On Monday, October 13, 2014 11:10:02 PM UTC+1, Nils Ohlmeier wrote: > Hi, > > > > On 10/13/14 11:26 AM, Henrique Rosa wrote: > > > Hello, > > > > > > I've been working on WebRTC support for Mobicents Media Server (MMS). > > > I already succeeded at interop between MMS and Firefox/Chrome. > > > > > > Unfortunately, while testing conference calls I noticed that there is a > > tendency for RTT times and Delay to increase as the call goes on (on both > > browsers). > > > > > > On Chrome, the delay ranges from 3 to 10 seconds for a conference call > > between two participants with duration of 3-4 minutes. > > > On Firefox the results are much better: the delay ranges from 1 to 3 > > seconds in the same scenario. > > > Note that while testing with regular SIP clients (jitsi, linphone, xlite) > > there is no delay whatsoever, so the issue is related with WebRTC calls. > > > > > > My question is: what can possibly cause such delay? > > > > > The one obvious difference between WebRTC and "regular SIP clients" is > > the additional media encryption. But should not take seconds, at least > > if you are using modern desktop PC. > > > > The other question is how much delay FF and Chrome add between capturing > > audio and actually sending them out on the wire. And on the other end > > how delay there is between receiving audio packets and rendering to your > > audio device. > > > > Best regards > > Nils Ohlmeier
Hi Nils, Thank you for taking you time, really appreciate it. The media encryption is not the issue, I debugged that and its fast. I'm sending some logs from the test call I did on Firefox v32. Firefox will make a call to an IVR application that plays announcements for 40 seconds. In the meantime, I had the Media Server tracing the RTCP reports and RTT calculations. I remind you it's a local call, so network congestion shouldn't be a problem either. - Media Server log: https://dl.dropboxusercontent.com/u/47489176/webrtc-mux-ffox32/mms-log.txt - Firefox webrtc log: https://dl.dropboxusercontent.com/u/47489176/webrtc-mux-ffox32/about-webrtc.txt - Wireshark dump: https://dl.dropboxusercontent.com/u/47489176/webrtc-mux-ffox32/mms-ffox32.pcapng Performing same test with SIP clients has no delay and RTT values range from 5 to 25ms approx. Reading the Media Server log, you can see the RTT growth rate is as follows: 16:35:25,169 INFO [RtpMember] rtt=3242011459 - 3241960996 - 47825 = 2638 => 40ms 16:35:29,570 INFO [RtpMember] rtt=3242299883 - 3242208002 - 87674 = 4207 => 64ms (+24ms) 16:35:36,989 INFO [RtpMember] rtt=3242786095 - 3242723311 - 57543 = 5241 => 79ms (+15ms) 16:35:42,508 INFO [RtpMember] rtt=3243147788 - 3242997252 - 144437 = 6099 => 93ms (+14ms) 16:35:46,891 INFO [RtpMember] rtt=3243435032 - 3243274862 - 152405 = 7765 => 118ms (+25ms) 16:35:52,248 INFO [RtpMember] rtt=3243786108 - 3243546968 - 229851 = 9289 => 141ms (+23ms) 16:35:55,109 INFO [RtpMember] rtt=3243973607 - 3243796725 - 166610 = 10272 => 156ms (+15ms) 16:36:01,929 INFO [RtpMember] rtt=3244420562 - 3244165365 - 243715 = 11482 => 175ms (+19ms) If you want to compare results with a similar test I did on Chrome Canary v40, you can refer to the following thread: https://groups.google.com/forum/#!topic/discuss-webrtc/DmZ2eYKGPWA Lately I've been reading about RTP usage on WebRTC and came across RTP/SAVPF profile and RTCP XR report. At the moment we only support RTP/SAVP and regular RTCP reports. Do you think this can have an impact on RTT calculations and delay? Best Regards, - H _______________________________________________ dev-media mailing list [email protected] https://lists.mozilla.org/listinfo/dev-media
