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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 ....

...ope 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

...ner that mentions that Opus is better than Speex in all aspects. The supported bitrate range for Speex seems to be as low as 2kbps though but Opus can only go as low as 6kbps. Is this one aspect where Speex is still preferred? (I understand that it's not a very common scenario though). Thanks, Manpreet. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20141104/2c855dd5/attachment.htm

For anything below 6 kb/s, I would actually recommend using codec2 rather than Speex or Opus. Cheers, Jean-Marc On 04/11/14 05:55 PM, Manpreet Singh wrote: > Hi, > > I noticed that speex.org <http://speex.org> has a banner that mentions > that Opus is better than Speex in all aspects. The supported bitrate > range for Speex seems to be as low as 2kbps though but Opus can only go > as low as 6kbps. Is this one aspe...

...gnificant downside in using a higher frame size for VoIP applications? Secondly, the sweet spot is mentioned as 16-20kbps for wideband speech "for a 20ms frame". Assuming this is just the codec bitrate (without any headers), does the frame size really influence the Opus bitrate? Thanks, Manpreet. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/opus/attachments/20141104/286c0cc7/attachment.htm

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 enc...

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