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11: Multimedia Networking 11-1 Chapter 11 Multimedia Networking A note on the use of these ppt slides: We’re making these slides freely available to all (faculty, students, readers). They’re in PowerPoint form so you can add, modify, and delete slides (including this one) and slide content to suit your needs. They obviously represent a lot of work on our part. In return for use, we only ask the following: If you use these slides (e.g., in a class) in substantially unaltered form, that you mention their source (after all, we’d like people to use our book!) If you post any slides in substantially unaltered form on a www site, that you note that they are adapted from (or perhaps identical to) our slides, and note our copyright of this material. Thanks and enjoy! JFK / KWR All material copyright 1996-2009 J.F Kurose and K.W. Ross, All Rights Reserved Computer Networking: A Top Down Approach 5 th edition. Jim Kurose, Keith Ross Addison-Wesley, April 2009.
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11: Multimedia Networking 11-2 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
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11: Multimedia Networking 11-3 Chapter 7: goals Principles r classify multimedia applications r identify network services applications need r making the best of best effort service Protocols and Architectures r specific protocols for best-effort r mechanisms for providing QoS r architectures for QoS
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11: Multimedia Networking 11-4 Chapter 7 outline 7.1 multimedia networking applications 7.2 streaming stored audio and video 7.3 making the best out of best effort service 7.4 providing multiple classes of service 7.5 providing QoS guarantees
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11: Multimedia Networking 11-5 MM Networking Applications Fundamental characteristics: r typically delay sensitive m end-to-end delay m delay jitter r loss tolerant: infrequent losses cause minor glitches r antithesis of data, which are loss intolerant but delay tolerant. Classes of MM applications: 1) stored streaming 2) live streaming 3) interactive, real-time Jitter is the variability of packet delays within the same packet stream
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11: Multimedia Networking 11-6 Streaming Stored Multimedia Stored streaming: r media stored at source r transmitted to client r streaming: client playout begins before all data has arrived r timing constraint for still-to-be transmitted data: in time for playout r so collect ~10 sec buffer before playout starts
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11: Multimedia Networking 11-7 Streaming Stored Multimedia: What is it? 1. video recorded 2. video sent 3. video received, played out at client Cumulative data streaming: at this time, client playing out early part of video, while server still sending later part of video network delay time
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11: Multimedia Networking 11-8 Streaming Stored Multimedia: Interactivity r VCR-like functionality: client can pause, rewind, FF, push slider bar m 10 sec initial delay OK m 1-2 sec until command effect OK r timing constraint for still-to-be transmitted data: in time for playout
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11: Multimedia Networking 11-9 Streaming Live Multimedia Examples: r Internet radio talk show r live sporting event Streaming (as with streaming stored multimedia) r playback buffer r playback can lag tens of seconds after transmission r timing constraint Interactivity r fast forward impossible r rewind, pause possible!
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11: Multimedia Networking 11-10 Real-Time Interactive Multimedia r end-end delay requirements: m audio: < 150 msec good, < 400 msec OK includes application-level (packetization) and network delays higher delays noticeable, impair interactivity r session initialization m coordinate IP address, port number, encoding algorithms? Signaling protocols (SIP, SKYPE) r applications: IP telephony, video conference, distributed interactive worlds
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11: Multimedia Networking 11-11 Multimedia Over Today’s Internet TCP/UDP/IP: “best-effort service” r no guarantees on delay, loss Today’s Internet multimedia applications use application-level techniques to mitigate (as best possible) effects of delay, loss But you said multimedia apps requires QoS and level of performance to be effective! ? ? ?? ? ? ? ? ? ? ?
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11: Multimedia Networking 11-12 How should the Internet evolve to better support multimedia? Integrated services philosophy: r fundamental changes in Internet so that apps can reserve end-to-end bandwidth r requires new, complex software in hosts & routers Laissez-faire r no major changes r more bandwidth when needed r content distribution, application-layer multicast m application layer Differentiated services philosophy: r fewer changes to Internet infrastructure, yet provide 1st and 2nd class service What’s your opinion?
