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

The DESCRIBE method retrieves the description of a presentation

DESCRIBE

ANNOUNCE

A client can issue a SETUP request for a stream that is already playing to change transport parameters
A client can issue a SETUP request for a stream that is already playing to change transport parameters The PLAY method tells the server to start sending data via the mechanism specified in SETUP

SETUP,

PLAY,

RTSP Method types

PAUSE

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

TEARDOWN

GET_PARAMETER, SET_PARAMETER

REDIRECT

RECORD

RTSP Client FSM

state Init message sent next state after response SETUP Ready TEARDOWN Init Ready PLAY 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 Init message received next state 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 object 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)

Another FEC Example

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

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/course s/CMSC691C/lab5-kurose-ross.html