RTSP Real Time Streaming Protocol

Presented by Group NEO -Jitin Keith -Prashant Morgaonkar

RTSP

Real Time Streaming Protocol IETF RFC 2326

RTSP

RTSP is a signaling protocol which acts as a “Remote Control” for controlling a IP media server. Very similar to HTTP /1.1

RTSP can request    Server to send media Server invited to join a conference Server announces that additional media available

Overall Function

    A presentation description file defines the type of media streams and its properties such as encoding language etc.

A presentation may contain several media streams RTSP can use TCP, UDP or MUDP Popular RTSP clients: Windows Media Player Real Player

RTSP Typical Method Types

    Setup: Causes the server to allocate resources for a stream and start an RTSP session Play-Record: Starts data transmission on a stream allocated via SETUP Pause: Temporarily halts a stream without freeing server resources. Teardown: Frees resources associated with the stream. The RTSP session ceases to exist on the server.

RTSP Status Codes

 Success 2XX  Redirect 3XX  Client Error 4XX

RTSP Method types

     OPTIONS  If the client is about to try a nonstandard request DESCRIBE  The DESCRIBE method retrieves the description of a presentation ANNOUNCE  A client can issue a SETUP request for a stream that is already playing to change transport parameters SETUP,  A client can issue a SETUP request for a stream that is already playing to change transport parameters PLAY,  The PLAY method tells the server to start sending data via the mechanism specified in SETUP

RTSP Method types

     PAUSE  The PAUSE request causes the stream delivery to be interrupted (halted) temporarily TEARDOWN  The TEARDOWN request stops the stream GET_PARAMETER, SET_PARAMETER  The GET_PARAMETER request retrieves the value of a parameter of a presentation or stream REDIRECT  A redirect request informs the client that it must connect to another server location RECORD  Start recording the media

RTSP Client FSM

state message sent next state after response Init SETUP Ready PLAY Ready TEARDOWN Init Playing RECORD Recording TEARDOWN Init SETUP Ready Playing PAUSE Ready TEARDOWN Init PLAY Playing SETUP Playing (changed transport) Recording PAUSE Ready TEARDOWN Init RECORD Recording SETUP Recording (changed transport)

RTSP Server FSM

state message received next state Init SETUP Ready TEARDOWN Init Ready PLAY Playing SETUP Ready TEARDOWN Init RECORD Recording Playing PLAY Playing PAUSE Ready TEARDOWN Init SETUP Playing Recording RECORD Recording PAUSE Ready TEARDOWN Init SETUP Recording The "next state" column indicates the state assumed after sending a success response (2xx). If a request results in a status code of 3xx, the state becomes Init. A status code of 4xx results in no change

Security issues

 Denial of service: Flood any arbitrary machine on the Internet with unrequested RTSP streams  Session hijacking  Authentication

Streaming Stored Multimedia (1/2)

Streaming Stored Multimedia (2/2)

Stored media streaming

 Internet multimedia: simplest approach  Internet multimedia: simplest approach  files transferred as HTTP  received in entirety at client  then passed to player  audio, video not streamed:  no, “pipelining,” long delays until playout!

Streaming vs. Download of Stored Multimedia

Download: Receive entire content before playback begins High “start-up” delay as media file can be large ~ 4GB for a 2 hour MPEG II movie Streaming: Play the media file while it is being received Reasonable “start-up” delays Reception Rate >= playback rate.

Progressive Download

browser GETs metafile browser launches player, passing metafile player contacts server server downloads audio/video to player

RTSP Example

 Scenario: metafile communicated to web browser browser launches player player sets up an RTSP control connection, data connection to streaming server

RTSP Operation

Client Buffering

Client-side buffering, playout delay compensate for network-added delay, delay jitter

Client Buffering

Client-side buffering, playout delay compensate for network-added delay, delay jitter

Streaming Multimedia: UDP or TCP?

  UDP server sends at rate appropriate for client (oblivious to network congestion !) often send rate = encoding rate = constant rate then, fill rate = constant rate - packet loss short playout delay (2-5 seconds) to compensate for network delay jitter error recover: time permitting TCP send at maximum possible rate under TCP fill rate fluctuates due to TCP congestion control larger playout delay: smooth TCP delivery rate HTTP/TCP passes more easily through firewalls

Packet Loss

 network loss: IP datagram lost due to network congestion (router buffer overflow)  delay loss: IP datagram arrives too late for playout at receiver

Forward Error Correction (FEC)

For every group of n packets generate k redundant packets Send out n+k packets, increasing the bandwidth by factor k/n.

Can reconstruct the original n packets provided at most k Packets are lost from the group Works well at high loss rate (for a proper choice of k) Handles “bursty” packet losses Cost: increase in transmission cost (bandwidth)

• “Piggyback lower quality stream” • Example: send lower resolution audio stream as the redundant information

Another FEC Example

Whenever there is non-consecutive loss, thereceiver can conceal the loss.

Can also append (n-1)st and (n-2)nd low-bit ratechun

Interleaving

Interleaving Re-sequence packets before transmission Better handling of “burst” losses Results in increased playout delay

Multimedia Networking

 Exciting, industry relevant research topic  Multimedia is everywhere  Tons of open problems

References

[1] www.wikipedia.com

[2] http://www.ietf.org/rfc/rfc2326.txt

[3] http://tesys.cres.it/jsp/activities/rtsp/rtsp.jsp

[4] http://www.csee.umbc.edu/~pmundur/courses /CMSC691C/lab5-kurose-ross.html