaging

- Electronic Whiteboard - Ad-Hoc Conferencing

- File Transfer - Call No-Waiting

- Instant Messaging - Web Call Parking

- Click to Dial - Internet Call Waiting

- Voicemail Anywhere - Service Portability


The H.323 Architecture

Fig.2 – Scope of H.323 in the network


The H.323 Architecture

Fig.1 - H.323 Network Architecture and Components

Entities

Terminals

Gateways

Gatekeepers

MCUs

Protocols

Parts of H.225.0 - RAS, Q.931

H.245

RTP/RTCP

Audio/video codecs


Terminals

Endpoint on a LAN

Supports real-time, 2-way

communications with another H.323

entity

Must support:

Voice - audio codecs

Signaling and setup - Q.931, H.245,

RAS

Optional support:

Video

DataFig.3 – H.323 terminal


Gateways

Interface between the LAN and the

switched circuit network

Mandatory Functions

Transmission Format Translation

Communication Procedure Translation

Call Setup and Clearing On Both Sides

Optional Function

Media Format Translation

Example: IP/PSTN gateway

Fig.4 – H.323 gateway


MCUs

MCU - Multipoint Control Unit Endpoint that supports conferences between 3 or more endpoints

Can be stand-alone device or integrated into a gateway, gatekeeper or terminal

Typically consists of multi-point controller (MC) and multi-point processor (MP) MC - handles control and signaling for conference support

MP - receives streams from endpoints, processes them, and returns them to the endpoints in the conference

T1521250-96

MC

MC MC MP MC

MC

Gateway 1 MCU 1

LAN

MCU 2

Gatekeeper 1 Terminal 1 Terminal 2

NOTE – Gateway, Gatekeeper and MCU can be a single device.

Gatekeeper 2

Gateway 2 Gateway 3

Gatekeeper 3

MC MP

MC MP

Fig.5 – H.323 MCUs


Gatekeepers

Manage a zone (a collection of H.323 devices)

Usually one gatekeeper per zone

alternate gatekeeper might exist for backup and load balancing

Typically a software application

implemented on a PC, but can be integrated in a gateway or terminal

Mandatory functions:

Address translation (routing)

Admission control

Minimal bandwidth control - request processing

Zone management

Optional functions: Call control signaling - direct handling of Q.931 signaling between endpoints

Call authorization, bandwidth management, and call management using some policy

Gatekeeper management information (MIB)

Directory services


H.323 Protocol Stack


NETWORK

DATA LINK

PHYSICAL

TRANSPORT

SESSION

PRESENTATION

APPLICATION

Supplementary Services

Audio Signal

Video SignalData

Control

G.711 G.728

H.261 H.263 T.127

T.126

T.124

T.125/

T.122

G.722 G.729

G.723.1

RTCP RAS RTP

H.450.3 H.450.2

H.450.1H.235

H.245 H.225UDP TCP

X.224.0

RTP/RTCP (RFC 3550) is used to transmit

media such as audio and video over IP

networks

H.225.0 define the procedures and protocol

for communication within and between Peer

Elements (defines the call signaling

between endpoints and the Gatekeeper)

H.245 is the protocol used to control

establishment and closure of media channels

within the context of a call and to perform

conference controlH.450.x is a series of supplementaryservice protocols

H.460.x is a series of version-independentextensions to the base H.323 protocol

T.120 specifies how to do dataconferencingH.235 defines security within H.323systems

H.323 Protocol Stack (cont.)

RAS, Q.931 and H.245 RAS to obtain permission from a gatekeeper

RAS channel

Q.931 to establish communication and set up the call

Call-signaling channel

H.245 to negotiate media parameters

H.245 control channel

Media streams over logical channels


H.323 Call Establishment & Teardown (cont.)