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11: Multimedia Networking 11-13 Chapter 7 outline 7.1 multimedia networking applications 7.2 streaming stored audio and video 7.3 making the best out of best effort service 7.4 providing multiple classes of service 7.5 providing QoS guarantees
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11: Multimedia Networking 11-14 Streaming Stored Multimedia application-level streaming techniques for making the best out of best effort service: m client-side buffering m use of UDP versus TCP m multiple encodings of multimedia r jitter removal r decompression r error concealment r graphical user interface w/ controls for interactivity Media Player
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11: Multimedia Networking 11-15 constant bit rate video transmission Cumulative data time variable network delay client video reception constant bit rate video playout at client client playout delay buffered video Streaming Multimedia: Client Buffering r client-side buffering, playout delay compensate for network-added delay, delay jitter
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11: Multimedia Networking 11-16 Streaming Multimedia: UDP or TCP? UDP r server sends at rate appropriate for client (oblivious to network congestion !) m often send rate = encoding rate = constant rate m then, fill rate = constant rate - packet loss r short playout delay (2-5 seconds) to remove network jitter r error recover: time permitting TCP r send at maximum possible rate under TCP r fill rate fluctuates due to TCP congestion control r larger playout delay: smooth TCP delivery rate r HTTP/TCP passes more easily through firewalls
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11: Multimedia Networking 11-17 User Control of Streaming Media: RTSP HTTP r does not target multimedia content r no commands for fast forward, etc. RTSP: RFC 2326 r client-server application layer protocol r user control: rewind, fast forward, pause, resume, repositioning, etc… What it doesn’t do: r doesn’t define how audio/video is encapsulated for streaming over network r doesn’t restrict how streamed media is transported (UDP or TCP possible) r doesn’t specify how media player buffers audio/video
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11: Multimedia Networking 11-18 Chapter 7 outline 7.1 multimedia networking applications 7.2 streaming stored audio and video 7.3 making the best out of best effort service 7.4 providing multiple classes of service 7.5 providing QoS guarantees
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11: Multimedia Networking 11-19 Internet Phone: Packet Loss and Delay r network loss: IP datagram lost due to network congestion (router buffer overflow) r delay loss: IP datagram arrives too late for playout at receiver m delays: processing, queueing in network; end- system (sender, receiver) delays m typical maximum tolerable delay: 400 ms r loss tolerance: depending on voice encoding, losses concealed, packet loss rates between 1% and 10% can be tolerated.
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11: Multimedia Networking 11-20 Internet Phone: Fixed Playout Delay r receiver attempts to playout each chunk exactly q msecs after chunk was generated. m chunk has time stamp t: play out chunk at t+q. m chunk arrives after t+q: data arrives too late for playout, data “lost” r tradeoff in choosing q: m large q: less packet loss m small q: better interactive experience
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11: Multimedia Networking 11-21 Fixed Playout Delay sender generates packets every 20 msec during talk spurt. first packet received at time r first playout schedule: begins at p second playout schedule: begins at p’
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11: Multimedia Networking 11-22 Adaptive Playout Delay (1) dynamic estimate of average delay at receiver: where u is a fixed constant (e.g., u =.01). r Goal: minimize playout delay, keeping late loss rate low r Approach: adaptive playout delay adjustment: m estimate network delay, adjust playout delay at beginning of each talk spurt. m silent periods compressed and elongated. m chunks still played out every 20 msec during talk spurt.
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11: Multimedia Networking 11-23 Adaptive playout delay (2) also useful to estimate average deviation of delay, v i : estimates d i, v i calculated for every received packet (but used only at start of talk spurt for first packet in talk spurt, playout time is: where K is positive constant remaining packets in talkspurt are played out periodically
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11: Multimedia Networking 11-24 Adaptive Playout (3) Q: How does receiver determine whether packet is first in a talkspurt? r if no loss, receiver looks at successive timestamps. m difference of successive stamps > 20 msec -->talk spurt begins. r with loss possible, receiver must look at both time stamps and sequence numbers. m difference of successive stamps > 20 msec and sequence numbers without gaps --> talk spurt begins.
