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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)
EL-447: Multimedia Systems & Networks 2
Introduction (2)
Since different networks operate in different way,
there are number of standards each intended for
use with specific type of networks.
EL-447: Multimedia Systems & Networks 3
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
EL-447: Multimedia Systems & Networks 4
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).
EL-447: Multimedia Systems & Networks 5
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
EL-447: Multimedia Systems & Networks 7
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.
EL-447: Multimedia Systems & Networks 8
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
EL-447: Multimedia Systems & Networks 9
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
EL-447: Multimedia Systems & Networks 10
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
EL-447: Multimedia Systems & Networks 13
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.
EL-447: Multimedia Systems & Networks 14
• 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
16 EL-447: Multimedia Systems & Networks
HTTP
file
Web Server
RTSP/MMS/HTTP
RTP/RTCP
Streaming Server
AV
File
meta
Web
Browser
Media
Player
Real-time Delivery: Streaming
17 EL-447: Multimedia Systems & Networks
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
20 EL-447: Multimedia Systems & Networks
Popular Stream Media Server and Player
Delay and Jitter
21 EL-447: Multimedia Systems & Networks
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
22 EL-447: Multimedia Systems & Networks
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?
23 EL-447: Multimedia Systems & Networks
Unicast Multicast
Key Points in Streaming Media Service (Cont)
24 EL-447: Multimedia Systems & Networks
Network
Unicast Example: Multiple Independent Streams
25 EL-447: Multimedia Systems & Networks
Servers Intermediaries Clients
Multicast Example: Single Stream and Copy
26 EL-447: Multimedia Systems & Networks
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)
27 EL-447: Multimedia Systems & Networks
Capture Encoding Serving Internet distribution Playback
Media
Player
Source
Encoder Media IP network Server Media
Player
Media
Proxy
Proxy Media Server
28 EL-447: Multimedia Systems & Networks
Reduce network traffic
Reduce response
time to client
Reduce server’s load
Server Intermediary Client
Proxy Server: Reduce Traffic, Time, Load
29 EL-447: Multimedia Systems & Networks
Media Streaming Service Access Process
30 EL-447: Multimedia Systems & Networks
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
31 EL-447: Multimedia Systems & Networks
TCP (till now)
RTP RTCP
RTSP
Protocol Stack for Multimedia Services
32 EL-447: Multimedia Systems & Networks
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?
33 EL-447: Multimedia Systems & Networks
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
34 EL-447: Multimedia Systems & Networks
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
35 EL-447: Multimedia Systems & Networks
•
•
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?
36 EL-447: Multimedia Systems & Networks
• 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
EL-447: Multimedia Systems & Networks 38
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.
EL-447: Multimedia Systems & Networks 39
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
EL-447: Multimedia Systems & Networks 41
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.)
EL-447: Multimedia Systems & Networks 43
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)
44 EL-447: Multimedia Systems & Networks
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
45 EL-447: Multimedia Systems & Networks
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.
EL-447: Multimedia Systems & Networks 46
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
EL-447: Multimedia Systems & Networks 47
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?
48 EL-447: Multimedia Systems & Networks
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
49 EL-447: Multimedia Systems & Networks
Packet-based Network (IP Network)
play no-continuously
samples/frames
Network
50 EL-447: Multimedia Systems & Networks
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
51 EL-447: Multimedia Systems & Networks
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
52 EL-447: Multimedia Systems & Networks
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
53 EL-447: Multimedia Systems & Networks
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
54 EL-447: Multimedia Systems & Networks
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
55 EL-447: Multimedia Systems & Networks
ISDN
NetMeeting
Example of Audiovisual Conference
56 EL-447: Multimedia Systems & Networks
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?
57 EL-447: Multimedia Systems & Networks
•
•
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
58 EL-447: Multimedia Systems & Networks
• 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
59 EL-447: Multimedia Systems & Networks
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
60 EL-447: Multimedia Systems & Networks
• 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
61 EL-447: Multimedia Systems & Networks
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
62 EL-447: Multimedia Systems & Networks
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
63 EL-447: Multimedia Systems & Networks
• 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
64 EL-447: Multimedia Systems & Networks
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
65 EL-447: Multimedia Systems & Networks
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
66 EL-447: Multimedia Systems & Networks
• 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)
67 EL-447: Multimedia Systems & Networks
• 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
H.323 VoIP Call Setup Procedures (2)
68 EL-447: Multimedia Systems & Networks
• Step 3: Setup H.245 Control Channel with the MCU
RAS Channel RAS Channel
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
H.323 VoIP Call Setup Procedures (3)
69 EL-447: Multimedia Systems & Networks
Step 4: Setup additional logical channels for audio/video/data
RAS Channel RAS Channel
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
H.323 VoIP Call Setup Procedures (4)
70 EL-447: Multimedia Systems & Networks
• 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
71 EL-447: Multimedia Systems & Networks