Admission Request (ARQ)

The type of the call (e.g., two-party or multi-party)

The endpoint’s own id

A call identifier (a unique string)

A call-reference value (an integer)

Information of the other party

Aliases

Signaling address

Bandwidth (mandatory)

TransportQOS: endpoint or GK to reserve the resource

Admission Confirm (ACF)

Many of the same parameters as ARQ

A firm order from the GK

CallModel

Optional in ARQ; mandtory in ACF

The endpoint sends call signaling directly or via the GK

Admission Reject (ARJ)

With a reason

Bandwidth, address translation, unregistered endpoint

Disengage Confirm (DCF)



Setup

The first call-signaling message

Q.931 Protocol Discriminator

A call reference

Bearer Capability

RTP information, such as payload type

A gateway needs to perform the mapping

User-to-User information element

Mandatory: Call id, call type, conference id, the caller information

Optional: source alias, destination alias, H.245 address, destination call-signaling address

Call Proceeding

Optional

call-establishment procedures are underway

Mandatory

Protocol discriminator, call reference, and message type, user-to-user

User-to-user: similar to the setup message

Alerting

The called user is being alerted

The same parameters as Call Proceeding

Progress sent by a called gateway

the Cause information – in-band tones

the User-to-User info. – same as Call-Proceeding



Connect The called party has accepted the call

Must be sent if the call is to be completed

Call Proceeding and Alerting are optional

User-to-User information

The same as Call Proceeding

Plus Conference Identifier

The same as Setup

Release Complete

Terminate a call

No Release message

In ISDN, Release and Release Complete

Cause information element, optional

Otherwise, a Release reason in User-to-User

Facility (Q.932)

A call should be redirected

Also be used for supplementary services

User-to-User contains reason parameter

E.g., routeCallToGatekeeper


SIP - IETF Standard

Session Initiation Protocol (SIP)

SIP Elements: User Agent Client (UAC), User Agent Server (UAS)

Easy to locate users due to the flexibility in SIP to contact external location

servers to determine user or routing policies (url, email ID, e.g.

[email protected])

Server Types: Redirect Server, Proxy Server and Registrar

SIP Proxy: perform application layer routing of SIP requests and

responses.

SIP Registrar: UAC sends a registration message and the Registrar stores

registration information in a location service using a non-SIP protocol (E.g.

LDAP)


SIP…

SIP is

SIP is the core protocol for initiating, managing and terminating

sessions in the Internet

These sessions may be text, voice, video or a combination of these

SIP sessions involve one or more participants and can use unicast or

multicast communication.

An Application-layer control (signaling) protocol for creating, modifying and terminating sessions with one or more participants.

Sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution.

Members in a session can communicate via multicast or via a mesh of unicast relations, or a combination of these.


Where is SIP ?


Application

Transport

Network

Physical/Data LinkEthernet

IP

TCP UDP

RTSP SIP

SDP codecs

RTP DNS

SIP Basic Functionality

Supports 5 facets of communication:

User location: determination of the end system to be used for

communication;

User capabilities: determination of the media and media parameters to

be used;

User availability: determination of the willingness of the called party to

engage in communications;

Call setup: "ringing", establishment of call parameters at both called

and calling party;

Call handling: including transfer and termination of calls.


SIP Specifications

SIP Vulnerabilities

Proxy Impersonation

Message Tampering

Session Teardown

Spoofed BYEs

Denial of Service

Malformed packets

REGISTER and INVITE flooding

SIP Security Registration hijacking

Authenticate originators of requests

Proxy impersonation

Authenticate servers

Message tampering

Secure body and certain headers end-to-end

Session teardown

Authenticate sender of BYE

Confidentiality so attacker can’t learn To, From tags

Denial of Service

Authenticate and authorize registrations


More Detail in Appendix A

Comparison of H.323 and SIP

H.323 is another signaling protocol for real-time, interactive

H.323 is a complete, vertically integrated suite of protocols for multimedia conferencing: signaling, registration, admission control, transport, codecs

SIP is a single component. Works with RTP, but does not mandate it. Can be combined with other protocols, services

H.323 comes from the ITU (telephony).

