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Ch 7. Multimedia Networking Myungchul Kim

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Presentation on theme: "Ch 7. Multimedia Networking Myungchul Kim"— Presentation transcript:

1 Ch 7. Multimedia Networking Myungchul Kim mckim@icu.ac.kr

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

3 3 – Sensitive to end-to-end delay and delay variation – Streaming stored audio/video – Streaming live audio/video – Real-time interactive audio/video

4 4 Multimedia networking applications o Examples of multimedia applications – Streaming stored audio and video  Stored media  Streaming: RealPlayer, QuickTime, Media  Continous playout – Streaming live audio and video  Internet radio and IPTV  IP multicasting  Application-layer multicast – Real-time interactive audio and video  Internet telephony (150 msec)

5 5 o Hurdles for multimedia in Today’s Internet – Best-effor service o How should the Internet evolve to support multimedia better? – Hard guarantee vs soft guarnatee – Reservation approach  Protocol  Modification of scheduling policies in the router queues  Description of the application traffic  Available bandwidth in the network – Laissez-faire approach  Overprovision bandwidth and switching capacity  Content distribution networks (CDN)  Multicast overlay networks o Differentiated service (Diffserv)

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7 7 o Audio compression in the Internet – 8,000 samples per second – 256 quantization with 8 bits – 64Kbps – Pulse code modulation (PCM) – GSM, G.729, G.723.3, MPEG 1 player 3 (MP3) o Video compression in the Internet – MPEG1, 2, 4 – H.261

8 8 Streaming Stored Audio and Video o Medio player – Decompression – Jitter removal

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10 10 o Real-time Streaming Protocol (RTSP)

11 11 Making the best of the best-effort service – Packet loss – End-to-end delay – Packet jitter o Removing jitter at the receiver for audio – Sequence number – Timestamp – Delaying playout at the receiver

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13 13 o Recovering from packet loss

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15 15 o Content Distribution Networks

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17 17 o Dimensioning best-effort networks to provide Quality of Service – Bandwidth provisioning – Network dimensioning – Models of traffic demand between network end points – Well-defined performance requirements – Workload model

18 18 Protocols for Real-time Interactive Applications o RTP – UDP – RTP header: the type of audio encoding, a sequence number, and a timestamp

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21 21 o RTP control protocol (RTCP) – Using IP multicast – Reports about statistics – Reception report  SSRC of the RTP streams  The fraction of packets lost  The last sequence number received  The interarrival jitter – Sender report  The SSRC of the RTP streams  The timestamp and wall clock time of the most recently generated RTP packet  The number of packets sent  The number of bytes sent

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23 23 o Session Initiation Protocol (SIP) – Protocol does  Establishing calls between a caller and a callee over an IP network  For the caller to determine the current IP address of the callee  Call management – Key characteristics  Out-of-band protocol  ASCII-readable  All messages to be acknowledged

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25 25 Setting up a call to known IP address  Alice’s SIP invite message indicates her port number, IP address, encoding she prefers to receive (PCM ulaw)  Bob’s 200 OK message indicates his port number, IP address, preferred encoding (GSM)  SIP messages can be sent over TCP or UDP; here sent over RTP/UDP.  default SIP port number is 5060.

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27 27 Example Caller jim@umass.edu with places a call to keith@upenn.edu (1) Jim sends INVITE message to umass SIP proxy. (2) Proxy forwards request to upenn registrar server. (3) upenn server returns redirect response, indicating that it should try keith@eurecom.fr (4) umass proxy sends INVITE to eurecom registrar. (5) eurecom registrar forwards INVITE to 197.87.54.21, which is running keith’s SIP client. (6-8) SIP response sent back (9) media sent directly between clients. Note: also a SIP ack message, which is not shown.

28 28 o H.323

29 29 Providing multiple classes of service – Divide traffic into classes and provide different levels of service to the different classes of traffic. – Differentiated service is provided among aggregates of traffic. – Type-of-service (ToS) in the IPv4

30 30 o Scenario 1: a 1 Mbps audio application and an FTP transfer – FIFO – Give strict priority to audio packets at R1 – Each packet must be marked as belonging to one of these two classes of traffic, e.g., ToS in IPv4

31 31 o Scenario 2: a 1 Mbps audio application and a high- priority FTP transfer – Packet classification allows a router to distinguish among packets belonging to different classes of traffic. – A policy decision

32 32 o Scenario 3: A misbehaving audio application and an FTP transfer

33 33 Scheduling and policing mechanisms o Link-scheduling mechanisms – First-In-First-Out (FIFO)

34 34 – Priority Queueing

35 35 – Round robin and weighted fair queueing (WFQ)

36 36 – Policing: The Leaky Bucket: regulate the injecting rate of packets into the networks  Average rate  Peak rate  Burst size

37 37 o Diffserv – Edge function: packet classification and traffic conditioning: the diffentiated service field of the packet header – Core function: forwarding, per-hop behavior, aggregation

38 38 – Diffserv traffic classfication and conditioning

39 39 – Per-hop behaviors  Differences in performance among classes  Differences in performance observable and measureable  Expedited forwarding, assured forwarding

40 40 Providing quality of service guarantees o Resouce reservation, call admission, call setup – Traffic characterization and specification of the desired QoS – Signaling for call setup – Pre-element call admission

41 41 o Guaranteed QoS: Intserv and RSVP – Individualized QoS guarantees – Reservations for bandwidth in multicast trees – Receiver-oriented – Provisioning? Using the policing and scheduling


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