Multimedia applications can be classified into one of the three categories:

(1)

UNIT-IV:

NETWORKED MULTIMEDIA

EL-447: Multimedia Systems & Networks 1

(2)

Introduction (1)



Multimedia applications can be classified into one of the three categories:



Interpersonal communication



Interactive applications over the Internet



Multimedia for entertainments



All these applications involve more than one media integrated together.



Standards are needed for:



Compression of different types of media; (covered earlier)



How integrated information streams are structured?

(to be discussed in this unit)

(3)

Introduction (2)



Since different networks operate in different way, there are number of standards each intended for use with specific type of networks.

(4)

Multimedia Transmission Requirements (Qualitative)



Response of the human Ear:

 One important property of our ear is it is more sensitive to the changes of the signal levels rather than the absolute values.



Response of the human Eye:

 Retains for few msec before decaying.



Tolerance to error:

 Higher error rate tolerance for uncompressed signals.



Tolerance to Delay and variation in delay:

 Small delay for live application



Lip Synchronization

 The time gap between the audio objects and the video objects.

 Most critical aspect

(5)

Performance Parameters



Synchronization Accuracy Specification (SAS) factors used to specify goodness of sync:



Delay: Acceptable time gap between transmission and reception.



Delay Jitter: Instantaneous difference between the desired presentation times and actual presentation times of streamed multimedia objects.



Delay skew: Average difference between the desired and actual presentation times.



Error rate: Level of error specified in terms of bit error rate (BER).

(6)

SAS Factors for Audio and Video



SAS factors for Audio:

 Delay: For conversation, one-way delay should be in 100-500 msec range, which requires echo cancellation.

 Delay Jitter: 10 times better than delay

 For example, if the delay is 100 milliseconds, then the delay jitter should be less than 10 milliseconds, so it should be ten time better than the delay.

 Lip Synchronization: Should be better than 80 msec.

 Error rate:

 Less than 0.01 for telephones.

 Less than 0.001 for uncompressed CD.

 Less than 0.0001 for compressed CD quality audio.

 SAS factors for video:

6

Delay/Jitter Error rate

HDTV < 50 msec <10^-5

Broadcast TV <100 msec < 10^-4 Video Conferencing <500 msec <10^-3

(7)

Traffic Characterization Parameters



Due to the variability of the bit frame



Two categories:



Constant bit-rate (CBR) applications:

 Example: Uncompressed digitized voice/video transmission



Variable bit-rate (VBR) applications:

 Compressed audio and video transmission



Most multimedia applications generate VBR traffic.



VBR traffic causes burstiness in the traffic.



Burstiness ratio= Mean bit-rate/Peak bit-rate

(8)

Quality of Service (QoS) Parameters



QoS is the concept for specifying how “good” the offered services are.



Quality of service is a concept based on the statement that not all applications need the same performance from the system/network over which they run.



Thus, applications may indicate their specific

requirements to the network, including cost, before they actually start transmitting data.



QoS parameters can be categorized as:

 Network QoS

 Parameters associated with a communication network

 Application QoS

 Parameters that determines the quality of particular application

 Same as SAS parameters discussed earlier.

(9)

Network QoS



Major parameters that defines QoS are:

 Throughput – the total amount of work completed during a specific time interval.

 Bit-rate, bandwidth

 Burstiness

 Ratio of average to peak bit-rate

 Delay – the elapsed time from when a request is first submitted to when the desired result is produced.

 Minimum/maximum transit delay

 Important for response-time and perception

 Jitter – the delays that occur during playback of a stream.

 Maximum Jitter (delay variance)

 Important for synchronization

 Reliability – how errors are handled during transmission and processing of continuous media.

 Acceptable bit-error rate

 Acceptable packet error rate

(10)

 QoS for CBR channel (circuit switched network):

 Bit-rate

 The mean bit-rate

 Transmission delay

 QoS for packet-switched network

 Maximum packet-size

 Mean packet transfer rate

 Mean packet error rate

 Mean packet transfer delay

 Worst-case jitter

 Transmission delay

(11)

Delay in packet-switched networks (1)

Packets experience delay on end-to-end path



four sources of

delay at each hop:



nodal processing:

 check bit errors

 determine output link



queuing

 time waiting at output link for transmission

 depends on congestion level of router

A B

propagation transmission

nodal

processing queueing

(12)

Delay in packet-switched networks (2)

Transmission delay:



R = link bandwidth (bps)



L = packet length (bits)



time to send bits into link = L/R

Propagation delay:



d = length of physical link



s = propagation speed in medium (~2x10

⁸

m/sec)



propagation delay = d/s

A B

propagation transmission

nodal

processing queueing

Note: s and R are very different quantities!

