Matteo Fortini
2010-Nov-11 15:35 UTC
[asterisk-users] Asterisk Playback sound dropping on linphone
Hi, I dial on A* from a linphonec to a Playback() extension, then suddenly the sound stops after a while, without any notice. I enabled debug both in linphone and A*, and the RTP packets are sent from A* and received from linphone. It doesn't matter whether I choose alaw, ulaw, gsm as codec (besides changing cpu load of course). How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. I just need a console scriptable softphone, so maybe there's an alternative to linphone (which seemed good enough anyway!)... Thank you, Matteo
Matteo Fortini
2010-Nov-11 17:40 UTC
[asterisk-users] Asterisk Playback sound dropping on linphone
I did some more tests, and it's not really a problem with linphone: the rtp capture shows empty packets sent by Asterisk. Since the channel which is doing Playback() is in a MeetMe conference, I tried also to speak on another phone on the same conference: well the rtp capture shows the stream from A* becoming silent, then the new sound from the phone comes up. Do I have to file a bug? Thank you, Matteo Il 11/11/2010 16:35, Matteo Fortini ha scritto:> Hi, > I dial on A* from a linphonec to a Playback() extension, then suddenly > the sound stops after a while, without any notice. > I enabled debug both in linphone and A*, and the RTP packets are sent > from A* and received from linphone. It doesn't matter whether I choose > alaw, ulaw, gsm as codec (besides changing cpu load of course). > > How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. > > I just need a console scriptable softphone, so maybe there's an > alternative to linphone (which seemed good enough anyway!)... > > Thank you, > Matteo > >
Sebastian
2010-Nov-12 09:23 UTC
[asterisk-users] Asterisk Playback sound dropping on linphone
Hi On 11/11/2010 03:35 PM, Matteo Fortini wrote:> Hi, > I dial on A* from a linphonec to a Playback() extension, then suddenly > the sound stops after a while, without any notice. > I enabled debug both in linphone and A*, and the RTP packets are sent > from A* and received from linphone. It doesn't matter whether I choose > alaw, ulaw, gsm as codec (besides changing cpu load of course). > > How can I debug it? I'm using A* 1.6.2 and both linphone 2.x and 3.x. > > I just need a console scriptable softphone, so maybe there's an > alternative to linphone (which seemed good enough anyway!)...I use linphonec as well - and haven't found another console sip phone either. I'd be interested if there is another one. Sebastian> > Thank you, > Matteo >