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1、Multimedia ApplicationsrMultimedia requirementsrStreamingrPhone over IPrRecovering from Jitter and LossrRTPrDiff-serv, Int-serv, RSVPApplication Classes rTypically sensitive to delay, but can tolerate packet loss (would cause minor glitches that can be concealed)rData contains audio and video conten
2、t (“continuous media”), three classes of applications:mStreamingmUnidirectional Real-TimemInteractive Real-TimeApplication Classes (more)rStreamingmClients request audio/video files from servers and pipeline reception over the network and displaymInteractive: user can control operation (similar to V
3、CR: pause, resume, fast forward, rewind, etc.)mDelay: from client request until display start can be 1 to 10 secondsApplication Classes (more)rUnidirectional Real-Time:msimilar to existing TV and radio stations, but delivery on the networkmNon-interactive, just listen/view rInteractive Real-Time :mP
4、hone conversation or video conferencemMore stringent delay requirement than Streaming and Unidirectional because of real-time naturemVideo: 150 msec acceptablemAudio: 150 msec good, 400 msec acceptableChallengesrTCP/UDP/IP suite provides best-effort, no guarantees on expectation or variance of packe
5、t delayrStreaming applications delay of 5 to 10 seconds is typical and has been acceptable, but performance deteriorate if links are congested (transoceanic)rReal-Time Interactive requirements on delay and its jitter have been satisfied by over-provisioning (providing plenty of bandwidth), what will
6、 happen when the load increases?.Challenges (more)rMost router implementations use only First-Come-First-Serve (FCFS) packet processing and transmission schedulingrTo mitigate impact of “best-effort” protocols, we can: mUse UDP to avoid TCP and its slow-start phasemBuffer content at client and contr
7、ol playback to remedy jittermAdapt compression level to available bandwidthSolution Approaches in IP NetworksrJust add more bandwidth and enhance caching capabilities (over-provisioning)!rNeed major change of the protocols :mIncorporate resource reservation (bandwidth, processing, buffering), and ne
8、w scheduling policies mSet up service level agreements with applications, monitor and enforce the agreements, charge accordinglyrNeed moderate changes (“Differentiated Services”):mUse two traffic classes for all packets and differentiate service accordinglymCharge based on class of packetsmNetwork c
9、apacity is provided to ensure first class packets incur no significant delay at routersStreamingrImportant and growing application due to reduction of storage costs, increase in high speed net access from homes, enhancements to caching and introduction of QoS in IP networksrAudio/Video file is segme
10、nted and sent over either TCP or UDP, public segmentation protocol: Real-Time Protocol (RTP)StreamingrUser interactive control is provided, e.g. the public protocol Real Time Streaming Protocol (RTSP)rHelper Application: displays content, which is typically requested via a Web browser; e.g. RealPlay
11、er; typical functions:mDecompressionmJitter removalmError correction: use redundant packets to be used for reconstruction of original streammGUI for user controlStreaming From Web ServersrAudio: in files sent as HTTP objectsrVideo (interleaved audio and images in one file, or two separate files and
12、client synchronizes the display) sent as HTTP object(s)rA simple architecture is to have the Browser requests the object(s) and after their reception pass them to the player for display- No pipeliningStreaming From Web Server (more)rAlternative: set up connection between server and player, then down
13、loadrWeb browser requests and receives a Meta File (a file describing the object) instead of receiving the file itself; rBrowser launches the appropriate Player and passes it the Meta File; rPlayer sets up a TCP connection with Web Server and downloads the fileMeta file requestsUsing a Streaming Ser
14、verrThis gets us around HTTP, allows a choice of UDP vs. TCP and the application layer protocol can be better tailored to Streaming; many enhancements options are possible (see next slide)Options When Using a Streaming ServerrUse UDP, and Server sends at a rate (Compression and Transmission) appropr
15、iate for client; to reduce jitter, Player buffers initially for 2-5 seconds, then starts displayrUse TCP, and sender sends at maximum possible rate under TCP; retransmit when error is encountered; Player uses a much large buffer to smooth delivery rate of TCPReal Time Streaming Protocol (RTSP)rFor u
16、ser to control display: rewind, fast forward, pause, resume, etcrOut-of-band protocol (uses two connections, one for control messages (Port 554) and for media stream)rRFC 2326 permits use of either TCP or UDP for the control messages connection, sometimes called the RTSP ChannelrAs before, meta file
17、 is communicated to web browser which then launches the Player; Player sets up an RTSP connection for control messages in addition to the connection for the streaming mediaMeta File ExampleTwister RTSP OperationRTSP Exchange Example C: SETUP rtsp:/ RTSP/1.0 Transport: rtp/udp; compression; port=3056
18、; mode=PLAY S: RTSP/1.0 200 1 OK Session 4231 C: PLAY rtsp:/ RTSP/1.0 Session: 4231 Range: npt=0- C: PAUSE rtsp:/ RTSP/1.0 Session: 4231 Range: npt=37 C: TEARDOWN rtsp:/ RTSP/1.0 Session: 4231 S: 200 3 OKReal-Time (Phone) Over IPs Best-EffortrInternet phone applications generate packets during talk
