ch 6. multimedia networking myungchul kim mckim@icu.ac.kr

Ch 6. Multimedia Networking

Myungchul Kim

mckim@icu.ac.kr

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Networked multimedia applications: timing and tolerance of data loss

Delay-sensitive and loss-tolerant Streaming stored audio and video

– Stored media– Streaming: avoids having to download the entire file before begi

nning playout. Realplayer, QuickTime and Media Player– Continuous playout

Streaming live audio and video– Not stored, not fast-forward– Use the IP multicast

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Real-time interactive audio and video– Real-time– Interactive– Internet phone– For voice, 150 msec, 150-400 msec, 400 msec

Hurdles for multimedia– End-to-end delay for a packet– Variation of packet delay, packet jitter– Packet loss

Supporting multimedia better in Internet– Reservation approach– Laissez-faire approach: ISP, CDN, multicast overlay networks– Differentiation approach

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Audio and video compression– 1024 pixels * 1024 pixels with each pixel encoded into 24 bits => 3 Mbyt

e– 7 Min over a 64 kbps link– If the image is compressed at 10:1,

Audio compression– 8000 samples per second -> quantization with 256 values (8bits)-> 64,0

00 bits/second– Pulse code modulation– GSM, G.729, MPEG 1 layer 3(MP3),…

Video compression– MPEG 1, 2, 4, H.261

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Real-time streaming protocol (RTSP)– User interactivity– RealPlayer and Media Player– Decompression, jitter removal, and correction– Fig 6.2

Streaming stored audio and video

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Fig 6.3

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RTSP– Control the playback of continuous media– No related with compression schemes, encapsulation in packets,

transportation, buffering– Out-of-band protocol– Over either TCP or UDP– Pause/resume, playback, fast-forward, and rewind

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Fig 6.5

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Every 20 msec over UDP Packet loss, end-to-end delay, and packet jitter Removing jitter at the receiver for audio

– With a sequence number, a timestamp or – delaying playout at the receiver

Internet phone

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Fig 6.6

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RTP– For sound and video– On UDP– RTP header: the type of audio encoding, a sequence number, a

nd a timestamp– Sequence number: detect packet loss – Timestamp: synchronous playout at the receiver– Synchronization source identifier (SSRC): identify the source of t

he RTP stream– Fig 6.9

Protocols for real-time interactive applications

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Table 6.1 and 6.2

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Developing software applications with RTP

Fig 6.10 and 6.11

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RTP control protocol (RTCP)– In conjunction with RTP– Report statistics including number of packets sent, number of pa

ckets lost, and interarrival jitter.– RTP traffic vs RTSP traffic grows linearly with the number of rec

eivers (5 % of the session bandwidth).– Fig 6.12

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SIP– Establish calls between a caller and a callee over an IP network– Caller determines the current IP address of the callee– Call management– Fig 6.13

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SIP– SIP proxy and registrar (cf. DNS)– Fig 6.14

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H.323– Fig 6.15

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FIFO– Fig 6.21

Priority queuing using ToS– Fig 6.23

Scheduling and policing mechanisms

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Round robin and weighted fair queuing (WFQ)– WFQ differs from round robin in that each class may receive a di

fferential amount of service in any interval of time.– Fig 6.26

Priority queuing using ToS– Fig 6.23

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Policing: the leaky bucket– Policing criteria: average rate, peak rate, and burst rate– Polices a traffic flow– At most b tokens in the bucket = max burst size for leaky-bucket-

policed flow– Long-term average rate = Max number of packets entering the n

etwork in time t = rt + b– Fig 6.27

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Individual applications sessions– Reserved resources– Call setup

Steps– Traffic characterization and specification of the desired QoS– Signaling for call setup– Per-element call admission– Fig 6.29

Integrated services

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– Fig 6.30

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Resources– Link bandwidth– Router buffers

Characteristics– Reservations for bandwidth in multicast trees– Receiver-oriented– Not specify how the network provides the reserved bandwidth to

the data flow -> provisioning done with the scheduling mechanisms

– Not a routing protocol– Signaling protocol

RSVP

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Example– Video and audio be encoded in layers– The receiver pick out the layers that are appropriate for their rec

eiving rates.– Multicast routing protocol -> multicast tree -> rsvp– Fig 6.32

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Difficulties of the Interserv– Scalability: per-flow using RSVP– Flexible service models: limited service classes

Architecture– Edge function: packet classification with marking and traffic cond

itioning, behavior aggregate– Core function: forwarding, per-hop behavior– Fig 6.34

Differentiated service

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Fig 6.35

Fig 6.36

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– Traffic profile– Metering function: compare the incoming packet flow with the

negotiated traffic profile and determine whether a packet is within the negotiated traffic profile.

– Fig 6.37

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per-Hop behaviors– Expedited forwarding:– Assured forwarding

Criticisms of differentiated services– Multiple ISPs– How to police and authenticate