SIP comes from IETF: Borrows much of its concepts from HTTP

SIP has Web flavor, whereas H.323 has telephony flavor.

More Detail in Appendix B


Session Description Protocol (SDP)

Multicast IP service discovery

• Multicast IP services are typically

organized into sessions

• Each session consists of one or

more media streams and exists

for a duration of time

• Problem - how to find out which

sessions are available and what are their parameters?

addresses, ports, media formats, keys, session times

• Solution - session announcements or invitations

Need for a description format

Need for a delivery mechanism


Session Description Protocol (SDP)

It is a protocol used to describe media objects and presentations

Usually, in multimedia streaming, SDP messages are sent in RTSP requests

A SDP message contains information about the session, the media streams

included in the session and information necessary to receive the media (e.g. IP

addresses, ports, formats etc.)

Applications

Setting up voice-over-IP calls.

Setting up multimedia conferences.

Event notification (subscribe / notify) IM and presence.

Text and general messaging.

Signaling transport.


Structure of an SDP description


•An SDP description uses a

textual notation describing the

parameters of a single

multimedia session

• Consists of a number of lines,

each of the format

<parameter type>=<value>

• Logically structured in three

sections

RTP/RTCP/RTSPReal-Time Protocols

Real-Time Protocol (RTP)

RTP provides end-to-end transport functions for real-time

applications Supports different payload types

All RTP and RTCP PDUs are sent to same multicast group (by all

participants)

All RTP PDUs sent to an even-numbered UDP port, 2p

All RTCP PDUs sent to UDP port 2p+1

RTP Does NOT:Define media data formats or encodings

Need media specific profiles

Handle connection setups or tear-downsNeed other protocols like SIP or H.323

Handle resource reservation Need other protocols like RSVP

Guarantee timely delivery or Quality of ServiceHowever, it does provide necessary data to application to order packets and

adjust signal quality


RTP Session

RTP session is sending and receiving of RTP data by a group of participants

For each participant, a session is a pair of transport addresses used to

communicate with the group

If multiple media types are communicated by the group, the transmission of

each medium constitutes a session.


Fig.7- Examples of RTP Session

RTP Synchronization Source

Synchronization source - each source of RTP PDUs, identified by a unique,

randomly chosen 32-bit ID (the SSRC)

A host generating multiple streams within a single RTP must use a different

SSRC per stream


Fig.8- Example of RTP Synchronization source

Some of the RTP Applications

Simple Multicast Audio Conference

Each participant uses two ports. One for audio data and the other for control

(RTCP) packets

Each participant sends audio data in small chunks of say 20 mS duration

A RTP header is added which contains the timing field that ensures that the

chunks of data are continuously played for every 20mS

Audio and Video Conference

Both audio and video are transmitted as two separate RTP sessions.

separate RTP and RTCP packets are transmitted for each medium using two

UDP port pairs and multicast addresses.

no direct coupling at the RTP level between the audio and video sessions


Mixers and Translators

Accommodate participant network resources

Mixer – Low Bandwidth

Mixer – Combining media streams

Translator – Forward RTP packets to private networks


Mixers and Translators (cont.)

RTP mixer - an intermediate system that receives & combines RTP PDUs of

one or more RTP sessions into a new RTP PDU

A mixer

may change the data format (coding) and combine the streams in any manner

Helps in re-synchronizing the incoming audio packets and forward it to the low speed

link.

Also combines several flows in a single new one appears as a new source

Will typically have to define synchronization relationships between streams =>

Sources that are mixed together become contributing sources (CSRC)

Mixer itself appears as a new source having a new SSRC


end system 1

end system 2

mixer end system 3from ES1: SSRC=6

from ES2: SSRC=23

from M: SSRC=52

CSRC list={6, 23}

Fig.9- Examples of RTP Mixer

Mixers and Translators (cont.)