(13)

Application QoS Parameters



Required bit-rate or mean packet transfer rate



Maximum start-up delay



Maximum end-to-end delay



Maximum Delay variation/Jitter



Maximum round-trip delay

(14)

Multimedia Streams



Multimedia stream may consists of combination of following streams:

 Video (H.261, H.263, MPEG-1/2, etc)

 Audio (G711, G722, MP3, AAC etc)

 Data (eg. shared presentation tools)

 Signalling (metadata, channel setup)



Need to store or transmit combination of these streams together.



Different transmission channels have different error rates.

 Need to protect data against corruption.



Need to allow re-synchronization after corruption, fast- forward, channel switching, etc.

(15)

• Media distribution

- Deliver media contents to users

 Delivery via disc:

– Merits: Large storage, high audiovisual quality – Demerits: long delivery time, inflexible

Delivery via PSTN/ISDN Delivery via Internet:

 Non realtime delivery

:

•

download service

: download all data, save to disc, and play using data file transfer protocols like ftp and http via ftp and web-server.

 Realtime delivery

:

•

streaming service

:



>download & play simultaneously, partial data in buffer, no data in disc

• May use http and web server to provide limited streaming service

• Often use RTSP/RTP and media server for rich streaming service

Multimedia Distribution



15 EL-447: Multimedia Systems & Networks

(16)

HTTP

Web Server

Long start-up latency Potential waste of traffic

AV File

Web Browser

Media Player

Non Real time Delivery: Downloading

(17)

HTTP

file

Web Server

RTSP/MMS/HTTP RTP/RTCP

Streaming Server

AV File meta

Web Browser

Media Player

Real-time Delivery: Streaming

(18)

Internet

Media Server Client 1

Media Data 1

Request Media Data 2

Client 2

Streamed Media

Files MoD example

• Media on demand

media are

(MoD)

saved in media server as streamed file format - -

- - -

Streamed

Clients, i.e., media player, access media contents independently

Media content is played from the file beginning for each client’s request User can control playing, such fast forward, pause, …

Like rent a video tape or DVD and replay it in your cassette/DVD palyer

Media Player 2 Media Player 1 Media

Streaming

& Access Control

18

Streamed Media On Demand Delivery

(19)

Internet Client 1 Media Server

Streaming

Client 2

Streamed Media

Files Audio

Video

broadcast example broadcast example Live Broadcast

• Media Internet Broadcast (MIB) or Webcast

- - - - -

Media may be stored in server or captured lively and encoded in realtime Clients can join a broadcast and same media content goes to all clients Users watch/listen the broadcast from the current state not from beginning Users can’t control its playing such fast forward, stop, etc.

Like conventional radio and TV broadcast Realtime

Encoder

Media Player 2

Join

Media

& Access Control

Media Player 1

Join

19

Streamed Media Broadcast

(20)

Stream Server

with encoder Stream Client

with decoder Routers

 Real Networks - -

-

Real Producer: create streamed media file, end with “filename.rm”

Real Server: streaming media to delivery across network Real Player: streamed media player in RM format

 Windows Multimedia Technologies - Media

- Media - Media

Encoder: create streamed media file, end with “filename.asf/.wmv”

Server: streaming media to delivery across network Player: streamed media player in ASF/WMV format

 QuickTime - -

-

QuickTime QuickTime QuickTime

Pro: create streamed media file, end with “filename.qt”

Streaming Server (Mac) and Darwin Streaming Server Player: streamed media player in QT format

 Audio/MP3: Liquid Audio, SHOUTcast, icecast

Popular Stream Media Server and Player

(21)

 Delay and Jitter

Key Points in Streaming Media Service

(22)



Smooth Dealy & Jitter via buffer

* Client-side buffering,

* Playout delay,

* Compensate for network delay & jitter

constant bit

constant bit (drain rate) video

playout at client

variable _stop_continuously

networ k

- How large for prefetched data - How long for playout waiting time

time client playout

delay

buffered video

rate

without Questions:

client video reception

- rate video

transmission

delay

Key Points in Streaming Media Service (Cont)

(23)

 Trade-off between media quality and network bandwidth

- - -

Data amount of continuous media, especially video, is extremely large

Current Internet bandwidth is relative small, 28K/56K modem, ADSL, Cable, LAN, etc.