19、spurtsrBit rate is 8 KBytes, and every 20 msec, the sender forms a packet of 160 Bytes + a header to be discussed belowrThe coded voice information is encapsulated into a UDP packet and sent out; some packets may be lost; up to 20 % loss is tolerable; using TCP eliminates loss but at a considerable
20、cost: variance in delay; FEC is sometimes used to fix errors and make up lossesReal-Time (Phone) Over IPs Best-EffortrEnd-to-end delays above 400 msec cannot be tolerated; packets that are that delayed are ignored at the receiverrDelay jitter is handled by using timestamps, sequence numbers, and del
21、aying playout at receivers either a fixed or a variable amountrWith fixed playout delay, the delay should be as small as possible without missing too many packets; delay cannot exceed 400 msecInternet Phone with Fixed Playout DelayAdaptive Playout DelayrObjective is to use a value for p-r that track
22、s the network delay performance as it varies during a phone callrThe playout delay is computed for each talk spurt based on observed average delay and observed deviation from this average delayrEstimated average delay and deviation of average delay are computed in a manner similar to estimates of RT
23、T and deviation in TCPrThe beginning of a talk spurt is identified from examining the timestamps in successive and/or sequence numbers of chunksRecovery From Packet LossrLoss is in a broader sense: packet never arrives or arrives later than its scheduled playout timerSince retransmission is inapprop
24、riate for Real Time applications, FEC or Interleaving are used to reduce loss impact.rFEC is Forward Error CorrectionrSimplest FEC scheme adds a redundant chunk made up of exclusive OR of a group of n chunks; redundancy is 1/n; can reconstruct if at most one lost chunk; playout time schedule assumes
25、 a loss per groupRecovery From Packet LossrMixed quality streams are used to include redundant duplicates of chunks; upon loss playout available redundant chunk, albeit a lower quality onerWith one redundant chunk per chunk can recover from single lossesPiggybacking Lower Quality StreamInterleavingr
26、Has no redundancy, but can cause delay in playout beyond Real Time requirementsrDivide 20 msec of audio data into smaller units of 5 msec each and interleaverUpon loss, have a set of partially filled chunksReal-Time Protocol (RTP)rProvides standard packet format for real-time applicationrTypically r
27、uns over UDPrSpecifies header fields belowrPayload Type: 7 bits, providing 128 possible different types of encoding; eg PCM, MPEG2 video, etc.rSequence Number: 16 bits; used to detect packet lossReal-Time Protocol (RTP)rTimestamp: 32 bytes; gives the sampling instant of the first audio/video byte in
28、 the packet; used to remove jitter introduced by the networkrSynchronization Source identifier (SSRC): 32 bits; an id for the source of a stream; assigned randomly by the sourceRTP Control Protocol (RTCP)rProtocol specifies report packets exchanged between sources and destinations of multimedia info
29、rmationrThree reports are defined: Receiver reception, Sender, and Source descriptionrReports contain statistics such as the number of packets sent, number of packets lost, inter-arrival jitterrUsed to modify sender transmission rates and for diagnostics purposesRTCP Bandwidth ScalingrIf each receiv
30、er sends RTCP packets to all other receivers, the traffic load resulting can be largerRTCP adjusts the interval between reports based on the number of participating receiversrTypically, limit the RTCP bandwidth to 5% of the session bandwidth, divided between the sender reports (25%) and the receiver
31、s reports (75%)Improving QOS in IP NetworksrIETF groups are working on proposals to provide better QOS control in IP networks, i.e., going beyond best effort to provide some assurance for QOSrWork in Progress includes RSVP, Differentiated Services, and Integrated ServicesrSimple model for sharing an
32、d congestion studies:Principles for QOS GuaranteesrConsider a phone application at 1Mbps and an FTP application sharing a 1.5 Mbps link. mbursts of FTP can congest the router and cause audio packets to be dropped. mwant to give priority to audio over FTPrPRINCIPLE 1: Marking of packets is needed for
33、 router to distinguish between different classes; and new router policy to treat packets accordinglyPrinciples for QOS Guarantees (more)rApplications misbehave (audio sends packets at a rate higher than 1Mbps assumed above); rPRINCIPLE 2: provide protection (isolation) for one class from other class
34、es rRequire Policing Mechanisms to ensure sources adhere to bandwidth requirements; Marking and Policing need to be done at the edges:Principles for QOS Guarantees (more)rAlternative to Marking and Policing: allocate a set portion of bandwidth to each application flow; can lead to inefficient use of
35、 bandwidth if one of the flows does not use its allocationrPRINCIPLE 3: While providing isolation, it is desirable to use resources as efficiently as possiblePrinciples for QOS Guarantees (more)rCannot support traffic beyond link capacityrPRINCIPLE 4: Need a Call Admission Process; application flow
36、declares its needs, network may block call if it cannot satisfy the needs Summary Scheduling And Policing MechanismsrScheduling: choosing the next packet for transmission on a link can be done following a number of policies;rFIFO: in order of arrival to the queue; packets that arrive to a full buffe