Translator is an intermediate system that…

Connects two or more networks

Multicasting through a firewall

Modifies stream encoding, changing the stream’s timing

Transparent to participants

SSRC’s remain intact

Two translators are installed, on either side of the firewall, with the outside one tunneling all multicast

packets received through a secure connection to the translator inside the firewall.

Data may pass through the translator intact or may be encoded differently

The identity of individual flows remains intact

Example: going through a firewall:

end system 1

end system 2

transl.1from ES1: SSRC=6

from ES2: SSRC=23

transl.2from ES2: SSRC=23

from ES1: SSRC=6

authorized tunnel

Firewallfrom ES2: SSRC=23

from ES1: SSRC=6

Digital Media Lab - Sharif University of Technology

Fig.9- Examples of RTP Translator

33

RTP Control Protocol (RTCP)

RTCP responsible for periodic transmission of control packets

RTCP specifies report PDUs exchanged between sources and destinations of

multimedia information

receiver reception report

sender report

source description report

Reports contain statistics such as the number of

RTP-PDUs sent, number of RTP-PDUs lost,

inter-arrival jitter

Used by application to modify sender

transmission rates and for diagnostics purposes


RTP Control Protocol (RTCP) (cont.)

Functions

Feedback on the quality of data distribution

Permits everybody to evaluate the number of participants

Persistant transport-level canonical name for a source, CNAME

Usually: user@host

Will not change; even if SSRC does

Provides binding across multiple media tools for a sngle user

Distribution

use same distribution mechanisms as data packets

Underlying protocol should provide multiplexing

for data and control packets (in this case UDP)

multiple RTCP packets can be concatenated by translators/mixers

compound RTCP packet


RTCP bandwidth scaling

Solution

RTCP attempts to limit its traffic

to 5% of the session bandwidth

to ensure it can scale!

RTCP gives 75% of this rate to

the receivers; and the remaining

25% to the sender.

Example

Suppose one sender, sending video at

a rate of 2 Mbps. Then RTCP

attempts to limit its traffic to 100 Kbps.

The 75 kbps is equally shared among

receivers: With R receivers, each receiver gets

to send RTCP traffic at 75/R kbps.

Sender gets to send RTCP traffic at

25 kbps.

Problem

What happens when there is one sender

and many receivers?

RTCP reports scale linearly with the

number of participants and would

match or exceed the amount of RTP

data! More overhead than useful data!


Real-Time Streaming Protocol (RTSP)

Application layer protocol

(default port 554)

Usually runs on RTP for

stream & TCP for control

Provides the control

channel

Uses out-of-band

signaling

Usable for Live

broadcasts / multicast Also known as “Network

remote control” for multi-media servers.


web

browser

media player

Web Server

Web

Server/Media

serverRTSPpres. desc,streaming commands

RTP/RTCPaudio/video content

Presentation

descriptor

HTTPpresentation descriptor

RTSP Features

“Rough” synchronization (fine-grained)

Virtual presentations = synchronized playback from several servers

Load balancing using redirection at connect, during stream

Supports any session description

Device control camera pan, zoom, tilt

Caching: similar to HTTP

Application

Streaming of multi-media via Internet

Video conferencing & lectures.

Broadcasting of entertainment.

Remote digital editing

Voice mail


RTSP Limitations

Bandwidth: 64 – 128 kb/s for talking heads, 1.5Mb/s for movies

Quality: packet loss, predictability

Billing Infrastructure: pay-per-view?

