IAB CONCERNS ABOUT CONGESTION CONTROL. Iffat Hasnian 1832659

IAB CONCERNS ABOUT CONGESTION CONTROL Iffat Hasnian 1832659

IAB CONCERNS Outline 1- Introduction 2- Persistent High Drop rate Problem 3- Current Efforts in the IETF 3.1 RTP 3.2 TFRC 3.3 DCCP 3.4 Audio Codecs 4- A Simple Heuristic 5- Constraints on VoIP Systems 6- Conclusions 2

Introduction Voice service over best-effort broadband Internet Connection is an available service now with growing demand. Qos not generally available in current internet Voice traffic occasionally deployed as best effort over some links and we expect this occasional deployment to continue. Concerns Lack of effective end-to-end congestion control for this best effort voice traffic Fairness User Quality Danger of Congestion Collapse (on telephone,video,other real time traffic) (Arise from rapid growth in best effort telephony traffic on best effort networks) 3

Introduction (Cont d) Deployment of Technologies Requiring changes subject to commercial & technical considerations Without Changes high speed of deployment Growth rate of internet telephony is likely to be greatest in developing countries Reason: Economic Factor So congestion control is important topic for developing countries Reason: Congested core links 4

Persistent High Drop Rate Reason Of Occurrence in internet Routing failure Other major disruption Reason of non-occurrence in internet Internet uses TCP and TCP Self corrects If congestion detected two ends of connection reduces the rate of packet Problems: Congestion Collapse User Quality The Amorphous problem of Fairness 5

Persistent High Drop Rate (Cont d) Congestion Collapse Observed in 1980s(in early growth phase of internet) Reason of Occurrence: networks having multiple congested links with high persistent packet drop rate Injected Packets are dropped on downstream congested links After Occurrence: Traffic slows down No acceptable packet delivery No acceptable performance If both ends of VoIP call on congested broadband connection (i.e DSL)-- Congestion collapse is a potential problem in VoIP networks 6

Persistent High Drop Rate (Cont d) User Quality Consider network scenario VoIP flows = N Link =128 Kbps Each flow sending = 64 Kbps Assumptions: This link is only congested link No other traffic except N VoIP flows Ignore Extra bandwidth used for FEC and packet headers Ignore the 2 streams composing a bidirectional VoIP calls 7

Persistent High Drop Rate (Cont d) Arrival rate to link = N * 64 Kbps Traffic forwarded = 2* 64 Kbps Traffic dropped = (N* 64) (2* 64) = (N-2) * 64 Kbps Fraction of dropped arriving traffic =(N-2) / N Each flow receives on average a fraction = 1/ N of link bandwidth Quality of VoIP can be improved if Each VoIP uses end-to-end congestion control has codec which can adapt the bit rate to the bandwidth actually received by that flow Effect of these measures: To reduce the aggregate packet drop rate Packet drop rate > 20 % (Audio quality is degraded) 8

Persistent High Drop Rate The Amorphous problem of Fairness: Adding TCP traffic Non congestion controlled + Congestion controlled TCP Same packet drop rate (for both) = (N-2) / N TCP flow < 64 Kbps VoIP is crowding out the TCP traffic Relative fairness b/w VoIP & TCP traffic Packet sizes Round-trip times (Cont d) Larger packet TCP Flow Shorter RTT Fixed Packet drop rate More BPS 9

Persistent High Drop Rate (Cont d) High packet drop rate Sending rate depends on algorithm for setting RTO TCP connections With fine-grained timestamps high sending rate RTT measured retransmitted packet received & acknowledged Without timestamps Low sending rate RTT measured new packet received & acknowledged 10

Current Efforts in IETF RTP An upgrade of RTP specification TFRC DCCP Work on Audio codecs RFC3551 says: RTP receiver SHOULD monitor packet loss Packet loss is acceptable if Av.Throughput TCP flow > RTP Flow Condition is implemented by Congestion control mechanism to adapt transmission rate By arranging receiver to leave session if loss is unacceptably high 11

Current Efforts in IETF (Cont d) TFRC TCP Friendly Rate Control Equation based congestion control Throughput/time Low variation than TCP Suitable Telephony Streaming media Designed For: Fixed packet size Varying sending rate (packet/sec) in response to congestion 12

Current Efforts in IETF (Cont d) DCCP Datagram congestion control protocol Transparent Protocol For unreliable flows Two congestion control identifiers (CCIDs) CCID2 For TCP like congestion control CCID3 For TFRC congestion control 13

Current Efforts in IETF (Cont d) WORK ON CODECS Selection of appropriate codecs Low sending rate as throughput decreases/packet loss increases Results: improved scaling of no. of VoIP or TCP session capable of sharing congested links acceptable performance to users RFC 3267 describes: Adaptive Multi rate audio codecs(amr) Adaptive Multi rate wideband audio codec(amr-wb) Internet Low bit Rate codec (ILBC) LPC Codec (old multi rate codec) 14

Current Efforts in IETF (Cont d) AMR For 3G cellular system 8 speeches encoding modes Bit rate b/w 4.75 Kbps & 12.2 Kbps Speech encoding performed on 20 ms speech frames Reduced transmission rate in silence periods ILBC Designed for graceful speech quality degradation in case of packet loss Payload bit rate of 13.3 Kbps for 30ms &15.20 Kbps for 20ms frames LPC codec Old multi rate codec Two bit rates i.e 2.4 Kbps &9.6 Kbps Can send additional ``residual`` bits (optional) Reason :enhanced quality at higher bit rate 15

A Simple heuristic Use RTCP loss rate for estimation congestion Example: If RTCP loss rate > 30% OR N back-to-back RTCP reports missing Network congested Results: Terminate/suspend ``sending`` 16

Constraints on VoIP VoIP systems exhibit limited ability to scale their packets no. of packets decreases audio/packet increases high error concealment Error longer than phoneme length( 40 to 100ms) is unrecoverable Voice media streams exhibit greater loss sensitivity at lower data rates. Lower data rate codecs are more sensitive to loss 17

Conclusions Current real time media encoding and transmission practice ignores congestion considerations become a broadly deployed service in the near to intermediate term Problems: Poor user quality Unfairness to other VoIP & TCP users Possibility of sporadic episodes of congestion collapse Can be mitigated: Use Fixed rate codecs by requiring the best effort VoIP application to specify its minimum bit throughput rate. Codecs that are able to vary their bit rate are more effective in providing good quality servics under high load condition 18

References Request For Comments 3714 19

IAB Concerns About Congestion Control THANKS 20