Hi Syd, Honestly, got no clue for now. However, I think the symptom should also happen for other endpoint types, not just PSTN. If so, perhaps you can try to make call to another pjsua using L16/PCM codec (and still avoid resampling). If the symptom was reproducible with this scenario, you probably could pin point the problem location using debugger by doing byte-comparison between frame being sent to recorder and 'the same' frame being sent to stream. BR, nanang On Fri, Jan 29, 2010 at 7:27 PM, Syd Brearley <cherkazoo at live.co.uk> wrote: > > > ________________________________ > Date: Fri, 29 Jan 2010 14:12:06 +0700 > From: nanang@xxxxxxxxx > To: pjsip at lists.pjsip.org > Subject: Re: Windows XP service pack 3 and outgoing RTP issues. > > Hi Syd, > > Looking at this simple audio flow diagram: > --- > aud dev 16khz/PCM -> conf bridge 16khz/PCM --> stream 8khz/PCMU (garbled) > ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? +-> recorder 16khz/PCM (ok) > --- > the audio processing differences will be resampling 16->8 (in conf_bridge) > and (PCMU) encoding. > So, could you test again without resampling, i.e: running pjsua with > additional config "--clock-rate 8000" and "--snd-clock-rate 8000", and still > using PCMU. > BR, > nanang > > > > Hi Nanang, > > Thanks for the suggestion. I am sure I have tried it before, but this > morning requested another test using the suggested parameters. The capture > file I have received seems to be worse because even the beginning of > conversation seems to be garbled. If you have any other suggestions I can > get my colleagues to make another test shortly. > > The sip log and rtp capture is uploaded in the same location with the name > test_8000.zip > > Kind Regards > Syd > > > > ________________________________ > Not got a Hotmail account? Sign-up now - Free > _______________________________________________ > 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 > >
