13 - Multimedia
13 - Multimedia
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digital Integration of multiple media type (text, graphics, still and moving images, animation, sounds, and any other medium) can be represented, stored, transmitted and processed digitally
Sound
Continuous
Video
Animation
Continuous
Image
Discrete Captured From real world
Text
Graphics
Discrete
Synthesized By computer
Jitter is the variability of packet delays within the same packet stream
Example:
o Music/video online o You tube o VOA o CNN
1. video recorded
streaming: at this time, client playing out early part of video, while server still sending later part of video
VCR-like functionality: client can pause, rewind, fast forward, push slider bar timing constraint for still-to-be transmitted data: in time for playout
Examples: Internet radio talk show live sporting event Streaming (as with streaming stored multimedia) playback buffer playback can lag tens of seconds after transmission still have timing constraint Interactivity fast forward impossible rewind, pause possible!
But you said multimedia apps requires QoS and level of performance to be ? ? effective!
?
?
Todays Internet multimedia applications use application-level techniques to mitigate (as best possible) effects of delay, loss
1. How should the Internet evolve to better support multimedia? 2. Transport: TCP? UDP? Other protocol?
application-level streaming techniques for making the best out of best effort service:
o client-side buffering o use of UDP versus TCP o multiple encodings of multimedia
Media Player
jitter removal decompression error concealment graphical user interface w/ controls for interactivity
audio, video not streamed: no, pipelining, long delays until playout!
encoding, etc about the audio/video file) browser launches player, passing metafile player contacts server server streams audio/video to player
RTP
<smil> <head> <meta name="Title" content="Streaming Media Sample"/> <meta name="Author" content="Toongabbie Anglican Church"/> <meta name="Copyright" content="Toongabbie Anglican Church"/> <layout> <region id="pic" top="0" left="0" width="150" height="150" background-color="white"/> </layout> </head> <body> <par> <audio src="http://simon.job.id.au/files/streaming/stream.rm" title="Freedom"/> <img src="http://simon.job.id.au/files/streaming/logo_rm.gif? url=http://simon.job.id.au/articles/3/streaming-media-tutorial" region="pic" title="picture" fill="freeze"/> </par> </body> </smil>
Application protocol RFC 2326 acts as a network remote control allows a media player to control the transmission of a media stream for exchanging control information supports the following operations:
o retrieval of a media from a server o invitation of a media server to a conference o recording of a conference
client
audio server
video server
step 1: get description (in SDP format) step 2: open streams with RTSP step 3: play step 4: teardown
Major methods
o SETUP:
o PLAY: o PAUSE: o TEARDOWN: o o o o o o
Additional methods
server allocates resources for a stream and starts an RTSP session starts data tx on a stream temporarily halts a stream free resources of the stream, no RTSP session on server any more
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
HTTP GET presentation description (sdp) client C SETUP PLAY RTP audio/video RTCP TEARDOWN
web server W
Transport protocol RFC 3550 RTP specifies packet structure for packets carrying audio, video data
o RTP packets encapsulated in UDP segments
Payload Type: 7 bits, providing 128 possible different types of encoding; eg PCM, MPEG2 video, etc. Sequence Number: 16 bits; used to detect packet loss Timestamp: 32 bytes; gives the sampling instant of the first audio/video byte in the packet; used to remove jitter introduced by the network Synchronization Source identifier (SSRC): 32 bits; an id for the source of a stream; assigned randomly by the sour
works in conjunction with RTP. Used to exchange control information (report) between the sender and the receiver
o report statistics useful to application: # packets sent, # packets lost, interarrival jitter, etc.
Bi ging gio trnh Computer Networking: A Top Down Approach, 5th edition, J.F Kurose and K.W. Ross Bi ging EE442Multimedia Networking, Jane Dong, California State University, Los Angeles