37、r are either discarded, or a discard policy is used to determine which packet to discard among the arrival and those already queuedScheduling PoliciesrPriority Queuing: classes have different priorities; class may depend on explicit marking or other header info, eg IP source or destination, TCP Port
38、 numbers, etc.rTransmit a packet from the highest priority class with a non-empty queuerPreemptive and non-preemptive versionsScheduling Policies (more)rRound Robin: scan class queues serving one from each class that has a non-empty queueScheduling Policies (more)rWeighted Fair Queuing: is a general
39、ized Round Robin in which an attempt is made to provide a class with a differentiated amount of service over a given period of timePolicing MechanismsrThree criteria: m(Long term) Average Rate (100 packets per sec or 6000 packets per min?), crucial aspect is the interval lengthmPeak Rate: e.g., 6000
40、 p p minute Avg and 1500 p p sec Peakm(Max.) Burst Size: Max. number of packets sent consecutively, ie over a short period of timePolicing MechanismsrToken Bucket mechanism, provides a means for limiting input to specified Burst Size and Average Rate. Policing Mechanisms (more)rBucket can hold b tok
41、ens; token are generated at a rate of r token/sec unless bucket is full of tokens.rOver an interval of length t, the number of packets that are admitted is less than or equal to (r t + b).rToken bucket and WFQ can be combined to provide upperbound on delay.Integrated ServicesrAn architecture for pro
42、viding QOS guarantees in IP networks for individual application sessionsrrelies on resource reservation, and routers need to maintain state info (Virtual Circuit?), maintaining records of allocated resources and responding to new Call setup requests on that basisCall AdmissionrSession must first dec
43、lare its QOS requirement and characterize the traffic it will send through the networkrR-spec: defines the QOS being requestedrT-spec: defines the traffic characteristicsrA signaling protocol is needed to carry the R-spec and T-spec to the routers where reservation is required; RSVP is a leading can
44、didate for such signaling protocolCall AdmissionrCall Admission: routers will admit calls based on their R-spec and T-spec and base on the current resource allocated at the routers to other calls.Integrated Services: ClassesrGuaranteed QOS: this class is provided with firm bounds on queuing delay at
45、 a router; envisioned for hard real-time applications that are highly sensitive to end-to-end delay expectation and variancerControlled Load: this class is provided a QOS closely approximating that provided by an unloaded router; envisioned for todays IP network real-time applications which perform
46、well in an unloaded networkDifferentiated ServicesrIntended to address the following difficulties with Intserv and RSVP;rScalability: maintaining states by routers in high speed networks is difficult sue to the very large number of flows rFlexible Service Models: Intserv has only two classes, want t
47、o provide more qualitative service classes; want to provide relative service distinction (Platinum, Gold, Silver, )rSimpler signaling: (than RSVP) many applications and users may only w ant to specify a more qualitative notion of serviceDifferentiated ServicesrApproach: mOnly simple functions in the
48、 core, and relatively complex functions at edge routers (or hosts)mDo not define service classes, instead provides functional components with which service classes can be builtEdge FunctionsrAt DS-capable host or first DS-capable routerrClassification: edge node marks packets according to classifica
49、tion rules to be specified (manually by admin, or by some TBD protocol)rTraffic Conditioning: edge node may delay and then forward or may discardCore FunctionsrForwarding: according to “Per-Hop-Behavior” or PHB specified for the particular packet class; such PHB is strictly based on class marking (n
50、o other header fields can be used to influence PHB)rBIG ADVANTAGE:No state info to be maintained by routers!Classification and ConditioningrPacket is marked in the Type of Service (TOS) in IPv4, and Traffic Class in IPv6r6 bits used for Differentiated Service Code Point (DSCP) and determine PHB that
51、 the packet will receiver2 bits are currently unusedClassification and ConditioningrIt may be desirable to limit traffic injection rate of some class; user declares traffic profile (eg, rate and burst size); traffic is metered and shaped if non-conforming Forwarding (PHB)rPHB result in a different o
52、bservable (measurable) forwarding performance behaviorrPHB does not specify what mechanisms to use to ensure required PHB performance behaviorrExamples: mClass A gets x% of outgoing link bandwidth over time intervals of a specified lengthmClass A packets leave first before packets from class BForwar
53、ding (PHB)rPHBs under consideration:mExpedited Forwarding: departure rate of packets from a class equals or exceeds a specified rate (logical link with a minimum guaranteed rate)mAssured Forwarding: 4 classes, each guaranteed a minimum amount of bandwidth and buffering; each with three drop preference partitionsDifferentiated Services IssuesrAF and EF are not even in a standard track yet research ongoingr“Virtual Leased lines” and “Olympic” services are being discussed rImpact of crossing multiple ASs and routers that are not DS-capable
