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H.323 and Real Time Streaming H.323 and Real Time Streaming ProtocolProtocol

H.323 and Real Time Streaming H.323 and Real Time Streaming ProtocolProtocol

Dorgham SisalemMobile Integrated Services

FhG Fokus

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H.323 PrimerH.323 PrimerH.323 PrimerH.323 Primer

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H.323H.323H.323H.323

• May 1995 - H.323 work started

• June 1996 - Decided by ITU-T

• Designed for exchanging multimedia over IP networks (4 versions of H.323)

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H.323 ComponentsH.323 ComponentsH.323 ComponentsH.323 Components

• Terminals• Gatekeepers• Gateways (H.323 to H.320/H.324/POTS)• MCUs

– Multipoint Controller (MC)– Multipoint Processor (MP)

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H.323 GatekeeperH.323 GatekeeperH.323 GatekeeperH.323 Gatekeeper

• Address Translation– H.323 Alias to transport (IP) address based on

terminal registration– “email-like” names possible– “phone number like” names possible

• Admission control– Permission to complete call– Can apply bandwidth limits– Method to control LAN traffic

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H.323 Gatekeeper (cont.)H.323 Gatekeeper (cont.)H.323 Gatekeeper (cont.)H.323 Gatekeeper (cont.)

• Management of gateway – Which gateway to choose (load balancing and cost

regulation)

• Call Signaling– May route calls in order to provide supplementary

services or to provide Multipoint Controller functionality

• Call Management/Reporting/Logging

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H.323 GatewayH.323 GatewayH.323 GatewayH.323 Gateway

• Provide world wide connectivity and interoperability from LAN– H.320, H.324, regular POTS telephones

• Map Call Signaling (Q.931 to H.225.0)• Map Control (H.242/H.243 to H.245)• Media Mapping (FEC, multiplex, rate

matching, audio transcoding)

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H.323 MCUH.323 MCUH.323 MCUH.323 MCU

• Media Distribution– Unicast - send media to

one terminal (centralized in MP; traditional model)

– Multicast - send to each receiver directly

– Hybrid - some of each

• Manage Ad Hoc multipoint calls– Join, invite, control of

conference modes

• Multiprotocol via Gateways

PictureTel

PictureTel

PictureTel

PictureTel

PictureTel

PictureTel

MCU

PictureTel

PictureTel

ISDN (H.320)

Gateway)

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H.323 protocol stackH.323 protocol stackH.323 protocol stackH.323 protocol stack

H.225.0 H.245

G.7xx H.26x

RTP

RTCP

Gate-keeper

Reg,Adm,Status(RAS)

Control Data Audio Video A/V Cntl Control

TCP UDP

IP

T.120

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H.323 protocol stack H.323 protocol stack (cont.)(cont.)

H.323 protocol stack H.323 protocol stack (cont.)(cont.)

• H.323 - System Document• H.225.0 - Call Signaling, Packetization

– Gatekeeper Registration, Admission, and Status (RAS)

• H.245 - Control (also used in H.324, H.310)• T.120 - Data and Conference Control

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H.323 exampleH.323 exampleH.323 exampleH.323 example

A Call Setup Example

• a point to point call

• One Gatekeeper using the Direct Call Model

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H.323 example (cont.)H.323 example (cont.)H.323 example (cont.)H.323 example (cont.)

PictureTel

PictureTel

PictureTel

(1) ARQCan I call“Bob”?

(2a) GK resolves “Bob”to IP address through H.323registration or externalname service (e.g. DNS, Database, etc..)(2b) Admission PolicyApplied(3) ACF

Yes, use thisIP Address

Bill Bob

GK

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H.323 example (cont.)H.323 example (cont.)H.323 example (cont.)H.323 example (cont.)

PictureTel

PictureTel

PictureTel

(4) SETUP (Create)

Bill Bob

GK

(5) ARQMay I answer?

(6) ACFYes

(7) ALERTING

(8) CONNECT (User answers)

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H.323 example (cont.)H.323 example (cont.)H.323 example (cont.)H.323 example (cont.)