Before delivery, clarify targeted users and their available bandwidth

Low quality

GSM

Internet

Medium quality

Low quality Modem

GRPS

Multicast

Router High-speed LAN

Sender R

Video

Key Points in Streaming Media Service (Contd.)

 Limited Server Resource:

 Limited computational power in processing many media streams.

 Limited storage space in saving many media data in server.

 Limited I/O performance in outputting many streams to the network.

How to serve many users simultaneously?

(24)

 Unicast  Multicast

Key Points in Streaming Media Service (Cont)

(25)

Network

Unicast Example: Multiple Independent Streams

(26)

Servers Intermediaries Clients

Multicast Example: Single Stream and Copy

(27)

 Cache technology

- Increase IO via putting media data in memory - The larger memory, the better

 Distributed server cluster and proxy media server

- Use a group of servers to improve processing performance - Use proxy

Server Cluster

server to reduce number of users’ direct accesses to server

• Drop frames

– Drop B,P frames if not enough bandwidth Proxy

Server • Quality Adaptation – Transcoding

• Change quantization value

• Change coding rate

• Video staging, caching, patching –

– –

Staging: store partial frames in proxy

Prefix caching: store first few minutes of movie Patching: multiple users use same video

Client Client

Key Points in Streaming Media Service (Cont)

(28)

Capture Encoding Serving Internet distribution Playback

Media Player

Source

Encoder Media IP network

Server Media Player Media

Proxy

Proxy Media Server

(29)

Reduce

network traffic Reduce response time to client

Reduce server’s load

Server Intermediary Client

Proxy Server: Reduce Traffic, Time, Load

(30)

Media Streaming Service Access Process

(31)

HTTP (Control and Data)

RTSP/TCP (Control)

RTP/UDP (Media Data) RTCP/UDP (RTP Control)

Scheduler

Media Player

Media Server

RTSP Handler

RTP Handler

File Media Parsing Storage

Web Browser

Web Server

HTTP HTML Handler Files

Media Streaming Service Modules

(32)

TCP

(till now)

RTP RTCP

RTSP

Protocol Stack for Multimedia Services

(33)

 Real-Time Streaming Protocol (RTSP) is defined in RFC 2326 by IETF in 1998

RTSP is a control protocol intended for:

a standard



– –

retrieval of media from a media server

establishment of one or more synchronized, continuous-media streams

control of such streams –



RTSP RTSP

– use

can be seen as a “network remote is not used to deliver the streams

RTP or similar for that

control”

What is RTSP?

(34)

Web Server

HTTP

presentation descriptor

Presentation descriptor

Media server

RTSP

pres. desc,streaming commands

RTP/RTCP

audio/video content

media player

web browser

HTTP and RTSP

(35)

Default port 554

RTSP SETUP RTSP OK RTSP PLAY

RTSP OK RTSP TEARDOWN

RTSP OK

TCP

choose UDP port RTP VIDEO

RTP AUDIO UDP

RTCP

RTSP client

U

AV subsystem

media player RTSP

server

get U DP port

data source media server

RTSP Session

(36)

•

• Realtime Transport Protocol (RTP) is an IETF standard Primary objective: stream continuous media over a best- effort packet-switched network in an interoperable way.

Protocol requirements:

•

– Payload Type Identification: what kind of media are we streaming?

Sequence Numbering: to deal with lost and out-of-order packets.

–

– Timestamping: to compensate for network jitter in packet delivery.

Delivery Monitoring: how well is the stream being received by the – destinations?

RTP does not guarantee QoS (Quality of Service), i.e., reliable, on-time delivery of the packets (the underlying network is expected to do that).

RTP typically runs on top of UDP, but the use of other protocols is not precluded

•

• What is RTP?

