Thanks Benny - J On 4/11/08, Benny Prijono <bennylp at pjsip.org> wrote: > > On Thu, Apr 10, 2008 at 10:41 PM, Jeremy King > <jerry.ipe.thomas at gmail.com> wrote: > > Thanks for the RTP header tip! What was being printed wasn't making any > > sense! > > > > After my temporary timestamp generation patch, I can hear audio > playback, > > which is great, but quality is bad. I tried the same call with X-Lite as > > well as SJPhone and audio was good. Is there something I need to or can > > change/tweak to improve PJ audio quality? > > There is a problem if you use sequence number instead of timestamp to > feed the frame to jitter buffer, namely that you won't be able to > handle multiple frames inside one RTP packet. With G.711 codec, this > will happen all the time as the definition of G.711 frame is 10ms, > while normally an RTP packet contains 20ms worth of frame. So half of > the audio is lost! > > As you said remote is sending RTP with timestamp equal to zero (all > the time). This is just wrong, and you must fix that instead of fixing > PJSIP. Frankly I don't care if any other softphones handle this, I'm > not bothered. :) > > > I have attached the audio capture files for PJ, SJPhone and X-Lite > > annoucement playback for the same announcement. The attachment, > > 20080411014500_TS1.tar.bz2, is ~1.63 MB. I hope it doesn't get blocked. > > > > It does. It's too big for this list. > > Cheers > Benny > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > -- - Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080411/97ddd76d/attachment.html
