Manpreet Singh
2014-Dec-16 06:54 UTC
[opus] Estimating bitrate during a real-time voip call
Hi, Although this maybe considered out of scope here, but I'll ask anyway. Opus has remarkable flexibility for changing encoder bitrate during a call. Are there any suggestions about how bandwidth/capacity between the two endpoints can be measured/estimated during a call so that the outgoing bitrate can be adjusted accordingly? Thanks, Manpreet. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20141215/5ebe425a/attachment.htm
Dragos Oancea
2014-Dec-16 10:09 UTC
[opus] Estimating bitrate during a real-time voip call
Hi You can start a VOIP call with 50 kbps bitrate and reduce the bitrate if there is packet loss. ?You know if there's packet loss if you receive RTCP .?Linphone does this . Regards,Dragos Oancea From: Manpreet Singh <manpreets7 at gmail.com> To: opus at xiph.org Sent: Tuesday, December 16, 2014 7:54 AM Subject: [opus] Estimating bitrate during a real-time voip call Hi, Although this maybe considered out of scope here, but I'll ask anyway. Opus has remarkable flexibility for changing encoder bitrate during a call. Are there any suggestions about how bandwidth/capacity between the two endpoints can be measured/estimated during a call so that the outgoing bitrate can be adjusted accordingly? Thanks,Manpreet. _______________________________________________ opus mailing list opus at xiph.org http://lists.xiph.org/mailman/listinfo/opus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20141216/206e3a43/attachment.htm
Timothy B. Terriberry
2014-Dec-16 12:56 UTC
[opus] Estimating bitrate during a real-time voip call
Manpreet Singh wrote:> Opus has remarkable flexibility for changing encoder bitrate during a > call. Are there any suggestions about how bandwidth/capacity between the > two endpoints can be measured/estimated during a call so that the > outgoing bitrate can be adjusted accordingly?The topic is quite complicated. You can find code implementing the algorithm that Firefox and Chrome use in WebRTC at <http://www.webrtc.org/native-code/development>, though it is deeply integrated with the rest of the media stack there. It is documented at <https://tools.ietf.org/html/draft-alvestrand-rmcat-congestion>. The IETF has chartered a working group to standardize some techniques. You can find the other active proposals here: <https://datatracker.ietf.org/wg/rmcat/documents/>.
Manpreet Singh
2014-Dec-16 19:41 UTC
[opus] Estimating bitrate during a real-time voip call
Hi Dragos, The issue is that not all packet loss maybe congestion related. Often, reducing bitrate seems to have no impact on improving packet loss. Thanks, Manpreet. On Tue, Dec 16, 2014 at 2:09 AM, Dragos Oancea <droancea at yahoo.com> wrote:> > Hi > > You can start a VOIP call with 50 kbps bitrate and reduce the bitrate if > there is packet loss. You know if there's packet loss if you receive RTCP > . > Linphone does this . > > Regards, > Dragos Oancea > > ------------------------------ > *From:* Manpreet Singh <manpreets7 at gmail.com> > *To:* opus at xiph.org > *Sent:* Tuesday, December 16, 2014 7:54 AM > *Subject:* [opus] Estimating bitrate during a real-time voip call > > Hi, > > Although this maybe considered out of scope here, but I'll ask anyway. > > Opus has remarkable flexibility for changing encoder bitrate during a > call. Are there any suggestions about how bandwidth/capacity between the > two endpoints can be measured/estimated during a call so that the outgoing > bitrate can be adjusted accordingly? > > Thanks, > Manpreet. > > > _______________________________________________ > opus mailing list > opus at xiph.org > http://lists.xiph.org/mailman/listinfo/opus > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20141216/dd6da29c/attachment.htm