1 / 62

CE80N Introduction to Networks & The Internet

CE80N Introduction to Networks & The Internet. Dr. Chane L. Fullmer UCSC Winter 2002. Class Information. Web page tutorial available on-line Web page submission: Email to venkat@cse.ucsc.edu Subject: cmpe080n-assgn4 Final Exam Last class session March 14, 2002.

ivrit
Télécharger la présentation

CE80N Introduction to Networks & The Internet

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. CE80NIntroduction to Networks&The Internet Dr. Chane L. Fullmer UCSC Winter 2002

  2. Class Information • Web page tutorial available on-line • Web page submission: • Email to venkat@cse.ucsc.edu • Subject: cmpe080n-assgn4 • Final Exam • Last class session • March 14, 2002 CE80N -- Lecture #16, 2002

  3. Personal Web Page of the Day • A few brave souls…. • Presenting: CE80N -- Lecture #16, 2002

  4. AKA: Multimedia Networking

  5. Description of Functionality • Internet audio and video services make it possible to: • Send voice • Send live television images • Broadcast audio or video signals • Allow viewing and editing a document simultaneously CE80N -- Lecture #16, 2002

  6. Audio And Video Require Special Hardware • Live audio or video require high bandwidth. • Requires a computer with: • A microphone • A speaker • A camera • A high-speed processor • Controls all devices electronically CE80N -- Lecture #16, 2002

  7. Our goals: principles: network, application-level support for multimedia different forms of network multimedia, requirements making the best of best effort service mechanisms for providing QoS specific protocols, architectures for QoS Overview: multimedia applications and requirements Audio Video making the best of today’s best effort service Examples RealPlayer ISDN Videoconferencing Multimedia Networking CE80N -- Lecture #16, 2002

  8. Audio: Air pressure converted by microphone to voltage. Magnitude represents loudness or softness Frequency represents pitch, timbre. Sampled: Voltage saved at discrete points in time Quantized: Rounded off to a discrete value (x). Video: Light converted by camera into chemical deposition Magnitude represents brightness Frequency represents edges, contrast Sampled: Values saved at each horizontal/vertical (x,y) position in time Quantized: Rounded off to a discrete value (z) for each point (x,y). Digital Audio and Video CE80N -- Lecture #16, 2002

  9. Sample Size: 16 bit (2 byte) data representation Channels: 2 channels (stereo) Sampling Rate: 44,100 Samples Per Second(CD quality) One Channel: 2 bytes per sample x 44,100 samples per second = 88,200 bytes per second Total Data Rate: 88,200 bytes per second x 2 channels = 176,400 bytes per second Total Bit Rate = 1,411,200 bits per second This is DS1 rate!.. ADSL is typically a 384K downlink Digital Audio numbers CE80N -- Lecture #16, 2002

  10. YCC encoding scheme Luminance = brightness (Y) Chrominance = amount of color Cb = amount of blue Cr = amount of red Images Source: Peter Bourke http://astronomy.swin.edu.au/pbourke/colour/ycc/ CE80N -- Lecture #16, 2002

  11. Luminance = brightness (Y) Color Representation in YCC Source: Peter Bourke http://astronomy.swin.edu.au/pbourke/colour/ycc/ CE80N -- Lecture #16, 2002

  12. Chrominance Cr = amount of red Color Representation in YCC Source: Peter Bourke http://astronomy.swin.edu.au/pbourke/colour/ycc/ CE80N -- Lecture #16, 2002

  13. Chrominance Cb = amount of blue Color Representation in YCC Source: Peter Bourke http://astronomy.swin.edu.au/pbourke/colour/ycc/ CE80N -- Lecture #16, 2002

  14. Luminance stored for each pixel Chrominance stored for each 4x4 block of pixels YCCEncoding Source: Peter Bourke http://astronomy.swin.edu.au/pbourke/colour/ycc/ CE80N -- Lecture #16, 2002