(37)

• RTP is composed of two closely-linked parts:

– –

The Real-Time Transport Protocol (RTP), used to carry real-time The RTP Control Protocol (RTCP), used to:

data

•

Monitor and report Quality of Service

Convey information about the participants of a session

• Two connective ports are needed for media data transmissions – Even number 2n for RTP and odd number 2n+1 for RTCP

• RTP defines the concept of a profile, which completes the specification for a particular application:

– Media encoding specifications, Payload format specifications

RTT, RTCP and Session

37

(38)

Standards Relating to

Interpersonal Communication

(39)

Overview



Interpersonal communications (IPC) includes:

 Telephony

 Video telephony

 Data conferencing

 Video conferencing etc.



Networks for IPC:

 Circuit Switched networks

 PSTN

 ISDN

 Packet switched networks

 LAN

 Intranet

 Internet



Separate standards for each network, mainly defined by ITU-T.

(40)

PSTN: Public Switched Telephone Network

SCP

SS7 Signaling Network

Dial/Comm Control Most service logic in local switches Signaling

Circuit Switch

Circuit Switch Circuit

Switch Circuit-based Trunks 64 kb/s digital voice Typically analog

“loop”, conversion to

digital at local switch Media stream

•

Different pair of telephones travels over a parallel/separate links Features: High voice quality, low bandwidth efficiency, inflexible

Traditional Telephony over PSTN

40

(41)

Standards used for Circuit-Switched networks

Standard H.320 H.324 H.321 H.310

Network ISDN PSTN B-ISDN (ATM) B-ISDN (ATM)

Audio Codec G.711, G.722,

G.728 G.723, G.729 G.711, G.722, G.728

G.711, G.722, G.728, MPEG-1

Video Codec H.261 H.261

H.263 H.261 H.261

MPEG-2 User Data

Application T.120 T.120 T.120 T.120

Multiplexer/

Demultiplexer H.221 H.223 H.221 H.221

System

Control H.242 H.245 H.242 H.245

Call setup

(Signaling) Q.931 V.25 Q.931 Q.2931

(42)

H.320



Standard for use in end systems that supports a range of applications over an ISDN.



Data rate: p × 64 kbps, p=1,2,…,31



For video telephony: p=1 or 2



For video conferencing: p is greater than 2.



Audio:

 Audio/speech compression can be selected from one of the three ITU-T recommendations: G.711, G.721 and G.728. G.711 is the default standard.

 G.711 (mu-law), G.722 (64kbit/s), G.728 (16kbit/s) audio

 Choice of standard depend on the bandwidth available for the audio.

 G.711 and G.721 require 64kbps, they are used only when multiple 64kbps channels are available.

 G.728 requires only 16 kbps, it can be used when a single 64kbps channel is available. EL-447: Multimedia Systems & Networks 42

(43)

H.320 (Contd.)



Video:

 Video compression standard is H.261, with constant output bit-rate (achieved by varying the quantization parameter dynamically).

 It supports either QCIF or CIF resolutions only.

 Actual resolution used is negotiated at the start of the conference.



Call setup/System control:

 Signaling (call setup) procedure associated with an ISDN is defined in recommendation Q.931.

 This involves exchange of message over a separate 16kbps signaling channel.

 The bandwidth associated with audio, video and data streams are negotiated and fixed at the start of a conference.

 System control standard (H.242) is primarily concerned with the negotiation of bandwidth/bit-rate for each stream.

(44)

• T.120 defines multipoint data communications standards in a multimedia conferencing environment

Provides mechanism to identify the participating nodes and exchange information

•

Enables multiple simultaneous conference handling and Consists of a set of protocols:

participation

Core Protocols:

 T.123: Transport Protocol

 T.124: Generic Conference Control (GCC)

 T.125/T.122 Multipoint Communication Service (MCS) Optional Protocols

 T.121:

 T.126:

 T.127:

 T.128:

Generic Application Template (GAT)

MultiPoint Still Image and Annotation Protocol (NSIA) Multipoint Binary File Transfer Protocol (MBFT)

Application Sharing (AS)

H.320: T.120 Multipoint Data Conferencing

(45)

T.120 Application

Protocol Recommendations

Template (GAT)

Infrastructure Recommendations

T.121

Application Protocols Application Protocol

Generic Application

T.120

Generic Conference Control (GCC) T.124

Multipoint Communication Service (MCS) T.122/T.125

Network-Specific Transport Protocols T.123

User Application(s) - Using Standard and/or Non-Standard Application Protocols

File Transfer - T.127 . . .

