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

Ch 7. Multimedia Networking

Myungchul Kim

mckim@icu.ac.kr

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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

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– Sensitive to end-to-end delay and delay variation

– Streaming stored audio/video– Streaming live audio/video– Real-time interactive audio/video

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Multimedia networking applications

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)

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Hurdles for multimedia in Today’s Internet– Best-effor service

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

Differentiated service (Diffserv)

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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)

Video compression in the Internet– MPEG1, 2, 4– H.261

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Streaming Stored Audio and Video

Medio player– Decompression– Jitter removal

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Real-time Streaming Protocol (RTSP)

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Making the best of the best-effort service

– Packet loss– End-to-end delay– Packet jitter

Removing jitter at the receiver for audio– Sequence number– Timestamp– Delaying playout at the receiver

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Recovering from packet loss

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Content Distribution Networks

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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

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Protocols for Real-time Interactive Applications RTP

– UDP– RTP header: the type of audio encoding, a sequence number, a

nd a timestamp

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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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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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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.

time time

Bob'stermina l rings

A lice

167.180.112.24

Bob

193.64.210.89

port 38060

Law audio

G SMport 48753

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ExampleCaller jim@umass.edu with places a call to keith@upenn.edu

(1) Jim sends INVITEmessage to umass SIPproxy. (2) Proxy forwardsrequest to upenn registrar server. (3) upenn server returnsredirect 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.

SIP client217.123.56.89

SIP client197.87.54.21

SIP proxyum ass.edu

SIP registrarupenn.edu

SIPregistrareurecom .fr

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H.323

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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

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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

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Scenario 2: a 1 Mbps audio application and a high-priority FTP transfer– Packet classification allows a router to distinguish among packe

ts belonging to different classes of traffic.– A policy decision

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Scenario 3: A misbehaving audio application and an FTP transfer

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Scheduling and policing mechanisms

Link-scheduling mechanisms– First-In-First-Out (FIFO)

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– Priority Queueing

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– Round robin and weighted fair queueing (WFQ)

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

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Diffserv– Edge function: packet classification and traffic conditioning: the

diffentiated service field of the packet header– Core function: forwarding, per-hop behavior, aggregation

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– Diffserv traffic classfication and conditioning

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

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Providing quality of service guarantees

Resouce reservation, call admission, call setup– Traffic characterization and specification of the desired QoS– Signaling for call setup– Pre-element call admission

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