unit-iv: networked multimediatraffic characterization parameters due to the variability of the bit...

UNIT-IV:

NETWORKED MULTIMEDIA

EL-447: Multimedia Systems & Networks 1

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)


Introduction (2)

Since different networks operate in different way,

there are number of standards each intended for

use with specific type of networks.


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


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).


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

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


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.


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


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


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

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 (~2x108 m/sec)

propagation delay = d/s

A

B

propagation

transmission

nodal processing queueing

Note: s and R are very different

quantities!

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


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.


• 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

HTTP

Web Server

Long start-up latency Potential waste of traffic

AV

File

Web

Browser

Media

Player

Non Real time Delivery: Downloading


HTTP

file

Web Server

RTSP/MMS/HTTP

RTP/RTCP

Streaming Server

AV

File

meta

Web

Browser

Media

Player

Real-time Delivery: Streaming


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

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

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

Delay and Jitter


Key Points in Streaming Media Service

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 network

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

time client playout delay

buf

fere

d

video

rate

without

Questions:

client video reception

-

rate video transmission

delay


Key Points in Streaming Media Service (Cont)

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?


Unicast Multicast



Network

Unicast Example: Multiple Independent Streams


Servers Intermediaries Clients

Multicast Example: Single Stream and Copy


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



Capture Encoding Serving Internet distribution Playback

Media

Player

Source

Encoder Media IP network Server Media

Player

Media

Proxy

Proxy Media Server


Reduce network traffic

Reduce response

time to client

Reduce server’s load

Server Intermediary Client

Proxy Server: Reduce Traffic, Time, Load


Media Streaming Service Access Process


HTTP (Control and Data)

RTSP/TCP (Control)

RTP/UDP (Media Data)

RTCP/UDP (RTP Control)

Sch

ed

ule

r

Media

Player

Media Server

RTSP

Handler

RTP

Handler

File Media

Parsing Storage

Web Browser

Web Server

HTTP HTML

Handler Files

Media Streaming Service Modules


TCP (till now)

RTP RTCP

RTSP

Protocol Stack for Multimedia Services


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?


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


Default port 554

RTSP SETUP

RTSP OK

RTSP PLAY

RTSP OK

RTSP TEARDOWN

RTSP OK

TCP

choose UDP port

RTP VIDEO

UDP RTP AUDIO

RTCP

RTSP

client U

AV subsystem

media player

RTSP

server

get U DP port

data source

media server

RTSP Session


•

•

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?


• 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

Standards Relating to

Interpersonal Communication


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.


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

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


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

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.

• 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

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

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

Packet Switched networks

• Voice over IP (VoIP)

• H.323 standard


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?


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

… 1st 2nd Nth block block block

… 1st 2nd Nth packet packet 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)

play no-continuously

samples/frames

Network


Temporal Relations in Video and Audio

• 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


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

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


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


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


ISDN

NetMeeting

Example of Audiovisual Conference


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?


•

•

Self-developed communication software/middleware

Implementations of Internet telephony and

conference can use two types of popular standards

1st - H.323 standards from ITU (1996, Version)

*

*

*

*

Adopt some protocols (RTP/RTCP) from IETF

More implementations

Very complex

Poor interoperability between vendors

1st - 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


• 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


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

Terminal

H.323 System


• 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


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

LA

N

Unreliable Transport (UDP)

Reliable Transport (TCP)

T.123

Network Layer

Link Layer

Physical Layer


H.323 Protocol Stack

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


• 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) H.225/Q.931 (optional) Gatekeeper

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


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


RAS Q.931/

H.245

Signaling

Q.931/

H.245

RAS

(Q.931)

Gatekeeper Routed Signaling

Direct Routed Signaling

Terminal

Terminal

H.245

RTP/RTCP

Gatekeeper

Annex G

Gatekeeper

Q.931/H.245

H.323 Basic Protocols for VoIP


• 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)


• 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

MCU H.225

MC

Audio MP

Video MP

T.120 MCS

Terminal B

Terminal A

Gatekeeper



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


MCU Call Signaling 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

Audio MP

Video MP

T.120 MCS

Terminal B

Terminal A

Gatekeeper



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


MCU Call Signaling Call Signaling

Terminal A

Terminal B

MC H.245 Control

RTP/RTCP

H.245 Control

RTP/RTCP Audio MP

RTP/RTCP

RTP/RTCP Video MP

T.123

T.123 T.120 MCS

Gatekeeper



• 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

Alternative: SIP/SDP


unit-iv: networked multimediatraffic characterization parameters due to the variability of the bit...

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