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CE80N Introduction to Networks & The Internet Dr. Chane L. Fullmer UCSC Winter 2002.

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1 CE80N Introduction to Networks & The Internet Dr. Chane L. Fullmer UCSC Winter 2002

2 February 28, 2002CE80N -- Lecture #16, Class Information Web page tutorial available on-line Web page submission: – to Subject: cmpe080n-assgn4 Final Exam –Last class session March 14, 2002

3 February 28, 2002CE80N -- Lecture #16, Personal Web Page of the Day A few brave souls…. Presenting:

4 AKA: Multimedia Networking

5 February 28, 2002CE80N -- Lecture #16, 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

6 February 28, 2002CE80N -- Lecture #16, 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

7 February 28, 2002CE80N -- Lecture #16, Multimedia Networking 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

8 February 28, 2002CE80N -- Lecture #16, Digital Audio and Video 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).

9 February 28, 2002CE80N -- Lecture #16, Digital Audio numbers 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

10 February 28, 2002CE80N -- Lecture #16, Images YCC encoding scheme –Luminance = brightness (Y) –Chrominance = amount of color Cb = amount of blue Cr = amount of red Source: Peter Bourke

11 February 28, 2002CE80N -- Lecture #16, Color Representation in YCC Luminance = brightness (Y) Source: Peter Bourke

12 February 28, 2002CE80N -- Lecture #16, Color Representation in YCC Chrominance –Cr = amount of red Source: Peter Bourke

13 February 28, 2002CE80N -- Lecture #16, Color Representation in YCC Chrominance –Cb = amount of blue Source: Peter Bourke

14 February 28, 2002CE80N -- Lecture #16, YCC Encoding Luminance stored for each pixel Chrominance stored for each 4x4 block of pixels Source: Peter Bourke

15 February 28, 2002CE80N -- Lecture #16, Digital Video numbers 720 x 486 pixels per frame 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 x 8 = 83,896,819.2 bits per second –Chrominance R (Cr) 360 x 486 x x 8 = 41,948,409.6 bits per second –Chrominance B (Cb) 360 x 486 x x 8 = 41,948,409.6 bits per second Total = ~20 Megabytes per second –Phew…… !! Ethernet is only 1.25 Mbyte/s…

16 February 28, 2002CE80N -- Lecture #16, Compression Coding Methods 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

17 February 28, 2002CE80N -- Lecture #16, Audio and Video Compression 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.

18 February 28, 2002CE80N -- Lecture #16, Image Compression 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? Source: Peter Bourke

19 February 28, 2002CE80N -- Lecture #16, Image Compression Can be lossy –Left-most picture: Original –Center: 10:1 compression –Right-most picture: 45:1 compression Can you tell the difference? Source: Steven Smith

20 February 28, 2002CE80N -- Lecture #16, Audio quality and Transmission Rates Sound qualityBandwidthModeBitrateReduction ratio Telephone2.5 kHzMono8 kbps96:1 Shortwave4.5 kHzMono16 kbps48:1 AM radio7.5 kHzMono32 kbps24:1 FM11 kHzStereo56…64 kbps26…24:1 Near-CD15 kHzStereo96 kbps16:1 CD>15 KHzStereo kbps14:12:1

21 February 28, 2002CE80N -- Lecture #16, Video quality and Transmission Rates

22 February 28, 2002CE80N -- Lecture #16, Transmission Rates Useful chart of transmission rates:

23 February 28, 2002CE80N -- Lecture #16, Multimedia, Quality of Service: What is it? Multimedia applications: network audio and video network provides application with level of performance needed for application to function. QoS Source: Professor Jim Kurose Used with permission Slides 18-31, 34-51

24 February 28, 2002CE80N -- Lecture #16, 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

25 February 28, 2002CE80N -- Lecture #16, Interactive, Real-Time Multimedia 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 applications: IP telephony, video conference, distributed interactive worlds

26 February 28, 2002CE80N -- Lecture #16, 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.

27 February 28, 2002CE80N -- Lecture #16, 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.

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

29 February 28, 2002CE80N -- Lecture #16, 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

30 February 28, 2002CE80N -- Lecture #16, 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’

31 February 28, 2002CE80N -- Lecture #16, 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

32 February 28, 2002CE80N -- Lecture #16, 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

33 February 28, 2002CE80N -- Lecture #16, Piggybacking Lower Quality Stream

34 February 28, 2002CE80N -- Lecture #16, 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

35 February 28, 2002CE80N -- Lecture #16, Streaming: what is it? 1. video recorded 2. video sent 3. video received, played out at client Cumulative data streaming: at this time, client playing out early part of video, while server still sending later part of video network delay time

36 February 28, 2002CE80N -- Lecture #16, 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

37 February 28, 2002CE80N -- Lecture #16, Multimedia Over Today’s Internet TCP/UDP/IP: “best-effort service” no guarantees on delay, loss Today’s Internet multimedia applications use application-level techniques to mitigate (as best possible) effects of delay, loss But you said multimedia apps requires QoS and level of performance to be effective! ? ? ?? ? ? ? ? ? ? ?

