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    Trng Th M Trang [email protected]

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    Introduction

    Streaming stored multimedia

    Real-time protocols

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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

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    Classifications of various media typeso Captured vs. synthesized media

    Captured media (natural) : information captured from the realworld

    Example: still image, video, audio

    Synthesized media (artificial) : information synthesize bythe computer

    Example: text, graphics, animation

    o Discrete vs. continuous media Discrete media: spaced-based, media involve the space

    dimension only Continuous media: time-based, media involves both the

    space and the time dimension

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    Sound Video

    Image

    Animation

    Text Graphics

    Captured

    From real world

    Synthesized

    By computer

    Discrete Discrete

    Continuous Continuous

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    Local vs. networkedo Local: storage and presentation of multimedia

    information in standalone computers

    Sample applications: DVDoNetworked: involve transmission and distribution

    of multimedia information on the network

    Sample applications: videoconferencing, web video

    broadcasting, multimedia Email, etc.

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    network audio and video (continuous media)

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    Fundamental characteristics:

    typically delay sensitive

    o end-to-end delay

    o delay jitter

    loss tolerant: infrequent

    losses cause minor glitches

    Classes of MM applications:

    1) stored streaming

    2) live streaming

    3) interactive, real-time

    Jitter is the variability of packet delays within the

    same packet stream

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    Example:

    o Music/video onlineo You tube

    o VOA

    o CNN

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    1. video

    recorded

    2. video

    sent3. video received,

    played out at client

    streaming: at this time, client

    playing out early part of video,

    while server still sending later

    part of video

    network

    delaytime

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    VCR-like functionality: client can pause,rewind, fast forward, push slider bar

    timing constraint for still-to-be transmitted

    data: in time for playout

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    Examples: Internet radio talk show

    live sporting event

    Streaming (as with streamingstoredmultimedia)

    playback buffer

    playback can lag tens of seconds after transmission

    still have timing constraint

    Interactivity fast forward impossible

    rewind, pause possible!

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    applications: IP telephony, video conference,distributed interactive worlds

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    TCP/UDP/IP: best-effort service

    no guarantees on delay, loss

    Todays Internet multimedia applications

    use application-level techniques to mitigate

    (as best possible) effects of delay, loss

    But you said multimedia apps requiresQoS and level of performance to be

    effective!

    ?? ??

    ?

    ?

    ? ??

    ?

    ?

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    1. How should the Internet evolve to

    better support multimedia?2. Transport: TCP? UDP? Other

    protocol?

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    Introduction

    Streaming stored multimedia

    Real-time protocols

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    application-levelstreaming techniquesfor making the best out

    of best effort service:o client-side buffering

    o use of UDP versus TCP

    o multiple encodings of

    multimedia

    jitter removal

    decompression error concealment

    graphical user interfacew/ controls for interactivity

    Media Player

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    audio, video not streamed:

    no, pipelining, long delays until playout!

    audio or video stored in file

    files transferred as HTTP objecto received in entirety at client

    o then passed to player

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    browser GETs metafile (contains information of URL, type of

    encoding, etc about the audio/video file)

    browser launches player, passing metafile

    player contacts server

    server streams audio/video to player

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    Introduction

    Streaming stored multimedia

    Real-time protocols

    o RTSP

    o RTP

    o RTCP

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    the protocols and their application fieldo stream description: M3U, RAM, SMIL...

    describe the session and content

    o stream control: RTSPremote control the session

    o media transport: RTPsend data and metadata

    o resource reservation (if any!): RSVP, DiffServmake sure the communication path offers appropriate guaranties

    otherwise Best-Effort transmissions!

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    Application protocol RFC 2326

    acts as a network remote control

    allows a media player to control the transmissionof a media stream for exchanging controlinformation

    supports the following operations:o retrieval of a media from a servero invitation of a media server to a conference

    o recording of a conference

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    video

    server

    audio

    server

    media descr.

    web serverclient

    step 1: get description (in SDP format)

    step 2: open streams with RTSP

    step 3: play

    step 4: teardown

    C

    W A V

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    Major methodso SETUP: server allocates resources for a

    stream and starts an RTSP sessiono PLAY: starts data tx on a streamo PAUSE: temporarily halts a streamo TEARDOWN: free resources of the stream, no

    RTSP session on server any more

    Additional methodso OPTIONS: get available methodso ANNOUNCE: change description of media objecto DESCRIBE: get low level descr. of media objecto RECORD: server starts recording a streamo REDIRECT: redirect client to new servero SET_PARAMETER: device or encoding control

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    client

    C

    web

    server

    W

    media

    servers

    A & V

    HTTP GET

    presentation description (sdp)

    SETUP

    PLAY

    RTP audio/video

    RTCP

    TEARDOWN

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    Transport protocol RFC 3550

    RTP specifies packet structure for packets

    carrying audio, video datao RTP packets encapsulated in UDP segments

    RTP packet provides

    opayload type identification

    opacket sequence numbering

    o time stamping

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    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/videobyte in the packet; used to remove jitter introduced by the network

    Synchronization Source identifier (SSRC): 32 bits; an id for the source ofa stream; assigned randomly by the sour

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    works in conjunction with RTP. Used to exchange control information (report)

    between the sender and the receiver

    o report statistics useful to application: # packetssent, # packets lost, interarrival jitter, etc.

    feedback can be used to control performance

    o sender may modify its transmissions based onfeedback

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    Bi ging gio trnh Computer Networking: ATop Down Approach, 5th edition, J.F Kurose

    and K.W. Ross

    Bi ging EE442Multimedia Networking,Jane Dong, California State University, Los

    Angeles