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Introduction to Multimedia Networking

Introduction to Multimedia Networking. Classify multimedia applications Identify the network services the apps need Making the best of best effort service Streaming Stored Multimedia vs. Interactive Applications Adaptive Playback and Smart Error Recovery Algorithms Some Common Protocols:

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Introduction to Multimedia Networking

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  1. Introduction to Multimedia Networking • Classify multimedia applications • Identify the network services the apps need • Making the best of best effort service • Streaming Stored Multimedia vs. Interactive Applications • Adaptive Playback and Smart Error Recovery Algorithms • Some Common Protocols: • RTSP, RTP/RTCP, SIP Required Readings: Relevant chapters/sections from your csci5211/csci4221 Textbook Multimedia

  2. QoS network provides application with level of performance needed for application to function. Multimedia and Quality of Service Multimedia applications: network audio and video (“continuous media”) Multimedia

  3. Digital Audio • Sampling the analog signal • Sample at some fixed rate • Each sample is an arbitrary real number • Quantizing each sample • Round each sample to one of a finite number of values • Represent each sample in a fixed number of bits 4 bit representation(values 0-15) Multimedia

  4. Audio Examples • Speech • Sampling rate: 8000 samples/second • Sample size: 8 bits per sample • Rate: 64 kbps • Compact Disc (CD) • Sampling rate: 44,100 samples/second • Sample size: 16 bits per sample • Rate: 705.6 kbps for mono, 1.411 Mbps for stereo Multimedia

  5. Why Audio Compression • Audio data requires too much bandwidth • Speech: 64 kbps is too high for a dial-up modem user • Stereo music: 1.411 Mbps exceeds most access rates • Compression to reduce the size • Remove redundancy • Remove details that human tend not to perceive • Example audio formats • Speech: GSM (13 kbps), G.729 (8 kbps), and G.723.3 (6.4 and 5.3 kbps) • Stereo music: MPEG 1 layer 3 (MP3) at 96 kbps, 128 kbps, and 160 kbps Multimedia

  6. Analog signal sampled at constant rate telephone: 8,000 samples/sec CD music: 44,100 samples/sec Each sample quantized, i.e., rounded e.g., 28=256 possible quantized values Each quantized value represented by bits 8 bits for 256 values Example: 8,000 samples/sec, 256 quantized values --> 64,000 bps Receiver converts it back to analog signal: some quality reduction Example rates CD: 1.411 Mbps MP3: 96, 128, 160 kbps Internet telephony: 5.3 - 13 kbps A few words about audio compression Multimedia

  7. Digital Video • Sampling the analog signal • Sample at some fixed rate (e.g., 24 or 30 times per sec) • Each sample is an image • Quantizing each sample • Representing an image as an array of picture elements • Each pixel is a mixture of colors (red, green, and blue) • E.g., 24 bits, with 8 bits per color Multimedia

  8. The 2272 x 1704hand The 320 x 240hand Multimedia

  9. Video is sequence of images displayed at constant rate e.g. 24 images/sec Digital image is array of pixels Each pixel represented by bits Redundancy spatial temporal Examples: MPEG 1 (CD-ROM) 1.5 Mbps MPEG2 (DVD) 3-6 Mbps MPEG4 (often used in Internet, < 1 Mbps) Research: Layered (scalable) video adapt layers to available bandwidth A few words about video compression Multimedia

  10. Video Compression: Within an Image • Image compression • Exploit spatial redundancy (e.g., regions of same color) • Exploit aspects humans tend not to notice • Common image compression formats • Joint Pictures Expert Group (JPEG) • Graphical Interchange Format (GIF) Uncompressed: 167 KB Good quality: 46 KB Poor quality: 9 KB Multimedia

  11. Video Compression: Across Images • Compression across images • Exploit temporal redundancy across images • Common video compression formats • MPEG 1: CD-ROM quality video (1.5 Mbps) • MPEG 2: high-quality DVD video (3-6 Mbps) • Proprietary protocols like QuickTime and RealNetworks Multimedia

  12. Fundamental characteristics: Typically delaysensitive end-to-end delay delay jitter But loss tolerant: infrequent losses cause minor glitches Antithesis of data, which are loss intolerant but delay tolerant. Classes of MM applications: 1) Streaming stored audio and video 2) Streaming live audio and video 3) Real-time interactive audio and video MM Networking Applications • Jitter is the variability • of packet delays within • the same packet stream Multimedia

  13. Application Classes • Streaming • 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.) • Delay: from client request until display start can be 1 to 10 seconds Multimedia