38 February 28, 2002CE80N -- Lecture #16, 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 …..

39 February 28, 2002CE80N -- Lecture #16, 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!

40 February 28, 2002CE80N -- Lecture #16, Internet multimedia: streaming approach browser GETs metafile browser launches player, passing metafile player contacts server server streams audio/video to player

41 February 28, 2002CE80N -- Lecture #16, Downloadable vs. Streaming 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

42 February 28, 2002CE80N -- Lecture #16, constant bit rate video transmission Cumulative data time variable network delay client video reception constant bit rate video playout at client client playout delay buffered video Streaming Multimedia: Client Buffering Client-side buffering, playout delay compensate for network-added delay, delay jitter

43 February 28, 2002CE80N -- Lecture #16, Streaming Multimedia: Client Buffering Client-side buffering, playout delay compensate for network-added delay, delay jitter buffered video variable fill rate, x(t) constant drain rate, d

44 February 28, 2002CE80N -- Lecture #16, Streaming Multimedia: client rate(s) 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 1.5 Mbps encoding 28.8 Kbps encoding

45 February 28, 2002CE80N -- Lecture #16, 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

46 February 28, 2002CE80N -- Lecture #16, Metafile Example Twister

47 February 28, 2002CE80N -- Lecture #16, RTSP Operation

48 February 28, 2002CE80N -- Lecture #16, 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

49 February 28, 2002CE80N -- Lecture #16, Example: RealPlayer RealNetworks 1995, first streaming Internet audio (Progressive Networks) 1997 –RealSystem (RealVideo, RealAudio, and RealFlash) –RealServer (client/server software) Uses RTSP

50 February 28, 2002CE80N -- Lecture #16, 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

51 February 28, 2002CE80N -- Lecture #16, Radio Programs On The Internet A radio station uses computer equipment that converts the audio signal to digital form. –Requires software that contacts the station –Extracts digitized audio from the packets –Converts the data to sound –Plays the sound KPIG Radio on the Internet

52 February 28, 2002CE80N -- Lecture #16, Real-Time Transmission Is Called Webcasting To enable a browser to play real-time audio or video, the browser must be extended with a plugin. –Consists of software Ie, Real Audio, M$ Media Player –Keeps a list of packets in memory –Enables the browser to receive and play audio/video Monterey Bay Aquarium live videocast Monterey Bay Aquarium

53 February 28, 2002CE80N -- Lecture #16, Software for Collaboration Software exists that permits a group of users to examine and edit a single document. –Called a whiteboard service

54 February 28, 2002CE80N -- Lecture #16, Marking A Document A participant uses a mouse to control interaction with the whiteboard service. –Allows all users to see changes on their own screens, as well as reflected on the screens of others

55 February 28, 2002CE80N -- Lecture #16, The Participants Discuss And Mark A Document Usually participants may also engage in an audio teleconference at the same time. Changes: –Can be noted in colors –Saved to a word processing document

56 February 28, 2002CE80N -- Lecture #16, Video Teleconferencing Video teleconferencing enables face-to-face communication. –Requires software to start a video session M$ NetMeeting Vic/vat for Unix –Incorporates both video and audio technology

57 February 28, 2002CE80N -- Lecture #16, Video Teleconference Among Groups Of People The screen cannot hold individual images from many cameras. –Gather people in smaller groups with a: Camera Large screen monitor or TV Microphones for audio

58 February 28, 2002CE80N -- Lecture #16, A Combined Audio, Video, And Whiteboard Service Teleconferencing becomes more interesting when combined with a whiteboard service. –Provides the most flexibility

59 February 28, 2002CE80N -- Lecture #16, Conclusion How do you think telephone companies view Internet telephone service? (What!? Free long distance?) Would you be willing to watch the evening news on your computer?

60

61 February 28, 2002CE80N -- Lecture #16, Glossary Audio Teleconference –A service that allows a group of users to exchange audio information over the Internet similar to a telephone conference call. Whiteboard Service –A service that permits a group of users to establish a session that enables all of them to see and modify the same display.

62 February 28, 2002CE80N -- Lecture #16, Glossary Bandwidth –The capacity of a network, usually measured in bits per second. Video Teleconference Service –A service that allows a group of users to exchange video information over the Internet. Delay Jitter –The variance in delay seen at the receiver of an audio/video stream


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