Still Image - T.126

ITU-T Standard Node

Controller

. . .

Non-Standard

T.120 System Model

(46)

H.221: Multiplexing/De-multiplexing



ITU standard for videotelephony framing.

 Aimed primarily at ISDN (64 or 128kbit/s, but can go up to 1920kbit/s).



Almost outdated now.

 First standardized in 1988, but revised several times since.

 ISDN isn’t so popular anymore.



It describes how audio, video and data streams are multiplexed together for transmission over networks.



Based on the concept of TDM.



It ensures each stream is placed into its allocated position in output stream.

Framing ISDN Channel

H.221 Framing

CRC Audio Data H.261 Video

H.261 Video CRC

Audio H221 ISDN Channel Data

(47)

Packet Switched networks

•

Voice over IP (VoIP)

•

H.323 standard

(48)

Gateways allow PCs to also reach phones Public Switched

Telephone Network

PSTN (Country B)

Initially, PC to PC voice calls over the

Internet Gateway

Multimedia PC

IP Network

Gateway

Multimedia PC

PSTN

(Country A) …or phones to reach phones

What’s VoIP?

(49)

Original data stream: 10011…01 01001…11 … … … … … 10100…10

…

1^st block 2^nd block N^th block

1^st packet 2^nd packet

…

N^th packet

Maximum 64K Bytes

20 ~ 60 Bytes

Internet Packet Ethernet Packet

Header Data Payload

C-data 10100…10

C-data 01001…11

C-data 10011…01

10100…10 01001…11

10011…01

The data transmission method in computer communication is conceptually similar as the postal system. A large data stream will be divided into relatively small blocks, called packet, before transmission. Each packet is transmitted individually and

independently over networks  Packet-based Communication/Network

Packet-based Network (IP Network)

(50)

play no-continuously samples/frames ^Network

Temporal Relations in Video and Audio

(51)

• Internet telephony, also called Voice over IP (VoIP), refers to using the IP network infrastructure (LAN, WLAN, WAN, Internet) for voice communication.

IP (Internet Protocol) transmission unit: packet First product appeared in February of 1995:

•

– –

Internet Phone Software by Vocaltec, Inc., “free” long distance call via PC Software compressed the voice and sent it as IP packets.

• Other software/products soon followed  NetMeeting, Skype, Gphone, …

Delay & jitter

VoIP Basic Features and History

(52)

Internet

• Issues:

– – – – –

Addressing, i.e., VoIP phone number

Call admission, setup, control, release, etc

IP network related: delay, jitter, packet loss, out-of-order Transmission overhead: Headers

.. ..

Small delay

 Small packet size Voice data

Total > 100 bytes Can’t be large for voice delay

Voice data rate: 1~8KBytes/Second

or 8~64Kbps (bits-per-second)

RTP Header UDP Header IP Header

Scenario 1: PC to PC

(53)

SIP Signaling SS7 Signaling

Phone

Network IP

Network

Gateway

PCM Coding G.72x/MPEG

• A Gateway is

network: needed to connect the PSTN to the IP

– Signaling conversion – Format conversion

Scenario 2: PC to Phone

(54)

Phone IP Phone

Network Network Network

Gateway Gateway

• Gateways will connect the phone network to the network.

The IP Network can be a dedicated backbone or IP

• intranet (to provide guaranteed QoS) or can be the Internet (no guarantees …)

The phone network can be a company PBX

(Private Branch Exchange)

or carrier switches

• Scenario 3: Phone to Phone

(55)

Internet

Conference Chair

Internet teleconference: A group of people communicate each other via voice, video and/or other data over the Internet

- - - -

Conference initiation, start, join, leave, end, control, etc.

Sending audio/video data from one-to-many (multicast)

Sharing other conference data (data conferencing) among all participants Synchronization and network delay, jitter, packet loss, …

Internet Teleconference

(56)

ISDN

NetMeeting

Example of Audiovisual Conference

(57)

Data conferencing is a virtual connection between two or more computers where:

• All computers in the conference display a common graphical image of text, graphics or a combination of both.