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11: Multimedia Networking 11-25 Recovery from packet loss (1) Forward Error Correction (FEC): simple scheme r for every group of n chunks create redundant chunk by exclusive OR-ing n original chunks r send out n+1 chunks, increasing bandwidth by factor 1/n. r can reconstruct original n chunks if at most one lost chunk from n+1 chunks r playout delay: enough time to receive all n+1 packets r tradeoff: m increase n, less bandwidth waste m increase n, longer playout delay m increase n, higher probability that 2 or more chunks will be lost
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11: Multimedia Networking 11-26 Recovery from packet loss (2) 2nd FEC scheme “piggyback lower quality stream” send lower resolution audio stream as redundant information e.g., nominal stream PCM at 64 kbps and redundant stream GSM at 13 kbps. whenever there is non-consecutive loss, receiver can conceal the loss. can also append (n-1)st and (n-2)nd low-bit rate chunk
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11: Multimedia Networking 11-27 Recovery from packet loss (3) Interleaving r chunks divided into smaller units r for example, four 5 msec units per chunk r packet contains small units from different chunks r if packet lost, still have most of every chunk r no redundancy overhead, but increases playout delay
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11: Multimedia Networking 11-28 Content distribution networks (CDNs) Content replication r challenging to stream large files (e.g., video) from single origin server in real time r solution: replicate content at hundreds of servers throughout Internet m content downloaded to CDN servers ahead of time m placing content “close” to user avoids impairments (loss, delay) of sending content over long paths m CDN server typically in edge/access network origin server in North America CDN distribution node CDN server in S. America CDN server in Europe CDN server in Asia
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11: Multimedia Networking 11-29 Content distribution networks (CDNs) Content replication r CDN (e.g., Akamai) customer is the content provider (e.g., CNN) r CDN replicates customers’ content in CDN servers. r when provider updates content, CDN updates servers origin server in North America CDN distribution node CDN server in S. America CDN server in Europe CDN server in Asia
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11: Multimedia Networking 11-30 CDN example origin server (www.foo.com) r distributes HTML r replaces: http://www.foo.com/sports.ruth.gif with h ttp://www.cdn.com/www.foo.com/sports/ruth.gif HTTP request for www.foo.com/sports/sports.html DNS query for www.cdn.com HTTP request for www.cdn.com/www.foo.com/sports/ruth.gif 1 2 3 origin server CDN’s authoritative DNS server CDN server near client CDN company (cdn.com) r distributes gif files r uses its authoritative DNS server to route redirect requests client
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11: Multimedia Networking 11-31 More about CDNs routing requests r CDN creates a “map”, indicating distances from leaf ISPs and CDN nodes r when query arrives at authoritative DNS server: m server determines ISP from which query originates m uses “map” to determine best CDN server r CDN nodes create application-layer overlay network
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11: Multimedia Networking 11-32 Chapter 7 outline 7.1 multimedia networking applications 7.2 streaming stored audio and video 7.3 making the best out of best effort service 7.4 providing multiple classes of service 7.5 providing QoS guarantees
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11: Multimedia Networking 11-33 Providing Multiple Classes of Service r thus far: making the best of best effort service m one-size fits all service model r alternative: multiple classes of service m partition traffic into classes m network treats different classes of traffic differently (analogy: VIP service vs regular service) 0111 r granularity: differential service among multiple classes, not among individual connections r history: ToS bits
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11: Multimedia Networking 11-34 Multiple classes of service: scenario R1 R2 H1 H2 H3 H4 1.5 Mbps link R1 output interface queue
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11: Multimedia Networking 11-35 Scenario 1: mixed FTP and audio r Example: 1Mbps IP phone, FTP share 1.5 Mbps link. m bursts of FTP can congest router, cause audio loss m want to give priority to audio over FTP packet marking needed for router to distinguish between different classes; and new router policy to treat packets accordingly Principle 1 R1 R2
