Real-Time Streaming Protocol-Computer Networking and Multimedia Applications-Lecture Slides, Slides for Multimedia Applications. Ankit Institute of Technology and Science
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paridhi24 July 2012

Real-Time Streaming Protocol-Computer Networking and Multimedia Applications-Lecture Slides, Slides for Multimedia Applications. Ankit Institute of Technology and Science

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Prof. Shailesh Rao delivered this lecture at Ankit Institute of Technology and Science for Computer Networking and Multimedia Applications course. It includes: Real-time, Streaming, Protocol, Multimedia, Content, Client,...
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26

Real-Time Streaming Protocol (RTSP)

HTTP

 Does not target multimedia content

 No commands for fast forward, etc.

RTSP: RFC 2326

 Client-server application layer protocol.

 For user to control display: rewind, fast forward, pause, resume, repositioning, etc…

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/video

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RTSP Example

Scenario:  metafile communicated to web browser

 browser launches player

 player sets up an RTSP control connection, data connection to streaming server

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Metafile Example

<title>Twister</title>

<session>

<group language=en lipsync>

<switch>

<track type=audio

e="PCMU/8000/1"

src = "rtsp://audio.example.com/twister/audio.en/lofi">

<track type=audio

e="DVI4/16000/2" pt="90 DVI4/8000/1" src="rtsp://audio.example.com/twister/audio.en/hifi">

</switch>

<track type="video/jpeg"

src="rtsp://video.example.com/twister/video">

</group>

</session>

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RTSP Operation

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RTSP Exchange Example C: SETUP rtsp://audio.example.com/twister/audio RTSP/1.0 Transport: rtp/udp; compression; port=3056; mode=PLAY S: RTSP/1.0 200 1 OK Session 4231 C: PLAY rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 Range: npt=0- C: PAUSE rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 Range: npt=37 C: TEARDOWN rtsp://audio.example.com/twister/audio.en/lofi RTSP/1.0 Session: 4231 S: 200 3 OK

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Outline

 Multimedia Networking Applications

 Streaming stored audio and video  Streaming Architectures

 Real Time Streaming Protocol

 Packet Loss Recovery

 Streaming stored audio and video

 Scalable Streaming Techniques (Hot Topic)

 Content Distribution Networks (Hot Topic)

 Beyond Best Effort

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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  delays: processing, queueing in network; end-system

(sender, receiver) delays

 Tolerable delay depends on the application

 How can packet loss be handled?  We will discuss this next …

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Receiver-based Packet Loss Recovery

 Generate replacement packet  Packet repetition

 Interpolation

 Other sophisticated schemes

 Works when audio/video stream exhibits short- term self-similarity

 Works for relatively low loss rates (e.g., < 5%)

 Typically, breaks down on “bursty” losses

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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)

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Another FEC Example

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

• Whenever there is non-consecutive loss, the receiver can conceal the loss. • Can also append (n-1)st and (n-2)nd low-bit rate chunk

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Interleaving: Recovery from packet loss

Interleaving

 Re-sequence packets before transmission

 Better handling of “burst” losses

 Results in increased playout delay

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Summary: Internet Multimedia: bag of tricks

 use UDP to avoid TCP congestion control (delays) for time-sensitive traffic

 client-side adaptive playout delay: to compensate for delay

 server side matches stream bandwidth to available client-to-server path bandwidth  chose among pre-encoded stream rates

 dynamic server encoding rate

 error recovery (on top of UDP)  FEC, interleaving

 retransmissions, time permitting

 conceal errors: repeat nearby data

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Outline

 Multimedia Networking Applications

 Streaming stored audio and video

 Scalable Streaming Techniques

 Content Distribution Networks

 Beyond Best Effort

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Streaming Popular Content

 Consider a popular media file  Playback rate: 1 Mbps

 Duration: 90 minutes

 Request rate: once every minute

How can a video server handle such high loads?  Approach 1: Start a new “stream” for each

request

 Allocate server and disk I/O bandwidth for each request

 Bandwidth required at server= 1 Mbps x 90

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Streaming Popular Content using Batching

 Approach 2: Leverage the multipoint delivery capability of modern networks

 Playback rate = 1 Mbps, duration = 90 minutes

 Group requests in non-overlapping intervals of 30 minutes:  Max. start-up delay = 30 minutes

 Bandwidth required = 3 channels = 3 Mbps

0 3

0

60 90 120 150 180 210 240

Time (minutes)

Channel 1

Channel 2

Channel 3

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