Heterogeneous Networks VidEv 2012 Varun Singh, J¨ org Ott, Igor Curcio Comnet, Aalto University, {varun,jo}@comnet.tkk.fi Nokia Research Center, [email protected] June 25, 2012 1 / 14
Is widely used for telephony, video conferencing, and telepresence applications Often used over best-effort UDP/IP networks RTP provides playout timing and packet sequencing Reception quality feedback every few seconds (RTCP) RTCP provides higher-level summary feedback instead of per-packet feedback 2 / 14
video streaming, video communication has less opportunity to buffer packets/frames Delay budget is of 200-400 ms. Upper-bound 400ms recommended by 3GPP Video emerging in the Web-browser (WebRTC), FaceTime and Skype for Mobile 4 / 14
Round Trip Time (RTT) Discard Rate Frame Inter- arrival time - Long term cue - Fairness - Late in indicating - Over-use - Early sign of congestion - Undershoot Variation indicates - Under-utilization - Over-use - Tolerant to short variations - Early sign of congestion - Sensitive to competition 7 / 14
Delay and Frame Inter-arrival time can be used for congestion control against competing traffic. Contributions to IETF: RTP Congestion Control: Circuit Breakers for Unicast Sessionsa. RTCP Extension Report for Discard RLE Packetsb. Future Work: More complex scenarios need to be simulated. Deploy in real-world, for example in WebRTC. ahttp://tools.ietf.org/html/draft-perkins-avtcore-rtp-circuit-breakers bhttp://tools.ietf.org/html/draft-ietf-xrblock-rtcp-xr-discard-rle-metrics 13 / 14