Cheap Receivers: shouldn’t cost more than set-top box


RTSP Session

media server

RTSPserver

datasource

media player

AVsubsystem

RTSPclient

RTSP OK

RTSP PLAY

RTSP OK

RTP AUDIO

RTP VIDEO

RTSP TEARDOWN

RTSP OK

get UDP portchoose

UDP port

RTSP SETUP

Default port

554

RTCP

TCP

UDP


More Detail in Appendix C

RTSP Example


41

SETUP

Specifies the transport mechanism to be used for the streamed media

Transport header specifies the transport parameters acceptable to the

client

Response contains transport parameters selected by the server

Server generates session identifiers in response to SETUP requests

Client can issue SETUP to change parameters for already streaming

media

42

PLAY

Tells server to start sending data via mechanism specified in SETUP

Plays from beginning to end of range specified

Pauses at end as if PAUSE has been issued

Requests may be pipelined

Range header can have time parameter

Legal to have PLAY without Range header

Scale header can be used to change viewing rate

Can be used for fast forward or rewind

43

PAUSE

Causes stream delivery to halt

Can pause entire presentation or selected stream

Server resources continue to be reserved

Server will terminate session after timeout period expires (specified in

SETUP)

Can contain Range header with one value to specify when to pause

Must fall in one of the PLAY ranges

44

TEARDOWN

Stops delivery of stream

Frees up resources on the server

Can stop single stream or entire presentation

Example

Client -> Server:

TEARDOWN rtsp://example.com/fizzle/foo RTSP/1.0

CSeq: 892

Session: 12345678

Server -> Client:

RTSP/1.0 200 OK

CSeq: 892

45

More Detail in Appendix D

Next Session

IP Multimedia Subsystem (IMS)


Appendix A

Digital Media Lab - Sharif University of Technology 47

SIP Commands


Method Description

INVITE A session is being requested to be setup using a specified media

ACK Message from client to indicate that a successful response to an INVITE has been received

OPTIONS A Query to a server about its capabilities

BYE A call is being released by either party

CANCEL Cancels any pending requests. Usually sent to a Proxy Server to cancel searches

REGISTER Used by client to register a particular address with the SIP server

Table 1 – SIP Commands

SIP Call Establishment & Teardown


A BProxy Proxy

(Registrar local services)

……. ……. …….

Register200 Ok

Register200 Ok

Invite100 Trying Invite

100 Trying Invite180 Ringing

180 Ringing180 Ringing

200 Ok200 Ok

200 Ok

AckMedia Session

Bye

200 Ok

An Example of SIP Message


INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 167.180.112.24

From: sip:[email protected]

To: sip:[email protected]

Call-ID: [email protected]

Content-Type: application/sdp

Content-Length: 885

c=IN IP4 167.180.112.24

m=audio 38060 RTP/AVP 0

Notes:

HTTP message syntax

sdp = session description protocol

Call-ID is unique for every call.

• Here we don’t know

Bob’s IP address.

Intermediate SIP

servers will be

necessary.

• Alice sends and

receives SIP messages

using the SIP default

port number 506.

• Alice specifies in Via:

header that SIP client

sends and receives

SIP messages over UDP

SIP Request Example


INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP host.wcom.com:5060

From: Alan Johnston <sip:[email protected]>

To: Jean Luc Picard <sip:[email protected]>


CSeq: 1 INVITE

Contact: sip:[email protected]

Subject: Where are you these days?


Content-Length: 124

v=0

o=ajohnston 5462346 332134 IN IP4 host.wcom.com

s=Let's Talk

t=0 0

c=IN IP4 10.64.1.1

m=audio 49170 RTP/AVP 0 3

SIP Response Example

SIP/2.0 200 OK

Via: SIP/2.0/UDP host.wcom.com

From: Alan Johnston <sip:[email protected]>

To: Jean Luc Picard <sip:[email protected]>


CSeq: 1 INVITE

Contact: sip:[email protected]

Subject: Where are you these days?


