計(jì)算機(jī)網(wǎng)絡(luò)課件：Chapter7

1、7: Multimedia Networking7-1Chapter 7: Multimedia Communication over IPV4 NetworkPrinciplesrclassify multimedia applicationsridentify network services applications needrmaking the best of best effort serviceProtocols and Architectures rspecific protocols for best-effort7: Multimedia Networking7-2MM N

2、etworking Applications Fundamental characteristics:rtypically delay sensitivemend-to-end delaymdelay jitter rloss tolerant: infrequent losses cause minor glitches rantithesis of data, which are loss intolerant but delay tolerant.Classes of MM applications:1) stored streaming2) live streaming3) inter

3、active, real-timeJitter is the variability of packet delays within the same packet stream7: Multimedia Networking7-3Streaming Stored Multimedia Stored streaming: rmedia stored at sourcertransmitted to clientrstreaming: client playout begins before all data has arrivedrtiming constraint for still-to-

4、be transmitted data: in time for playout7: Multimedia Networking7-4Streaming Stored Multimedia: What is it?1. videorecorded2. videosent3. video received,played out at clientCumulative datastreaming: at this time, client playing out early part of video, while server still sending laterpart of videone

5、tworkdelaytime7: Multimedia Networking7-5Streaming Stored Multimedia: InteractivityrVCR-like functionality: client can pause, rewind, FF, push slider barm10 sec initial delay OKm1-2 sec until command effect OKrtiming constraint for still-to-be transmitted data: in time for playout7: Multimedia Netwo

6、rking7-6Streaming Live MultimediaExamples:rInternet radio talk showrlive sporting eventStreaming (as with streaming stored multimedia)rplayback bufferrplayback can lag tens of seconds after transmissionrstill have timing constraintInteractivityrfast forward impossiblerrewind, pause possible!7: Multi

7、media Networking7-7Real-Time Interactive Multimedia rend-end delay requirements:maudio: 150 msec good, 400 msec OK includes application-level (packetization) and network delays higher delays noticeable, impair interactivityrsession initializationmhow does callee advertise its IP address, port number

8、, encoding algorithms?rapplications: IP telephony, video conference, distributed interactive worlds7: Multimedia Networking7-8Multimedia Over Todays InternetTCP/UDP/IP: “best-effort service”rno guarantees on delay, lossTodays Internet multimedia applications use application-level techniques to mitig

9、ate(as best possible) effects of delay, lossBut you said multimedia apps requiresQoS and level of performance to beeffective!?7: Multimedia Networking7-9About audio compressionranalog signal sampled and quantized at constant ratemtelephone: 64kbpsmCD music: 1.4MbpsmMP3: 96, 128, 160 kbpsmInternet te

10、lephony: 5.3 kbps and uprreceiver converts bits back to analog signal:msome quality reductionrBefore audio can be transmitted over a computer network, it must be digitized and compressedrAudio compressionmPCM:sampling rate:8Ksps and 8bit/samplemGSM:13Kbps；mG.729:8KbpsmG.723.1: 5.36.4Kbps7: Multimedi

11、a Networking7-10About video compressionrvideo: sequence of images displayed at constant rateme.g. 24 images/secrdigital image: array of pixelsmeach pixel represented by bitsrRedundancymspatial (within image)mtemporal (from one image to next)rBefore video can be transmitted over a computer network, i

12、t must be digitized and compressedm1024*1024 RGB results in 3MB of storage without compressionrVideo compressionrMPEG-XmMPEG 1 (CD-ROM) 1.5 MbpsmMPEG2 (DVD) 3-6 MbpsmMPEG4 (often used in Internet, 20 msec -talk spurt begins.rwith loss possible, receiver must look at both time stamps and sequence num

13、bers.mdifference of successive stamps 20 msec and sequence numbers without gaps - talk spurt begins.7: Multimedia Networking7-31Recovery from packet loss (1)Forward Error Correction (FEC): simple schemerfor every group of n chunks create redundant chunk by exclusive OR-ing n original chunksrsend out

14、 n+1 chunks, increasing bandwidth by factor 1/n.rcan reconstruct original n chunks if at most one lost chunk from n+1 chunksrplayout delay: enough time to receive all n+1 packetsrtradeoff: mincrease n, less bandwidth wastemincrease n, longer playout delaymincrease n, higher probability that 2 or mor

15、e chunks will be lost7: Multimedia Networking7-32Recovery from packet loss (2)2nd FEC schemeq “piggyback lower quality stream” q send lower resolutionaudio stream as redundant informationq e.g., nominal stream PCM at 64 kbpsand redundant streamGSM at 13 kbps.q whenever there is non-consecutive loss,

16、 receiver can conceal the loss. q can also append (n-1)st and (n-2)nd low-bit ratechunk7: Multimedia Networking7-33Recovery from packet loss (3)Interleavingrchunks divided into smaller unitsrfor example, four 5 msec units per chunkrpacket contains small units from different chunksrif packet lost, st

