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UW-Madison CS 640 - Lecture 19 - Multimedia Networking

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CS640: Introduction to Computer NetworksThe Road AheadApplication ClassesApplication Classes (more)Slide 5Multimedia TodaySolution Approaches in IP NetworksApplication Example: StreamingStreamingStreaming From Web ServersStreaming From Web ServerUsing a Streaming ServerOptions When Using a Streaming ServerReal Time Streaming Protocol (RTSP)Presentation Description ExampleRTSP OperationReal-Time (Phone) Over IP’s Best-EffortRemoving “Jitter”Fixed Playout DelayRecovery From Packet LossAnother FEC MechanismPiggybacking Lower Quality StreamInterleavingReal-Time Protocol (RTP)Slide 25RTP Control Protocol (RTCP)RTCP Bandwidth ScalingCS640: Introduction to Computer NetworksAditya AkellaLecture 19 -Multimedia Networking2The Road Ahead•Multimedia requirements–Streaming•RTSP–Recovering from Jitter and Loss–RTP•RTCP3Application Classes •Typically sensitive to delay, but can tolerate packet loss (would cause minor glitches that can be concealed)•Data contains audio and video content (“continuous media”), three classes of applications:–Streaming stored content–Unidirectional Real-Time–Interactive Real-Time4Application Classes (more)•Streaming stored content–Clients request audio/video files from servers and pipeline reception over the network and display•Interactive: user can control operation (similar to VCR: pause, resume, fast forward, rewind, etc.)–Streaming  start playing before all content arrives–Continuous playout: hard delivery constraints5Application Classes (more)•Unidirectional Real-Time:–similar to existing TV and radio stations, but delivery on the network–Non-interactive, just listen/view–Delivery constraints still important•Interactive Real-Time:–Phone conversation or video conference–More stringent delay requirement than Streaming and Unidirectional because of interactive nature–Video: < 150 msec acceptable–Audio: < 150 msec good, <400 msec acceptable6Multimedia Today•Network is best-effort. But still…–Streaming applications delay of 5 to 10 seconds is typical and has been acceptable–Real-Time apps work well where there is plentiful bandwidth•To mitigate impact of “best-effort” network and protocols, we can: –Use UDP, avoid TCP and its slow-start phase…–Buffer content at client, control playback, prefetch content to remedy delay variation–Adapt compression level to available bandwidth in the network–Send redundant information to make up for losses–Intelligent queueing tricks7Solution Approaches in IP Networks•Just add more bandwidth enhance caching capabilities etc. (previous slide)!•Need major change of the protocols :–Incorporate resource reservation (bandwidth, processing, buffering), and new scheduling policies –Set up service level agreements with applications, monitor and enforce the agreements, charge accordingly•Need moderate changes (“Differentiated Services”):–Use two traffic classes for all packets and differentiate service accordingly–Charge based on class of packets–Network capacity is provided to ensure first class packets incur no significant delay at routers8Application Example: Streaming•Important and growing application–Due to reduction of storage costs, increase in high speed net access from homes and enhancements to caching•Audio/Video file is segmented and sent over either TCP or UDP–Web server–Streaming server•Public segmentation protocol: Real-Time Protocol (RTP)•User Interaction: Real-time Streaming protocol (RTSP)9Streaming•Helper Application: displays content, which is typically requested via a Web browser; e.g. RealPlayer; typical functions:–Decompression–Jitter removal–Error correction: use redundant packets to be used for reconstruction of original stream–GUI for user control10Streaming From Web Servers•Audio: in files sent as HTTP objects•Video (interleaved audio and images in one file, or two separate files and client synchronizes the display) sent as HTTP object(s)•A simple architecture is to have the Browser request the object(s) and after their reception pass them to the player for display- No pipelining11Streaming From Web Server•Alternative: set up connection between server and player, then download•Web browser requests and receives a Meta File (a file describing the object) instead of receiving the file itself; •Browser launches the appropriate Player and passes it the Meta File; •Player sets up a TCP connection with Web Server and downloads the file using HTTP12Using a Streaming Server•This gets us around HTTP, allows use of UDP vs. TCP and the application layer protocol can be better tailored to Streaming; many enhancements options are possible …Separateout functionality13Options When Using a Streaming Server•UDP: Server sends at a rate (Compression and Transmission) appropriate for client; to reduce jitter, Player buffers initially for 2-5 seconds, then starts display•Use TCP, and sender sends at maximum possible rate under TCP; retransmit when error is encountered; Player uses a much large buffer to smooth delivery rate of TCP14Real Time Streaming Protocol (RTSP)•For user to control display: rewind, fast forward, pause, resume, etc…•Out-of-band protocol (uses two connections, one for control messages (Port 554) and one for media stream)•As before, meta file is communicated to web browser which then launches the Player; –Meta file contains “presentation description file” which has information on the multi-media content15Presentation Description Example<title>Xena: Warrior Princess</title> <session> <group language=en lipsync> <switch> <track type=audio e="PCMU/8000/1" src = "rtsp://audio.example.com/xena/audio.en/lofi"> <track type=audio e="DVI4/16000/2" pt="90 DVI4/8000/1" src="rtsp://audio.example.com/xena/audio.en/hifi"> </switch> <track type="video/jpeg" src="rtsp://video.example.com/twister/video"> </group> </session>16RTSP Operation•C: SETUP rtsp://audio.example.com/xena/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/xena/audio.en/lofi RTSP/1.0 Session: 4231 Range: npt=0- (npt = normal play time)•C: PAUSE rtsp://audio.example.com/xena/audio.en/lofi RTSP/1.0 Session: 4231 Range: npt=37 • C: TEARDOWN rtsp://audio.example.com/xena/audio.en/lofi RTSP/1.0 Session: 4231 •


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UW-Madison CS 640 - Lecture 19 - Multimedia Networking

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