  15. 720 x 486 pixels per frame 29.97 frames per second (Fps) Sample Size - 8 bits per pixel data representation Sampling - 4:2:2 (Every two horizontal pixels = 2 Y : 1 Cr : 1 Cb) Luminance (Y) 720 x 486 x 29.97 x 8 = 83,896,819.2 bits per second Chrominance R (Cr) 360 x 486 x 29.97 x 8 = 41,948,409.6 bits per second Chrominance B (Cb) 360 x 486 x 29.97 x 8 = 41,948,409.6 bits per second Total = ~20 Megabytes per second Phew…… !! Ethernet is only 1.25 Mbyte/s… Digital Video numbers CE80N -- Lecture #16, 2002

  16. Compression: Transmission of the same information in fewer bits. Run-length coding: Encode as symbol followed by number of symbols in a row. “0,0,0,0,0,0,0,0,0 ….0” replaced by “0 256” Huffman coding: (David Huffman, 1952) Builds a tree of symbols, assigning shorter bit codes to the more common symbols Arithmetic coding: Converts input symbols to a single real number Standards Images - JPEG (Joint Picture Experts Group) Video - MPEG (Moving Picture Experts Group) MP3 for audio Compression Coding Methods CE80N -- Lecture #16, 2002

  17. Compression: Transmission of the same information in fewer bits. Audio: Silence is transmitted as blank space in run-length coding. Video: Most objects stay fixed from frame to frame. Differences between frames transmitted. Audio and Video Compression CE80N -- Lecture #16, 2002

  18. Can save a lot of bits! Left-most picture: 24 bits per pixel Right-most picture: 9 bits per pixel Can you tell the difference? Image Compression Source: Peter Bourke http://astronomy.swin.edu.au/pbourke/colour/ycc/ CE80N -- Lecture #16, 2002

  19. Can be lossy Left-most picture: Original Center: 10:1 compression Right-most picture: 45:1 compression Can you tell the difference? Image Compression Source: Steven Smith http://www.dspguide.com/datacomp.htm CE80N -- Lecture #16, 2002

  20. Audio quality and Transmission Rates CE80N -- Lecture #16, 2002

  21. Video quality and Transmission Rates CE80N -- Lecture #16, 2002

  22. Useful chart of transmission rates: http://the-mid-west-web.com/thespeed.htm Transmission Rates CE80N -- Lecture #16, 2002

  23. QoS network provides application with level of performance needed for application to function. Multimedia, Quality of Service: What is it? Multimedia applications: network audio and video Source: http://www-net.cs.umass.edu/cs591/ Professor Jim Kurose Used with permission Slides 18-31, 34-51 CE80N -- Lecture #16, 2002

  24. Multimedia Performance Requirements Requirement: deliver data in “timely” manner • interactive multimedia: short end-end delay • e.g., IP telephony, telecon, virtual worlds • excessive delay impairs human interaction • streaming (non-interactive) multimedia: • data must arrive in time for “smooth” playout • late arriving data introduces gaps in rendered audio/video • reliability: 100% reliability not always required CE80N -- Lecture #16, 2002

  25. Interactive, Real-Time Multimedia • applications: IP telephony, video conference, distributed interactive worlds • end-end delay requirements: • video: < 150 msec acceptable • audio: < 150 msec good, < 400 msec OK • includes application-level (packetization) and network delays • higher delays noticeable, impair interactivity CE80N -- Lecture #16, 2002

  26. Interactive Multimedia: Videoconferencing Introduce Internet Phone by way of an example (note: there is no “standard” yet): • speaker’s audio: alternating talk spurts and silent periods. • pkts generated only during talk spurts • E.g., 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 the network every 20 msec during talkspurt. CE80N -- Lecture #16, 2002

  27. 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 can be concealed, packet loss rates between 1% and 10% can be tolerated. CE80N -- Lecture #16, 2002

  28. client reception constant bit rate playout at client variable network delay (jitter) buffered data client playout delay Delay Jitter constant bit rate transmission • Client-side buffering, playout delay compensate for network-added delay, delay jitter Cumulative data time CE80N -- Lecture #16, 2002

  29. 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 CE80N -- Lecture #16, 2002