PictureTel

PictureTel

PictureTel

Bill Bob

(9) H.245 connection established

- Capability Exchange

- Open Logical Channels (audio, video)

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Multimedia StreamingMultimedia StreamingMultimedia StreamingMultimedia Streaming

Based on slides from Vishal Misra,

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Streaming Stored Streaming Stored Audio & VideoAudio & Video

Streaming Stored Streaming Stored Audio & VideoAudio & Video

Streaming stored media:• Audio/video file is stored in a

server• Users request audio/video file

on demand.• Audio/video is rendered

within, say, 10 s after request.

• Interactivity (pause, re-positioning, etc.) is allowed.

Media player:– removes jitter– decompresses– error correction– graphical user

interface with controls for interactivity

• Plug-ins may be used to imbed the media player into the browser window.

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Streaming from Web server Streaming from Web server (1)(1)

Streaming from Web server Streaming from Web server (1)(1)

• Audio and video files stored in Web servers

naïve approach• browser requests file with

HTTP request message• Web server sends file in HTTP

response message• content-type header line

indicates an audio/video encoding

• browser launches media player, and passes file to media player

• media player renders file

• Major drawback: media playerinteracts with server throughintermediary of a Web browser

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Streaming from Web Streaming from Web server (2)server (2)

Streaming from Web Streaming from Web server (2)server (2)

Alternative: set up connection between server and player

• Web browser requests and receives a meta file

(a file describing the object) instead of receiving the file itself;

• Content-type header indicates specific audio/video application

• Browser launches media player and passes it the meta file

• Player sets up a TCP connection with server and sends HTTP request.

Some concerns:• Media player

communicates over HTTP, which is not designed with pause, fast forward, rewind commands

• May want to stream over UDP

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Streaming from a Streaming from a streaming serverstreaming serverStreaming from a Streaming from a streaming serverstreaming server

• This architecture allows for non-HTTP protocol between server and media player

• Can also use UDP instead of TCP.

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Real Time Streaming Real Time Streaming Protocol: Protocol: Why not HTTPWhy not HTTP

Real Time Streaming Real Time Streaming Protocol: Protocol: Why not HTTPWhy not HTTP

• Designers of HTTP had fixed media in mind: HTML, images, applets, etc.

• HTTP does not target stored continuous media (i.e., audio, video, SMIL presentations, etc.)

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Options when using a Options when using a streaming serverstreaming server

Options when using a Options when using a streaming serverstreaming server

• Send at constant rate over UDP. To mitigate the effects of jitter, buffer and delay playback for 1-10 s. Transmit rate = d, the encoded rate. Fill rate x(t) equals d except when there is loss.

• Use TCP, and send at maximum possible rate under TCP; TCP retransmits when error is encountered; x(t) now fluctuates, and can become much larger than d. Player can use a much larger buffer to smooth delivery rate of TCP.

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Streaming: The Streaming: The RealAudio WayRealAudio WayStreaming: The Streaming: The RealAudio WayRealAudio Way

• .ram files describe the location of the filertsp://bazzooka.de/videos/

NationalhymneVideo.rmuse RTSP for streaming the file

http://bazzooka.de/videos/NationalhymneVideo.rmuse RTSP over HTTP for streaming the media

• .rm, .ra files describe the content of the file– Which bandwidth, compression, timinglines ....

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Meta file exampleMeta file example (SDP) (SDP)Meta file exampleMeta file example (SDP) (SDP)

v=0o=- 2890844526 2890842807 IN IP4 192.16.24.202s=RTSP Sessionm=audio 0 RTP/AVP 0a=control:rtsp://audio.example.com/twister/audio.enm=video 0 RTP/AVP 31a=control:rtsp://video.example.com/twister/video

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Real Time Streaming Real Time Streaming ProtocolProtocol

RTSP (RFC 2326)RTSP (RFC 2326)

Real Time Streaming Real Time Streaming ProtocolProtocol

RTSP (RFC 2326)RTSP (RFC 2326)

• Client-server application layer protocol.• acts as a « network remote control »• supports the following operations:

– retrieval of a media from a server– invitation of a media server to a conference– recording of a conference

What it doesn’t do:• does not define how audio/video is encapsulated for

streaming over network• does not restrict how streamed media is transported; it can

be transported over UDP or TCP• does not specify how the media player buffers audio/videoRealNetworks• Server and player use RTSP to send control info to each other

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RTSP: out of band controlRTSP: out of band controlRTSP: out of band controlRTSP: out of band control

• A file is transferred over one channel.– The RTSP control messages use different port numbers

than the media stream, and are therefore sent out-of-band.

