Voice-TFCC: a TCP-Friendly Congestion Control scheme for VoIP flows Abdelbasset Trad PRINCE Computer Science Research Unit INFCOM Sousse, Tunisia Hossam Afifi National Institute of Telecommunications INT Paris, France PIMRC 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 IP Network PSTN IP Phone VoIP GW PSTN Mobile Network RTP Voice Flows Studied Network Architecture 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 IPv4 IPv6 Significant overhead 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 Unresponsive flows: Do not use end-to-end congestion control Do 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 TCP-friendly Equation-based Rate Control PIMRC 2008

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

11 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 Our Proposal: Voice-TFCC Scheme PIMRC 2008

12 Basic RTP Multiplexing Scheme IP UDP Voice 1IP UDP Voice 2IP UDP Voice n Adjustable Aggregate Buffer Source 1 Source 2 Source n … Voice 1Voice 2 Voice n.. IP UDP RTP header Payload 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 to TCP-friendly decision PIMRC 2008

14 Voice-TFCC Algorithm 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) ? Initially: m 0 =1 PIMRC 2008

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

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

17 Conclusions & Perspectives 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)

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: PlanetLab hosts used to emulate VoIP GWs Different payload sizes 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 P Introduction Voice-TFCC: Experimental Results Voice-TFCC PIMRC 2008

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

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

23 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) VoIP variant of TFRC (Floyd et al.)

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

25 Introduction Packet-based vs. Byte-based Environments: Floyd’s Simulation Results 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) MOS (Mean Opinion Score) Delay Impairment Codec and loss Impairments PIMRC 2008

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