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Voice-TFCC: a TCP-Friendly Congestion Control scheme for VoIP flows

Voice-TFCC: a TCP-Friendly Congestion Control scheme for VoIP flows. Hossam Afifi National Institute of Telecommunications INT Paris, France hossam.afifi@int-evry.fr. Abdelbasset Trad PRINCE Computer Science Research Unit INFCOM Sousse, Tunisia trad.abdelbasset@gmail.com. PIMRC 2008

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Voice-TFCC: a TCP-Friendly Congestion Control scheme for VoIP flows

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  1. Voice-TFCC: a TCP-Friendly Congestion Controlscheme for VoIP flows Hossam Afifi National Institute of Telecommunications INT Paris, France hossam.afifi@int-evry.fr Abdelbasset Trad PRINCE Computer Science Research Unit INFCOM Sousse, Tunisia trad.abdelbasset@gmail.com PIMRC 2008 15 Septembre 2008

  2. Presentation Outline • Motivations & Objectives • Studied Network Architecture • Congestion control for VoIP traffic • TCP-friendly Equation-based Rate Control • Our proposal: Voice-TFCC scheme • Architecture • Algorithm • Analysis • Conclusions and Perspectives PIMRC 2008

  3. Motivations of our Work • VoIP is one of the fastest growing Internet applications • The Internet is expected to carry a significant proportion of the world’s telephony traffic • New performance limitations have rised: • Scale in number of VoIP communications PIMRC 2008

  4. Our Objectives • Address the tradeoffs between efficiency and end-to-end overall performance of a large number of VoIP communications • Design adaptive congestion control mechanisms aiming to: • Maximize the overall VoIP transmission quality • Use network resources efficiently • Compete fairly with other Internet traffic (TCP) PIMRC 2008

  5. Studied Network Architecture RTP Voice Flows PSTN Mobile Network IP Network VoIP GW VoIPGW PSTN IP Phone • Large number of VoIP sources at an access network destined to different users in remote networks (PC to phone/Phone to Phone…) • VoIP communications scharing a common path between peer VoIP gateways PIMRC 2008

  6. Protocol Header Overhead • Typical payload duration: 20 ms • IP/UDP/RTP encapsulation • Minimum header length= 40 bytes • Generated Header throughput: 16 kbps Significant overhead IPv6 IPv4 PIMRC 2008

  7. Congestion Control for VoIP Traffic • Voice traffic is typically deployed as best-effort traffic • VoIP lacks effective and scalable congestion control • Performance limitations: • Inefficient use of network bandwidth (IP/UDP/RTP protocol headers) • Fairness problem with TCP traffic caused by the transmission of large number of uncontrolled UDP bursts of small VoIP packets • UDP traffic is unresponsive to congestion and can completely monopolize available bandwidth PIMRC 2008

  8. Protocol Header Overhead: Solutions • Reduce the header size: • IP/UDP/RTP Header compression (cRTP, RFC 2508) • - designed for low speed serial links • reduces IP/UDP/RTP header to 2 bytes (no UDP checksums) • applied on a single RTP flow • based on differential coding mechanism • reliability ensured by lower layers • Encapsulate several packets into one header (Multiplexing) • multiplex several RTP streams between two gateways • fixed number of flows to multiplex • introduces delays (mutlipexing delay, queuing delays) PIMRC 2008

  9. TCP-friendly Equation-based Rate Control • Unresponsive flows: • Do not use end-to-end congestion control • D o not reduce their load on the network when subjected to packet drops • Multimedia applications using unresponsive transport protocols (RTP/UDP) • TCP-friendly equation-based rate control • Introduced to ensure proper congestion avoidance for multimedia applications • Steady state TCP throughput approximation • Smoothly find available bandwidth • Do not halve the sending rate in response to a single loss • Increase the sending rate slowly in response to a decrease of the loss rate PIMRC 2008

  10. TFRC throughput as function of RTT and drop rate TFRC Throughput Payload size S=1460 bytes PIMRC 2008

  11. Our Proposal: Voice-TFCC Scheme • Voice TCP-Friendly Congestion Control Scheme • Novel generic scheme that controls both Packet and Codec rate of VoIP flows while maintaining a TCP-friendly throughput • Based on TCP-friendly decision: • Packet rate is adjusted by multiplexing several RTP VoIP flows over a single stream • Codec rate is also adapted using different audio codecs PIMRC 2008

  12. RTP header Payload IP UDP Voice 1 IP UDP Voice n IP UDP Voice 2 … .. Voice n Voice 1 Voice 2 IP UDP Basic RTP Multiplexing Scheme Source 1 Source 2 Source n Adjustable Aggregate Buffer • Basic assumption: VoIP flows from different sources accur simultanousely at the sender VoIP GW • Number of packets to multiplex ? PIMRC 2008

  13. Voice-TFCC Architecture • Transcoding module incorporated at the Voice-TFCC sender gateway: • - to handle homogeneous flows using the same voice codec rate • - to adapt this rate according toTCP-friendly decision PIMRC 2008

  14. Voice-TFCC Algorithm Initially: m0=1 Case of one VoIP flow Phase I: Reduce packet rate by multiplexing Phase II: Reduce codec rate also • How to determine the new packet and codec rate • to be used in the next time interval (i, i+1) ? PIMRC 2008