  30. Fixed Playout Delay • Sender generates packets every 20 msec during talk spurt. • First packet received at time r • First playout schedule: begins at p • Second playout schedule: begins at p’ CE80N -- Lecture #16, 2002

  31. Recovery From Packet Loss • loss: pkt never arrives or arrives too late • real-time constraints: little (no) time for retransmissions! • What to do? • Forward Error Correction (FEC): add error correction bits (recall parity bits?) • e.g.,: add redundant chunk made up of exclusive OR of n chunks; redundancy is 1/n; can reconstruct if at most one lost chunk • Interleaving: spread loss evenly over received data to minimize impact of loss CE80N -- Lecture #16, 2002

  32. Interleaving • Has no redundancy, but can cause delay in playout beyond Real Time requirements • Divide 20 msec of audio data into smaller units of 5 msec each and interleave • Upon loss, have a set of partially filled chunks CE80N -- Lecture #16, 2002

  33. Piggybacking Lower Quality Stream CE80N -- Lecture #16, 2002

  34. Streaming 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 CE80N -- Lecture #16, 2002

  35. 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: what is it? Cumulative data time CE80N -- Lecture #16, 2002

  36. Streaming Multimedia (more) Types of interactivity: • none: like broadcast radio, TV • initial startup delays of < 10 secs OK • VCR-functionality: client can pause, rewind, FF • 1-2 sec until command effect OK • timing constraint for still-to-be transmitted data: in time for playout CE80N -- Lecture #16, 2002

  37. ? ? ? ? ? ? ? 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 CE80N -- Lecture #16, 2002

  38. Streaming Internet Multimedia Application-level streaming techniques for making the best out of best effort service: • what is streaming? • client side buffering • multiple rate encodings of multimedia ….. let’s look at these ….. CE80N -- Lecture #16, 2002

  39. 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! CE80N -- Lecture #16, 2002

  40. Internet multimedia: streaming approach • browser GETs metafile • browser launches player, passing metafile • player contacts server • server streams audio/video to player CE80N -- Lecture #16, 2002

  41. Downloadable: Download and save Listen later or send to others Standards-based Better quality Bandwidth burden on user Downloading takes time Streaming: Disposable Listen “on-the-fly” live Proprietary Lower quality Bandwidth burden on the developer More resources/time/cost No time lost in downloading Live broadcasting Downloadable vs. Streaming CE80N -- Lecture #16, 2002

  42. 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 CE80N -- Lecture #16, 2002

  43. Streaming Multimedia: Client Buffering • Client-side buffering, playout delay compensate for network-added delay, delay jitter constant drain rate, d variable fill rate, x(t) buffered video CE80N -- Lecture #16, 2002

  44. 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 CE80N -- Lecture #16, 2002

  45. User control of streaming multimedia Real Time Streaming Protocol (RTSP): RFC 2326 • user control: rewind, FF, pause, resume, etc… • out-of-band protocol: • one port (544) for control msgs • one port for media stream • TCP or UDP for control msg connection Scenario: • metafile communicated to web browser • browser launches player • player sets up an RTSP control connection, data connection to server CE80N -- Lecture #16, 2002

  46. 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> CE80N -- Lecture #16, 2002

  47. RTSP Operation CE80N -- Lecture #16, 2002

  48. Summary: Internet Multimedia: bag of tricks use UDP to avoid TCP congestion control (delays) for time-sensitive traffic • client-side playout delay: to compensate for network delay/jitter • 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 • retransmissions, time permitting • mask errors: repeat nearby data CE80N -- Lecture #16, 2002

  49. Example: RealPlayer RealNetworks • 1995, first streaming Internet audio (Progressive Networks) • 1997 • RealSystem (RealVideo, RealAudio, and RealFlash) • RealServer (client/server software) • Uses RTSP CE80N -- Lecture #16, 2002

  50. Example: RealPlayer Applications: • Internet server • Wide audience (most complex/expensive) • video commercials/e-commerce capabilities • Intranet server • Internal to business • desk-top training for employees • Commerce Solution • Secure transmissions to small groups • B2B, distance-learning, briefings CE80N -- Lecture #16, 2002

More Related