  14. Application Classes (more) • Unidirectional Real-Time: • similar to existing TV and radio stations, but delivery on the network • Non-interactive, just listen/view • Interactive Real-Time : • Phone conversation or video conference • More stringent delay requirement than Streaming and Unidirectional because of real-time nature • Video: < 150 msec acceptable • Audio: < 150 msec good, <400 msec acceptable Multimedia

  15. Streaming Stored Multimedia Streaming: • media stored at source • transmitted to client • streaming: client playout begins before all data has arrived • timing constraint for still-to-be transmitted data: in time for playout Multimedia

  16. 2. video sent 3. video received, played out at client 1. video recorded network delay streaming: at this time, client playing out early part of video, while server still sending later part of video Streaming Stored Multimedia: What is it? Cumulative data time Multimedia

  17. Streaming Stored Multimedia: Interactivity • VCR-like functionality: client can pause, rewind, FF, push slider bar • 10 sec initial delay OK • 1-2 sec until command effect OK • RTSP often used (more later) • timing constraint for still-to-be transmitted data: in time for playout Multimedia

  18. Streaming Live Multimedia Examples: • Internet radio talk show • Live sporting event Streaming • playback buffer • playback can lag tens of seconds after transmission • still have timing constraint Interactivity • fast forward impossible • rewind, pause possible! Multimedia

  19. Interactive, Real-Time Multimedia • applications: IP telephony, video conference, distributed interactive worlds • end-end delay requirements: • audio: < 150 msec good, < 400 msec OK • includes application-level (packetization) and network delays • higher delays noticeable, impair interactivity • session initialization • how does callee advertise its IP address, port number, encoding algorithms? Multimedia

  20. ? ? ? ? ? ? ? But you said multimedia apps requires QoS and level of performance to be effective! ? ? ? ? Today’s Internet multimedia applications use application-level techniques to mitigate (as best possible) effects of delay, loss Multimedia Over Today’s Internet TCP/UDP/IP: “best-effort service” • no guarantees on delay, loss Multimedia

  21. Challenges • TCP/UDP/IP suite provides best-effort, no guarantees on expectation or variance of packet delay • Streaming applications delay of 5 to 10 seconds is typical and has been acceptable, but performance deteriorate if links are congested (transoceanic) • Real-Time Interactive requirements on delay and its jitter have been satisfied by over-provisioning (providing plenty of bandwidth), what will happen when the load increases?... Multimedia

  22. Challenges (more) • Most router implementations use only First-Come-First-Serve (FCFS) packet processing and transmission scheduling • To mitigate impact of “best-effort” protocols, we can: • Use UDP to avoid TCP and its slow-start phase… • Buffer content at client and control playback to remedy jitter • Adapt compression level to available bandwidth Multimedia

  23. Integrated services philosophy: Fundamental changes in Internet so that apps can reserve end-to-end bandwidth Requires new, complex software in hosts & routers Laissez-faire no major changes more bandwidth when needed content distribution, application-layer multicast application layer Differentiated services philosophy: Fewer changes to Internet infrastructure, yet provide 1st and 2nd class service. How should the Internet evolve to better support multimedia? Will discuss QoS later! Multimedia

  24. Solution Approaches in IP Networks • Just add more bandwidth and enhance caching capabilities (over-provisioning)! • 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 routers Multimedia

  25. Application-level streaming techniques for making the best out of best effort service: client side buffering use of UDP versus TCP multiple encodings of multimedia jitter removal decompression error concealment graphical user interface w/ controls for interactivity Streaming Stored Multimedia Media Player Multimedia

  26. Internet Multimedia: Simplest Approach • audio or video stored in file • files transferred as HTTP object • received in entirety at client • then passed to player audio, video not streamed: • no, “pipelining,” long delays until playout! Multimedia

  27. Internet multimedia: Streaming Approach • browser GETs metafile • browser launches player, passing metafile • player contacts server • server streams audio/video to player Multimedia

  28. Streaming from a streaming server • This architecture allows for non-HTTP protocol between server and media player • Can also use UDP instead of TCP. Multimedia

  29. 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 User Control of Streaming Media: RTSP Multimedia

  30. FTP uses an “out-of-band” control channel: A file is transferred over one TCP connection. Control information (directory changes, file deletion, file renaming, etc.) is sent over a separate TCP connection. The “out-of-band” and “in-band” channels use different port numbers. RTSP messages are also sent out-of-band: RTSP control messages use different port numbers than the media stream: out-of-band. Port 554 The media stream is considered “in-band”. RTSP: out of band control Multimedia