  15. Voice-TFCC Algorithm Basic equation: Phase I: Reduce packet rate by multiplexing Phase II: Reduce codec rate TFRC throughput formula PIMRC 2008

  16. Voice-TFCC Analysis: Bandwidth Saving • Bandwidth saved by multiplexing m RTP voice packets: where Payload=20 bytes Payload=160 bytes • Multiplexed packet size is bounded by the MTU PIMRC 2008

  17. Congestion control mechanisms for VoIP flows represents a promising solution to prevent the performance degradation of voice and TCP traffic. • Proposed Voice-TFCC mechanism dynamically adapts packet and codec rate of VoIP flows while being fair with coexisting Internet traffic • A promising extention to Voice-TFCC scheme: • Study the case of high traffic load caused by a large number of flows that can not be multiplexed within TCP-friendly flow • Path switching mechanism: incoming VoIP flows are redirected towards a GW presenting better network path conditions (signaling protocols like SIP can be used) Conclusions & Perspectives

  18. Thank you ! Questions ? Dr. A. Trad

  19. Voice-TFCC: Experimental Results • Prototype implementation of VoIP traffic generation based on UDP sockets. • An adaptive system that can switch between five bit-rates: Different payload sizes • PlanetLab hosts used to emulate VoIP GWs • Feedback reports are sent from destination to sender host evry 5 seconds • Initially 10 flows sent from host1 to host2 using G.711 codec PIMRC 2008

  20. Voice-TFCC: Experimental Results P • Introduction Voice-TFCC PIMRC 2008

  21. MOS and Voice-TFCC/TCP-friendly rate difference P • Introduction • TCP-friendliness condition achieved PIMRC 2008

  22. Experimental Results summary: Voice-TFCC mechanism • Introduction • Slight loss rate and jitter increase • Significant delay and quality Improvement PIMRC 2008

  23. VoIP variant of TFRC (Floyd et al.) • TFRC variant for applications that transmit small packets • Assumption: • Network bandwidth limitation in bytes/sec rather than in packets/sec for VoIP traffic • Design goal: • Achieve the same bandwidth in bytes/sec as a TCP flow using 1500 – byte data packets • Penalize VoIP applications that send small payload packets which increasing header overhead • Reduce the sending rate (rate reduction factor)

  24. Voice packet size TFRC throughput Header size VoIP variant of TFRC • The lower the payload size is the more • sending rate will be reduced

  25. Packet-based vs. Byte-based Environments: Floyd’s Simulation Results • Introduction • 5 TCP connections and 5 VoIP TFRC connections sharing a 3 Mbps link • In packet based environments each packet requires a single buffer and the decision to drop a packet is independent of the packet size • In byte-based environments small VoIP packets encounter less packet drops than TCP

  26. Packet-based vs. Byte-based Environments: Floyd’s Simulation Results • If the bottleneck link is in units of bytes rather than in packets: • Fairness results change significantly • VoIP TFRC flow sees a much smaller drop rate than TCP flow • Consequently VoIP flow receives a much larger sending rate

  27. Experimental Results summary: Voice-TFCC mechanism Phase I: without codec rate adaptation

  28. Introduction: VoIP Networks • Initial interest of communication cost reduction (entreprise telephone networks, long-distance calls) • Now considered as networks that will replace the telephone network • The base for the next generation of multimedia communications • Flexibility of IP-based packet switched networks • Convergence of data (packet switched) and Voice (traditionally circuit switched) into a single IP-based core architecture. • A single converged network for voice and data will be used • VoIP services are being increasingly offered to end users (e.g. Skype) PIMRC 2008

  29. Overview: VoIP Quality Assessment • ITU-T E-Model (G.107 Recommendation) Delay Impairment Codec and loss Impairments • MOS (Mean Opinion Score) PIMRC 2008

  30. Overview: VoIP Quality Assessment (2) Effect of packet loss Best Intrinsic codec quality 150 ms Effect of delay G.711 PIMRC 2008

  31. Overview: VoIP Quality Improvements • Research work focused on enhancing the low VoIP quality related to intrinsic properties of IP networks • Network QoS approach (DiffServ/IntServ) • End-to-end mechanisms (Application level) • FEC (Forward Error Correction) • Playout buffer mechanisms (alleviate the jitter effect) • Adaptive mechanisms • - End systems measure the service being delivered by the • network (using RTCP) • - Adapt their behavior according to packet delays and losses • - Adaptive FEC/Playout buffer mechanisms PIMRC 2008

  32. Overview: VoIP System • Voice Codecs for analog voice digitization and compression • Different techniques & different features • Voice transport over best-effort IP networks • RTP/RTCP over UDP • No performance garantees PIMRC 2008

  33. Introduction: VoIP main Advantages • Cost saving • - low Internet communication cost (packet switching technology) • toll charges associated with PSTN networks are reduced • reduced administration cost of a converged network • Efficiency • - VoIP achieves more efficiency than the circuit-switched voice transmission • VoIP dramatically improves bandwidth efficiency • (advanced voice compression techniques, silence suppression) • Integration of voice and data networks • integrated networks intend to provide voice transmission quality • and reliability of PSTN networks • combine voice communications with other media (e.g. video) PIMRC 2008

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