Content-Length: 107

v=0

o=picard 124333 67895 IN IP4 uunet.com

s=Engage!

t=0 0

c=IN IP4 11.234.2.1

m=audio 3456 RTP/AVP 0


SETUP Message

Client -> Server:

SETUP rtsp://example.com/foo/bar/baz.rm RTSP/1.0

CSeq: 302

Transport: RTP/AVP;unicast;client_port=4588-4589

Server -> Client:

RTSP/1.0 200 OK

Cseq: 302

Date: 23 Jan 1997 15:35:06 GMT

Session: 47112344

Transport: RTP/AVP;unicast;

client_port=4588-4589;server_port=6256-6257

53From RFC 2326

PLAY Message

Client -> Server:

PLAY `rtsp://audio.example.com/twister.en RTSP/1.0

CSeq: 833

Session: 12345678

Range: smpte=0:10:20-;time=19970123T153600Z

Server -> Client:

RTSP/1.0 200 OK

CSeq: 833

Date: 23 Jan 1997 15:35:06 GMT

Range: smpte=0:10:22-;time=19970123T153600Z

54From RFC 2326

Appendix B


Comparison of

H.323 and SIP (cont.)


H.323 Call Establishment & Teardown


a. Call with GKs and Direct-

Endpoint Call Signaling

Fig.6- Examples of H.323 Call Establishment

b. Gatekeeper routed/direct-

endpoint call signaling


Appendix C

RTCP Packet Type

SR: Sender report, for transmission and

reception statistics from participants that

are active senders

Carry statistics from the active senders


RR: Receiver-report, for transmission and

reception statistics from participants that

are active receivers

Carry statistics from the active receivers

Fig.10- RTCP Receiver Report Format

Fig.11- RTCP Sender Report Format

RTCP Packet Type (cont.)

SDES: Source Description packet

Allow the binding of SSRC value with an actual identification of the user.

End points need to send an SDES packet at the beginning of the session

BYE: ends a user’s participation in a call.

APP: Application-specific RTP packet.


RTSP Commands

OPTIONSC S

determine capabilities of server/clientC S

DESCRIBE C S get description of media stream

ANNOUNCE C S announce new session description

SETUP C S create media session

RECORD C S start media recording

PLAY C S start media delivery

PAUSE C S pause media delivery

REDIRECT C S redirection to another server

TEARDOWN C S immediate teardown

SET_PARAMETER C S change server/client parameter

GET_PARAMETER C S read server/client parameter



Appendix D

RTSP Message sequence

C

W

V

A

C->V : SETUP rtsp://video.example.com/twister/video.en RTSP/1.0

Cseq:1

Transport : RTP/AVP/UDP;unicast;client_port=3058-3059

A-> C : RTSP/1.0 200 OK

Cseq:1

Session: 23456789


server_port=5002-5003

C -> W : GET/Twister.sdp HTTP/1.1

Host: www.example.com

Accept: application/sdp

W-> C : HTTP/1.0 200 OK


C-> A : SETUP rtsp://audio.example.com/twister/audio.en RTSP/1.0

Cseq:1


A-> C : RTSP/1.0 200 OK

Cseq:1

Session: 12345678


server_port=5000-5001

63


RTSP Message sequence (cont.)

C

W

V

A

C->V: PLAY rtsp://video.example.com/twister/video RTSP/1.0

Cseq: 2

Session: 23456789

V->C: RTSP/1.0 200 OK

Cseq: 2

Session: 23456789

RTP-Info: url=rtsp://video.example.com/twister/video;

seq=12312232;

C->A: PLAY rtsp://audio.example.com/twister/audio.en RTSP/1.0

Cseq: 2

Session: 12345678

A->C: RTSP/1.0 200 OK

Cseq: 2

Session: 12345678

RTP-Info: url=rtsp://audio.example.com/twister/audio.en;

seq=876655;


RTSP Message sequence (contd.)

C

W

V

A

C->A: TEARDOWN rtsp://audio.example.com/twister/audio.en RTSP/1.0

Cseq: 3

Session: 12345678

A->C: RTSP/1.0 200 OK

Cseq: 3

C->V: TEARDOWN rtsp://video.example.com/twister/video RTSP/1.0

Cseq: 3

Session: 23456789

V->C: RTSP/1.0 200 OK

Cseq: 3


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