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2014 Dec 16
1
Estimating bitrate during a real-time voip call
...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 flexib...
2014 Dec 16
3
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
2014 Dec 16
0
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...
2014 Nov 04
2
Opus vs Speex NB
Hi, I noticed that 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 aspect where Speex is still preferred? (I understand that it's not a very common scenario though). Thanks, Manpreet. -------------- next part -------------- An HTML
2014 Nov 05
1
Opus frame size
The Opus RFC seems to recommend a frame size of 20ms for most applications. For wideband speech, the sweet spot range is recommended to be 16-20kbps. 20ms frames => 50 frames per second. For a VoIP application, the header overhead per frame (IP+UDP+RTP+SRTP) is 44bytes => 17.6kbps at 50 frames per second. So a 20ms frame size seems to cause a roughly 100% overhead of header data. Therefore,