  31. RTSP Example Scenario: • metafile communicated to web browser • browser launches player • player sets up an RTSP control connection, data connection to streaming server Multimedia

  32. 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> Multimedia

  33. RTSP Operation Multimedia

  34. 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 Multimedia

  35. client video reception constant bit rate video playout at client variable network delay buffered video client playout delay Streaming Multimedia: Client Buffering constant bit rate video transmission • Client-side buffering, playout delay compensate for network-added delay, delay jitter Cumulative data time Multimedia

  36. Streaming Multimedia: Client Buffering “constant” drain rate, d • Client-side buffering, playout delay compensate for network-added delay, delay jitter variable fill rate, x(t) buffered video Multimedia

  37. 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 Multimedia

  38. Streaming Multimedia: client rate(s) 1.5 Mbps encoding 28.8 Kbps encoding Q:how to handle different client receive rate capabilities? • 28.8 Kbps dialup • 100Mbps Ethernet A:server stores, transmits multiple copies of video, encoded at different rates Multimedia

  39. PC-2-PC phone instant messaging services are providing this, e.g. Yahoo messenger, googletalk Skype PC-2-phone Dialpad Net2phone Skype videoconference with Webcams Going to now look at a PC-2-PC Internet phone example in detail Real-time interactive applications Multimedia

  40. Voice Over IP (VoIP) • Delivering phone calls over IP • Computer to computer • Analog phone to/from computer • Analog phone to analog phone • Motivations for VoIP • Cost reduction • Simplicity • Advanced applications • Web-enabled call centers • Collaborative white boarding • Do Not Disturb, Locate Me, etc. • Voicemail sent as e-mail Multimedia

  41. 7040 External line 212-8538080 7041 Corporate/Campus Telephone switch Another switch Private Branch Exchange 7042 7043 Internet Corporate/Campus LAN Traditional Telecom Infrastructure Multimedia

  42. Corporate/Campus Another campus 7040 8151 External line 8152 7041 PBX PBX 8153 VoIP Gateway 8154 7042 VoIP Gateway 7043 Internet LAN LAN IP Phone Client VoIP Gateways Multimedia

  43. VoIP With an Analog Phone • Adapter • Converts between analog and digital • Sends and receives data packets • Communicates with the phone in standard way Multimedia

  44. Skype • Niklas Zennström and Janus Friis in 2003 • Developed by KaZaA • Instant Messenger (IM) with voice support • Based on peer-to-peer (P2P) networking technology Multimedia

  45. Skype Network Architecture • Login server is the only central server (consisting of multiple machines) • Both ordinary host and super nodes are Skype clients • Any node with a public IP address and having sufficient resources can become a super node • Skype maintains their own super nodes Multimedia

  46. Challenges of Firewalls and NATs • Firewalls • Often block UDP traffic • Usually allow hosts to initiate connections on port 80 (HTTP) and 443 (HTTPS) • NAT • Cannot easily initiate traffic to a host behind a NAT • … since there is no unique address for the host • Skype must deal with these problems • Discovery: client exchanges messages with super node • Traversal: sending data through an intermediate peer Multimedia

  47. Interactive Multimedia: Internet Phone Introduce Internet Phone by way of an example • speaker’s audio: alternating talk spurts, silent periods. • 64 kbps during talk spurt • pkts generated only during talk spurts • 20 msec chunks at 8 Kbytes/sec: 160 bytes data • application-layer header added to each chunk. • Chunk+header encapsulated into UDP segment. • application sends UDP segment into socket every 20 msec during talkspurt. Multimedia

  48. Internet Phone: Packet Loss and Delay • 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 • typical maximum tolerable delay: 400 ms • loss tolerance: depending on voice encoding, losses concealed, packet loss rates between 1% and 10% can be tolerated. Multimedia

  49. client reception constant bit rate playout at client variable network delay (jitter) buffered data client playout delay Delay Jitter constant bit rate transmission • Consider the end-to-end delays of two consecutive packets: difference can be more or less than 20 msec Cumulative data time Multimedia

  50. Internet Phone: Fixed Playout Delay • Receiver attempts to playout each chunk exactly q msecs after chunk was generated. • chunk has time stamp t: play out chunk at t+q . • chunk arrives after t+q: data arrives too late for playout, data “lost” • Tradeoff for q: • large q: less packet loss • small q: better interactive experience Multimedia

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