• Interleaved:– If the RTSP messages were to use the same port numbers

as the media stream, then RTSP messages would be said to be “interleaved” with the media stream.

• What is this good for?

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RTSP generalitiesRTSP generalitiesRTSP generalitiesRTSP generalities

• Protocol design– text-based protocol– transport protocol independant– supports any session description (sdp, xml, etc.)– similar design as HTTP with differences yet!

• client server and server client requests• server maintains a « session state »• data carried out-of-band

– works either with unicast or multicast

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RTSP messagesRTSP messagesRTSP messagesRTSP messages

• a request (client server or server client)

PLAY rtsp://video.example.com/twister/video RTSP/1.0 CSeq: 2 Session: 23456789 Range: smpte=0:10:00-

• and its response

RTSP/1.0 200 OK CSeq: 2 Session: 23456789 Range: smpte=0:10:00-0:20:00 RTP-Info: url=rtsp://video.example.com/twister/video; seq=12312232;rtptime=78712811

seq# for request/response pair

session identifier

method to apply URL RTSP version

play starting at that offset for anundefined duration

status codeversion

seq# for request/response pair

session identifier

reason phrase

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RTSP methodsRTSP methodsRTSP methodsRTSP methods

• major methods– SETUP: server allocates resources for a

stream and starts an RTSP session– PLAY: starts data tx on a stream– PAUSE: temporarily halts a stream– TEARDOWN: free resources of the stream, no

RTSP session on server any more

• additional methods– OPTIONS: get available methods– ANNOUNCE: change description of media object– DESCRIBE: get low level descr. of media object– RECORD: server starts recording a stream– REDIRECT: redirect client to new server– SET_PARAMETER: device or encoding control

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RTSP initiates and RTSP initiates and controls deliverycontrols delivery

RTSP initiates and RTSP initiates and controls deliverycontrols delivery• Client obtains a description of the

multimedia presentation, which can consist of several media streams..

clientC

web server

W

mediaserversA & V

HTTP GET

presentation description (sdp)

SETUP

PLAY

RTP audio/video

RTCP

TEARDOWN

• The browser invokes media player (helper application) based on the content type of the presentation description.• Presentation description includes references to media streams, using the URL method rtsp://

• Player sends RTSP SETUP request; server sends RTSP SETUP response.

• Player sends RTSP PLAY request; server sends RTSP PLAY response.

• Media server pumps media stream.

• Player sends RTSP TEARDOWN request; server sends RTSP TEARDOWN response

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Example: media on Example: media on demand, unicastdemand, unicast

Example: media on Example: media on demand, unicastdemand, unicast

• Scenario…videoserver

audioserver

media descr.web serverclient

step 1: get description (in SDP format)

step 2: open streams with RTSP

step 3: play

step 4: teardown

CW A V

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RTSP sessionRTSP sessionRTSP sessionRTSP session

• Each RTSP has a session identifier, which is chosen by the server.

• The client initiates the session with the SETUP request, and the server responds to the request with an identifier.

• The client repeats the session identifier for each request, until the client closes the session with the TEARDOWN request.

• RTSP port number is 554.• RTSP can be sent over UDP

or TCP. Each RTSP message can be sent over a separate TCP connection.

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RTSP: streaming cachingRTSP: streaming cachingRTSP: streaming cachingRTSP: streaming caching

• Caching of RTSP response messages makes little sense.

• But desirable to cache media streams closer to client.

• Much of HTTP/1.1 cache control has been adopted by RTSP.– Cache control headers can be put in RTSP SETUP

requests and responses:• If-modified-since: , Expires: , Via: , Cache-

Control:

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RTSP: streaming cachingRTSP: streaming cachingRTSP: streaming cachingRTSP: streaming caching

• Proxy cache may hold only segments of a given media stream.– Proxy cache may start serving a client from its

local cache, and then have to connect to origin server and fill missing material, hopefully without introducing gaps at client.