17、ill have most of every chunkrno redundancy overhead, but increases playout delay7: Multimedia Networking7-34Content distribution networks (CDNs)Content replicationrchallenging to stream large files (e.g., video) from single origin server in real timersolution: replicate content at hundreds of server

18、s throughout Internetmcontent downloaded to CDN servers ahead of timemplacing content “close” to user avoids impairments (loss, delay) of sending content over long pathsmCDN server typically in edge/access networkorigin server in North AmericaCDN distribution nodeCDN serverin S. AmericaCDN serverin

19、EuropeCDN serverin Asia7: Multimedia Networking7-35Content distribution networks (CDNs)Content replicationrCDN (e.g., Akamai) customer is the content provider (e.g., CNN)rCDN replicates customers content in CDN servers. rwhen provider updates content, CDN updates serversorigin server in North Americ

20、aCDN distribution nodeCDN serverin S. AmericaCDN serverin EuropeCDN serverin Asia7: Multimedia Networking7-36CDN exampleorigin server ()rdistributes HTMLrreplaces: http:/ with http:/ request for query for HTTP request for serverCDNs authoritative DNS server CDN server near clientCDN company ()rdistr

21、ibutes gif filesruses its authoritative DNS server to route redirect requestsclient7: Multimedia Networking7-37More about CDNsrouting requestsrCDN creates a “map”, indicating distances from leaf ISPs and CDN nodesrwhen query arrives at authoritative DNS server:mserver determines ISP from which query

22、 originatesmuses “map” to determine best CDN serverrCDN nodes create application-layer overlay network7: Multimedia Networking7-38Summary: Internet Multimedia: bag of tricksruse UDP to avoid TCP congestion control (delays) for time-sensitive trafficrclient-side adaptive playout delay: to compensate

23、for delayrserver side matches stream bandwidth to available client-to-server path bandwidthmchose among pre-encoded stream ratesmdynamic server encoding ratererror recovery (on top of UDP)mFEC, interleaving, error concealmentmretransmissions, time permittingrCDN: bring content closer to clients7: Mu

24、ltimedia Networking7-39Real-Time Protocol (RTP)rRTP specifies packet structure for packets carrying audio, video datarRFC 3550rRTP packet provides mpayload type identificationmpacket sequence numberingmtime stampingrRTP runs in end systemsrRTP packets encapsulated in UDP segmentsrinteroperability: i

25、f two Internet phone applications run RTP, then they may be able to work together7: Multimedia Networking7-40RTP runs on top of UDPRTP libraries provide transport-layer interface that extends UDP: port numbers, IP addresses payload type identification packet sequence numbering time-stamping7: Multim

26、edia Networking7-41RTP Examplerconsider sending 64 kbps PCM-encoded voice over RTP. rapplication collects encoded data in chunks, e.g., every 20 msec = 160 bytes in a chunk. raudio chunk + RTP header form RTP packet, which is encapsulated in UDP segment rRTP header indicates type of audio encoding i

27、n each packetm sender can change encoding during conference. rRTP header also contains sequence numbers, timestamps.7: Multimedia Networking7-42RTP and QoSrRTP does not provide any mechanism to ensure timely data delivery or other QoS guarantees. rRTP encapsulation is only seen at end systems (not)

28、by intermediate routers. mrouters providing best-effort service, making no special effort to ensure that RTP packets arrive at destination in timely matter. 7: Multimedia Networking7-43RTP HeaderPayload Type (7 bits): Indicates type of encoding currently being used. If sender changes encoding in mid

29、dle of conference, sender informs receiver via payload type field. Payload type 0: PCM mu-law, 64 kbpsPayload type 3, GSM, 13 kbpsPayload type 7, LPC, 2.4 kbpsPayload type 26, Motion JPEGPayload type 31. H.261Payload type 33, MPEG2 videoSequence Number (16 bits): Increments by one for each RTP packe

30、t sent, and may be used to detect packet loss and to restore packet sequence.7: Multimedia Networking7-44RTP Header (2)rTimestamp field (32 bytes long): sampling instant of first byte in this RTP data packetmfor audio, timestamp clock typically increments by one for each sampling period (for example

31、, each 125 usecs for 8 KHz sampling clock) mif application generates chunks of 160 encoded samples, then timestamp increases by 160 for each RTP packet when source is active. Timestamp clock continues to increase at constant rate when source is inactive.rSSRC field (32 bits long): identifies source

32、of t RTP stream. Each stream in RTP session should have distinct SSRC. 7: Multimedia Networking7-45Real-Time Control Protocol (RTCP)rworks in conjunction with RTP. reach participant in RTP session periodically transmits RTCP control packets to all other participants. reach RTCP packet contains sende

33、r and/or receiver reportsmreport statistics useful to application: # packets sent, # packets lost, interarrival jitter, etc.rfeedback can be used to control performancemsender may modify its transmissions based on feedback7: Multimedia Networking7-46RTCP - Continuedq each RTP session: typically a si