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11: Multimedia Networking 11-36 Principles for QOS Guarantees (more) r what if applications misbehave (audio sends higher than declared rate) m policing: force source adherence to bandwidth allocations r marking and policing at network edge: m similar to ATM UNI (User Network Interface) provide protection (isolation) for one class from others Principle 2 R1 R2 1.5 Mbps link 1 Mbps phone packet marking and policing
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11: Multimedia Networking 11-37 Principles for QOS Guarantees (more) r Allocating fixed (non-sharable) bandwidth to flow: inefficient use of bandwidth if flows doesn’t use its allocation While providing isolation, it is desirable to use resources as efficiently as possible Principle 3 R1 R2 1.5 Mbps link 1 Mbps phone 1 Mbps logical link 0.5 Mbps logical link
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11: Multimedia Networking 11-38 Scheduling And Policing Mechanisms r scheduling: choose next packet to send on link r FIFO (first in first out) scheduling: send in order of arrival to queue m real-world example? m discard policy: if packet arrives to full queue: who to discard? Tail drop: drop arriving packet priority: drop/remove on priority basis random: drop/remove randomly
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11: Multimedia Networking 11-39 Scheduling Policies: more Priority scheduling: transmit highest priority queued packet r multiple classes, with different priorities m class may depend on marking or other header info, e.g. IP source/dest, port numbers, etc.. m Real world example?
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11: Multimedia Networking 11-40 Scheduling Policies: still more round robin scheduling: r multiple classes r cyclically scan class queues, serving one from each class (if available) r real world example?
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11: Multimedia Networking 11-41 Scheduling Policies: still more Weighted Fair Queuing: r generalized Round Robin r each class gets weighted amount of service in each cycle r real-world example?
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11: Multimedia Networking 11-42 Policing Mechanisms Goal: limit traffic to not exceed declared parameters Three common-used criteria of traffic rate: r (Long term) Average Rate: how many packets can be sent per unit time (in the long run) m crucial question: what is the interval length: 100 packets per sec or 6000 packets per min have same average! r Peak Rate: e.g., 6000 pkts per min. (ppm) avg.; 15000 ppm peak rate r (Max.) Burst Size: max. number of packets sent consecutively (with no intervening idle)
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11: Multimedia Networking 11-43 Policing Mechanisms Token Bucket: limit input to specified Burst Size and Average Rate. r bucket can hold b tokens r tokens generated at rate r token/sec unless bucket full r over interval of length t: number of packets admitted less than or equal to (r t + b).
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11: Multimedia Networking 11-44 Edge router: per-flow traffic management marks packets as in-profile and out-profile Core router: per class traffic management buffering and scheduling based on marking at edge preference given to in-profile packets Diffserv Architecture scheduling... r b marking
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11: Multimedia Networking 11-45 Edge-router Packet Marking r class-based marking: packets of different classes marked differently r intra-class marking: conforming portion of flow marked differently than non-conforming one r profile: pre-negotiated rate A, bucket size B r packet marking at edge based on per-flow profile Possible usage of marking: User packets Rate A B
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11: Multimedia Networking 11-46 Classification and Conditioning r Packet is marked in the Type of Service (TOS) in IPv4, and Traffic Class in IPv6 r 6 bits used for Differentiated Service Code Point (DSCP) and determine per hop behavior (PHB) that the packet will receive r 2 bits are currently unused
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11: Multimedia Networking 11-47 Classification and Conditioning may be desirable to limit traffic injection rate of some class: r user declares traffic profile (e.g., rate, burst size) r traffic metered, shaped if non-conforming
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11: Multimedia Networking 11-48 Forwarding (PHB) r PHB result in a different observable (measurable) forwarding performance behavior r PHB does not specify what mechanisms to use to ensure required PHB performance behavior r Examples: m Class A gets x% of outgoing link bandwidth over time intervals of a specified length m Class A packets leave first before packets from class B
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11: Multimedia Networking 11-49 Forwarding (PHB) PHBs being developed: r Expedited Forwarding: pkt departure rate of a class equals or exceeds specified rate m logical link with a minimum guaranteed rate r Assured Forwarding: 4 classes of traffic m each guaranteed minimum amount of bandwidth m each with three drop preference partitions
