Hello again Norman, I ran another test on two different PC's and did not get the disrupted audio. It has to do with my "super fantastic enhanced" sound card I'm sure. I used the same pjsua.exe in all test cases. Anyway thank you for your help Nanang. If i succeed in tracing down the problem on my PC I will let you know. Best Regards, Hubert On Wed, Oct 8, 2008 at 8:52 PM, Nanang Izzuddin <nanang at pjsip.org> wrote: > Hi Hubert, > > That isn't normal actually. I've just tried the scenario using two > PJSUA instances on two separate PCs, unfortunately the distorted audio > problem didn't occur, everything seemed to be normal as expected (e.g: > one way audio after 'cd 0 1'). > > Did you do the test using PJSUA? If it is reproducible on PJSUA, > please share the PJSUA config. Also please send along log files & RTP > packets capture, in case it is not easily reproducible. > > Regards, > nanang > > > On Wed, Oct 8, 2008 at 5:20 AM, Hubert Langevin > <hubertlangevin at gmail.com> wrote: >> Hi Nanang, >> >> Yes you are right it is not a new behavior of the conference bridge >> but i still have a problem even when i'm not talking to Asterisk. I >> setup two PJSIP user agents on two separate PCs and make a call >> between them. Everything works fine but as soon as i disconnect my >> sound card by doing "cd 0 1" the incoming audio becomes hugely >> distorted, on either user agents. Is this normal? Because port 1, the >> call i made, is still transmitting to my sound card. >> >> I'm a bit confused, do i always have to have my sound card >> transmitting to port 1 or should I do what Norman did, that is >> creating a new port that just transmit low-level noise? >> >> Best Regards, >> >> Hubert >> >> >> On Tue, Oct 7, 2008 at 5:56 AM, Nanang Izzuddin <nanang at pjsip.org> wrote: >>> Hi, >>> >>> AFAIK, there is no such 'new' behavior of the conference bridge. And I >>> just recalled I read this somewhere that (by default) Asterisk's clock >>> for sending RTP is triggered by receiving RTP. So if it doesn't >>> receive any RTP packets, it won't send any RTP packets too. This might >>> be the case. >>> >>> Regards, >>> nanang >>> >>> >>> On Mon, Oct 6, 2008 at 5:17 AM, Hubert Langevin >>> <hubertlangevin at gmail.com> wrote: >>>> Hi everyone, >>>> >>>> I just realised a "new" behavior from pjsip that i haven't seen >>>> before. I make a call from my user agent to a soft PBX, asterisk in >>>> this case, in order to get music on hold. When i make the call, the >>>> conference port for the new call gets created and i hear music on my >>>> headset. Eveything looks fine. My sound card sits on port 0 and the >>>> new call i made sits on port 1 and they are both transmitting to each >>>> other. However if i do "cd 0 1", that is stop my sound card to >>>> transmit to port 1, i can't hear the music anymore, even though i >>>> still have Port 1, the new conference port playing music on hold, >>>> transmitting to port 0. >>>> >>>> Any ideas? >>>> >>>> Hubert >>>> >>>> _______________________________________________ >>>> 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 >>>> >>> >>> _______________________________________________ >>> 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 >>> >> >> _______________________________________________ >> 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 >> > > _______________________________________________ > 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 >
