Hi Mukesh, Thanks for the reply. I was trying to send the message below yesterday but wasn't sure if anyone was getting it or if it was appearing in the list, so just sent a test message to check, attached below is the original message, Am currently using Asterisk 1.4 and Pjsip 1.0.1 , Asterisk sends calls and playback sound files to pjsip clients connected to a sound card on a linux machine. The sound card is configured at 32khz, pjsip can open the devices all okay no issues there. The only problem is that the sound often clicks/breaks from time to time, in some calls its okay on other its clicks/breaks a lot. In the pjsip logs, I can see messages as below: 8:57:31.646 strm0x952b584 PUT prefetch_cnt=4/4 18:57:32.047 strm0x952b584 JB shrinking 1 frame(s), size=5 18:57:32.073 strm0x952b584 Jitter buffer empty (prefetch=4) 18:57:32.087 strm0x952b584 PUT prefetch_cnt=1/4 18:57:32.087 strm0x952b584 PUT prefetch_cnt=2/4 18:57:32.106 strm0x952b584 PUT prefetch_cnt=3/4 18:57:32.106 strm0x952b584 PUT prefetch_cnt=4/4 18:57:32.527 strm0x952b584 JB shrinking 1 frame(s), size=5 18:57:32.553 strm0x952b584 Jitter buffer empty (prefetch=4) 18:57:32.566 strm0x952b584 PUT prefetch_cnt=1/4 18:57:32.566 strm0x952b584 PUT prefetch_cnt=2/4 18:57:32.587 strm0x952b584 PUT prefetch_cnt=3/4 18:57:32.587 strm0x952b584 PUT prefetch_cnt=4/4 18:57:33.007 strm0x952b584 JB shrinking 1 frame(s), size=5 18:57:33.033 strm0x952b584 Jitter buffer empty (prefetch=4) 18:57:33.047 strm0x952b584 PUT prefetch_cnt=1/4 Seems that the JB shrinking coincides with the clicks/breaks in audio output. Not sure whether is it related to the issue, I have tried everything in the pjsip homepage i.e the audio checklist, CPU is fine, memory is all fine, logs were set to 1 , I did the sndtest and all seemed fine there, really not sure what else I can do. I am running the pjsip client mostly with default options, I think its using alaw/ulaw codec, ec-tail disabled, vad disabled, I tried setting playback-lat up to 200 but still having the issue. I even tried setting clock-rate to 32000 and few other things like quality from 5 to 10 but cant seem to get rid of this issue. I can play the sound files/asterisk recordings okay using alsa player, mostly they are 8khz/16bit/mono audio. The soundcard is running at 32khz sample rate so im not sure if the resampling is causing any issues or maybe some buffer issue or so , Another thing is that is there a way to make pjsip use the playback device only, as the system we are using is only for playing out there will be no recording, so any tips/hints here would be great, Hope someone can help, thanks in advance, DS _____ From: Mukesh Srivastav [mailto:muki_champs@xxxxxxxxx] Sent: Wednesday, February 04, 2009 4:31 PM To: pjsip list Cc: dsheth at konverge.com.my Subject: Re: test Hi, What do you by "Test", do you have any compilation problem, or what it is ? Have a look @ my small web-page: http://www.geocities.com/muki_champs Regards, Mukesh Kumar, Sr.Software Engineer, Mobile Application Developer. Hyderabad. India. +91-9397845485 (M) <http://www.geocities.com/muki_champs> _____ From: Dharmen Sheth <dsheth@xxxxxxxxxxxxxxx> To: pjsip at lists.pjsip.org Sent: Wednesday, February 4, 2009 1:54:54 PM Subject: test Hi just wondering if anyone is getting this , not sure if im sending this to the right place Regards DS -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090204/eca00f38/attachment-0001.html>