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11: Multimedia Networking 11-50 Chapter 7 outline 7.1 multimedia networking applications 7.2 streaming stored audio and video 7.3 making the best out of best effort service 7.4 providing multiple classes of service 7.5 providing QoS guarantees: IntServ, RSVP
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11: Multimedia Networking 11-51 Principles for QOS Guarantees (more) r Basic fact of life: can not support traffic demands beyond link capacity Call Admission: flow declares its needs, network may block call (e.g., busy signal) if it cannot meet needs Principle 4 R1 R2 1.5 Mbps link 1 Mbps phone 1 Mbps phone
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11: Multimedia Networking 11-52 QoS guarantee scenario r Resource reservation m call setup, signaling (RSVP) m traffic, QoS declaration m per-element admission control m QoS-sensitive scheduling (e.g., WFQ) request/ reply
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11: Multimedia Networking 11-53 IETF Integrated Services r architecture for providing QOS guarantees in IP networks for individual application sessions r resource reservation: routers maintain state info (like VC) of allocated resources, per flow QoS req’s r admit/deny new call setup requests: Question: can newly arriving flow be admitted with performance guarantees while not violating QoS guarantees made to already admitted flows?
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11: Multimedia Networking 11-54 Call Admission Arriving session must : r declare its QOS requirement m R-spec: defines the QOS being requested r characterize traffic it will send into network m T-spec: defines traffic characteristics r signaling protocol: needed to carry R-spec and T- spec to routers (where reservation is required) m RSVP
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11: Multimedia Networking 11-55 Intserv QoS: Service models [rfc2211, rfc 2212] Guaranteed service: r worst case traffic arrival: leaky-bucket-policed source r simple (mathematically provable) bound on delay [Parekh 1992, Cruz 1988] Controlled load service: r "a quality of service closely approximating the QoS that same flow would receive from an unloaded network element." WFQ token rate, r bucket size, b per-flow rate, R D = b/R max arriving traffic
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11: Multimedia Networking 11-56 Signaling in the Internet connectionless (stateless) traffic forwarding by IP routers best effort service no network signaling protocols needed in initial IP design + = r New requirement: reserve resources along end-to-end path (end system, routers) for QoS for multimedia applications r RSVP: Resource Reservation Protocol [RFC 2205] m “ … allow users to communicate requirements to network in robust and efficient way.” i.e., signaling !
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11: Multimedia Networking 11-57 RSVP Design Goals 1. accommodate heterogeneous receivers (different bandwidth along paths) 2. accommodate different applications with different resource requirements 3. make multicast a first class service, with adaptation to multicast group membership 4. leverage existing multicast/unicast routing, with adaptation to changes in underlying unicast, multicast routes 5. control protocol overhead to grow (at worst) linear in # receivers 6. modular design for heterogeneous underlying technologies
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11: Multimedia Networking 11-58 RSVP: does not… r specify how resources are to be reserved r rather: a mechanism for communicating needs r determine routes packets will take r that’s the job of routing protocols r signaling decoupled from routing
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11: Multimedia Networking 11-59 RSVP: overview of operation r senders, receiver join a multicast group m done outside of RSVP m senders need not join group r sender-to-network signaling m path message: make sender presence known to routers m path teardown: delete sender’s path state from routers r receiver-to-network signaling m reservation message: reserve resources from sender(s) to receiver m reservation teardown: remove receiver reservations r network-to-end-system signaling m path error m reservation error
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11: Multimedia Networking 11-60 Chapter 7: Summary Principles r classify multimedia applications r making the best of best effort service Protocols and Architectures r specific protocols for best-effort r mechanisms for providing QoS r architectures for QoS m multiple classes of service m QoS guarantees, admission control
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