Each computer in the conference displays any changes to the common image in near real time.

Participants have ability to interact with the displayed document WYSIWIS: What You See Is What I See

•

Presentation (group broadcast)

– Broadcast event where a single presenter’s electronic presentation is distributed to multiple remote computers.

Collaboration (group meeting)

– – –

Everyone can talk, operate, …

Usually involves a small conference of 3-10 participants

Two types of Collaboration: Whiteboarding & Application Sharing

What is Data Conferencing?

(58)

•

• Self-developed communication software/middleware Implementations of Internet telephony and

conference can use two types of popular standards 1

^st

- H.323 standards from ITU (1996, Version)

*

Adopt some protocols (RTP/RTCP) from IETF More implementations

Very complex

Poor interoperability between vendors

1

^st

- SIP standards from IETF (1998, Version)

*

Session Initiation Protocol (SIP) Similar functions as H.323

Relatively easy because of textual natural instead of Better interoperability

binary

Typical Standards: H.323 & SIP

(59)

• H.323 is a product of ITU-T Study Group 16.

Version 1: “visual telephone systems

•

and equipment for LANs that provide a nonguaranteed quality of service (QoS)” was accepted in October 1996.

– Focus on multimedia communication in a LAN

No support for guaranteed QoS –

• Version 2: “packet-based multimedia communications systems” was driven by the Voice-over-IP requirements and was accepted in January 1998.

Version 3 was accepted in September

• 1999 and has minor incremental

features (caller ID, …) over version 2.

Version 4 was accepted in November

• 2000 and has significant

improvements over version 3.

H.323 History

(60)

H.323 Entities: Terminal, Gatekeeper, Gateway, MCU (Multipoint Control Unit)

Guaranteed

QoS LAN

PSTN N-ISDN B-ISDN

- H.310 (B-ISDN) - H.320 (N-ISDN) - H.321 (ATM)

- H.322 (GQOS-LAN)

- H.324 (GSTN), H.324/M (mobile phone, 1998) - V.70 (DSVD - Digital Simultaneous Voice & Data)

H.321 Terminal H.320

Terminal Speech

Terminal H.322

Terminal Speech

Terminal H.324

Terminal V.70

Terminal

H.321 Terminal

H.323 Terminal

H.323 MCU

Non guaranteed QoS LAN

H.323 Gatekeeper

H.323 Gateway

H.323 Terminal

H.323 System

(61)

• Terminal

– An endpoint on the LAN which provides for real-time, two-way communications with another H.323 terminal, Gateway, or MCU – May provide audio, video, and/or data

Gatekeeper

– Provides address translation and controls access to the LAN – Performs bandwidth management

Multipoint Control Unit (MCU)

– Provides the capability for 3 or more terminals and Gateways to participate in a multipoint conference

Gateway

– Provides for real-time, two-way communication between H.323

terminals on a LAN and other ITU terminals on a wide-area network or another H.323 Gateway

•

H.323 Entities

(62)

H.323 Protocol Stack

H.323 Gateway

RAS: Registration, Admission, Status

AV App Terminal Control and Management Data App

Other Stacks

H.225.0 Stack

G.72X H.26x

RTCP

H.225.0 Terminal to Gatekeeper Signaling

(RAS)

H.225.0 Call

Signaling H.245 Q.931

T.124

T.125 RTP

LANUnreliable Transport (UDP) Reliable Transport (TCP)

T.123 Network Layer

Link Layer Physical Layer

H.323 Protocol Stack

(63)

Scope of Recommendation H.323

G.711, G.722

NETWORK

H.225.0 VIDEO CODEC

H.261, H.263

RECEIVE PATH DELAY

H.225 LAYER

LOCAL AREA INTERFACE VIDEO I/O EQUIPMENT

AUDIO CODEC G.723, G.728

G.729 AUDIO I/O EQUIPMENT

USER DATA APPLICATIONS T.120, etc

SYSTEM CONTROL H.245 CONTROL

SYSTEM CONTROL USER INTERFACE

CALL CONTROL

RAS CONTROL H.225.0

H.323 Terminal

(64)