34、ngle multicast address; all RTP /RTCP packets belonging to session use multicast address.q RTP, RTCP packets distinguished from each other via distinct port numbers. q to limit traffic, each participant reduces RTCP traffic as number of conference participants increases 7: Multimedia Networking7-47R

35、TCP PacketsReceiver report packets:r fraction of packets lost, last sequence number, average interarrival jitterSender report packets: rSSRC of RTP stream, current time, number of packets sent, number of bytes sent Source description packets: re-mail address of sender, senders name, SSRC of associat

36、ed RTP stream rprovide mapping between the SSRC and the user/host name7: Multimedia Networking7-48Synchronization of StreamsrRTCP can synchronize different media streams within a RTP session rconsider videoconferencing app for which each sender generates one RTP stream for video, one for audio. rtim

37、estamps in RTP packets tied to the video, audio sampling clocksmnot tied to wall-clock timereach RTCP sender-report packet contains (for most recently generated packet in associated RTP stream):mtimestamp of RTP packet mwall-clock time for when packet was created. rreceivers uses association to sync

38、hronize playout of audio, video 7: Multimedia Networking7-49RTCP Bandwidth ScalingrRTCP attempts to limit its traffic to 5% of session bandwidth.Example rSuppose one sender, sending video at 2 Mbps. Then RTCP attempts to limit its traffic to 100 Kbps. rRTCP gives 75% of rate to receivers; remaining

39、25% to senderr75 kbps is equally shared among receivers: mwith R receivers, each receiver gets to send RTCP traffic at 75/R kbps. rsender gets to send RTCP traffic at 25 kbps. rparticipant determines RTCP packet transmission period by calculating avg RTCP packet size (across entire session) and divi

40、ding by allocated rate 7: Multimedia Networking7-50SIP: Session Initiation Protocol RFC 3261SIP long-term vision:rall telephone calls, video conference calls take place over Internetrpeople are identified by names or e-mail addresses, rather than by phone numbersryou can reach callee, no matter wher

41、e callee roams, no matter what IP device callee is currently using7: Multimedia Networking7-51SIP ServicesrSetting up a call, SIP provides mechanisms .mfor caller to let callee know she wants to establish a callmso caller, callee can agree on media type, encodingmto end callrdetermine current IP add

42、ress of callee:mmaps mnemonic identifier to current IP addressrcall management:madd new media streams during callmchange encoding during callminvite others mtransfer, hold calls7: Multimedia Networking7-52Setting up a call to known IP addressq Alices SIP invite message indicates her port number, IP

43、address, encoding she prefers to receive (PCM ulaw)q Bobs 200 OK message indicates his port number, IP address, preferred encoding (GSM)q SIP messages can be sent over TCP or UDP; here sent over RTP/UDP. qdefault SIP port number is 5060.timetimeBobsterminal ringsAlice4Bob9po

44、rt 5060port 38060Law audioGSMport 48753INVITE bob9c=IN IP4 4m=audio 38060 RTP/AVP 0port 5060200 OKc=IN IP4 9m=audio 48753 RTP/AVP 3ACKport 50607: Multimedia Networking7-53Setting up a call (more)rcodec negotiation:msuppose Bob doesnt have PCM ulaw encoder. mBob w

45、ill instead reply with 606 Not Acceptable Reply, listing his encoders Alice can then send new INVITE message, advertising different encoderrrejecting a callmBob can reject with replies “busy,” “gone,” “payment required,” “forbidden”rmedia can be sent over RTP or some other protocol7: Multimedia Netw

46、orking7-54Example of SIP messageINVITE sip: SIP/2.0Via: SIP/2.0/UDP 4From: sip:To: sip: Call-ID: Content-Type: application/sdpContent-Length: 885c=IN IP4 4m=audio 38060 RTP/AVP 0Notes:rHTTP message syntaxrsdp = session description protocolrCall-ID is unique for every call.q

47、 Here we dont know Bobs IP address. Intermediate SIPservers needed. q Alice sends, receives SIP messages using SIP default port 506q Alice specifies in Via:header that SIP client sends, receives SIP messages over UDP7: Multimedia Networking7-55Name translation and user locataionrcaller wants to call

48、 callee, but only has callees name or e-mail address.rneed to get IP address of callees current host:muser moves aroundmDHCP protocolmuser has different IP devices (PC, PDA, car device)rresult can be based on:m time of day (work, home)mcaller (dont want boss to call you at home)mstatus of callee (ca

49、lls sent to voicemail when callee is already talking to someone)Service provided by SIP servers:rSIP registrar serverrSIP proxy server7: Multimedia Networking7-56SIP RegistrarREGISTER sip: SIP/2.0Via: SIP/2.0/UDP 9 From: sip:To: sip:Expires: 3600rwhen Bob starts SIP client, client sends SIP REGISTER message to Bobs registrar server (similar function needed by Instant Messaging)Register Message:7: Multimedia Networking7-57SIP ProxyrAlice sends invite message to her proxy servermcontains address sip:rproxy responsible for routing SIP messages to calleempossibly throu