• Provides the following services:

– Address translation between Transport Addresses and Alias Addresses

# Transport Addresses: LAN IP Address + TSAP Identifier (port number)

# Alias Addresses: phone number, user name, email address, etc.

Admission control based on authorization, bandwidth, or other criteria Dynamic bandwidth control during a conference

– –

• Transport address for the H.245

Call Signaling Channel Control Channel is exchanged on the

H.225/RAS messages

over RAS channel H.225/RAS messages

over RAS channel

H.225/Q.931 (optional) Gatekeeper H.225/Q.931 (optional) H.245 messages (optional) H.245 messages (optional)

H.225/Q.931 messages over call signaling channel

PSTN

H.245 messages over call control channel

Gateway Terminal

Gatekeeper

(65)

Tokyo London

Gatekeeper

ports New York

MC: Multipoint Controller, MP: Multipoint Processor

Conf B Conf A

• MC performs capability exchanges with each endpoint and determines the media format used in a conference

- Assigns terminal numbers to each endpoint in the conference - Maintains a list of all conference participants

MP is used for processing of audio/video/data streams in a

centralized or hybrid multipoint conference MCU

•

Note: - MC/MP may be co-located with a Gateway or Gatekeeper - Gateway, Gatekeeper and MCU may be a single device

3 Terminal 1

MC

Terminal 2 Gatekeeper

MC 1 Gatekeeper

MC 2 _MP Gatekeeper

LAN

MC

Gateway 1

MC MP

Gateway 2 Gateway 3

MC MP

MCU 1

MC

MCU 2

MC MP audio video T.120 MCS

MCU

Multipoint Entities & MCU

^MCU _MCU

(66)

RAS Q.931/

H.245 Signaling

Q.931/

H.245 RAS (Q.931)

Gatekeeper Routed Signaling Direct Routed Signaling

Terminal H.245 Terminal

RTP/RTCP Gatekeeper

Annex G

Gatekeeper Q.931/H.245

H.323 Basic Protocols for VoIP

(67)

• Step 1: Endpoint - Gatekeeper communication

RAS Channel H.225

RAS Channel H.225 MCU

- Gatekeeper discover

- Registration/Unregistration - Location Request

(Alias/Transport address lookup) - Admission control

- Bandwidth changes - Status Request MC

Audio MP Video MP T.120 MCS

Terminal B Terminal A

Gatekeeper

H.323 VoIP Call Setup Procedures (1)

(68)

• Step 2: Setup initial connection with the MCU using the Call Signaling Channel via gatekeeper

RAS Channel RAS Channel

Call Signaling Call Signaling

H.225 H.225 MCU

MC

Gatekeeper

H.323 VoIP Call Setup Procedures (2)

(69)

• Step 3: Setup H.245 Control Channel with the MCU

Call Signaling MCU Call Signaling

H.245 Control H.245 Control

• All endpoints transmit a Terminal Capability Set

• Transport address for the H.245 Control Channel is exchanged on the Call Signaling Channel Used to exchange

capabilities, create logical channels, and exchange multipoint commands

– List of all audio, video, and data capabilities supported by the endpoint

• • MCU receives the

capabilities and determines the Selected Communication Mode (SCM)

MC

Gatekeeper

H.323 VoIP Call Setup Procedures (3)

(70)

Step 4: Setup additional logical channels for audio/video/data

Call Signaling MCU Call Signaling

Terminal A Terminal B

H.245 Control MC RTP/RTCP

H.245 Control RTP/RTCP Audio MP

RTP/RTCP RTP/RTCP

Video MP

T.123 T.123

T.120 MCS Gatekeeper

H.323 VoIP Call Setup Procedures (4)

(71)

• The Session Initiation Protocol (SIP, RFC 2543) has been proposed as an alternative to H.323

SIP is capable of negotiating a call

SDP is used to describe capabilities: media, coding, protocol, address/port, crypto key Media still runs over RTP

Each has merits and demerits, but quite similar

•

IP

Call Control and Signaling Signaling and Gateway Control

Media

Audio/

Video H.323

H.225

H.245 Q.931 RAS SIP/SDP MGCP RTP RTCP RTSP

TCP UDP