Hi All, I should have some time to work on this by the end of the week.
Our asterisk setup makes it hard to reproduce the problem, so we installed a new asterisk server in a testing environment. However, the server is on the local network, we should consider moving it on the wan, Thanks all for the recent useful feedbacks. These are most helpful. I notice on my side that, using g722, the first call fails but the second succeed. This seems to confirm that the problem reside in codec initialization. Regards, Alexandre ----- Original Message ----- From: "Jan Middelkoop" <[email protected]> To: [email protected] Sent: Monday, February 7, 2011 10:19:15 AM Subject: Re: [SFLphone] the other site can't hear me - not NAT/FW problem I think I can confirm this. The problem I e-mailed about this morning (sound stopping when a call is answered) was resolved when I switched asterisk to use ulaw instead of alaw. Kindest regards, Jan Middelkoop Met vriendelijke groeten, Jan Middelkoop ICT beheerder Recreatie& Zorg Groep Website: http://www.recreatie-zorg.nl/ E-mail: [email protected] Telefoon: +31 10 714 2297 Op 3-2-2011 15:30, kapetr schreef: > Hello, > > Besides the persistent problem of often missing output RTP stream, I > have > a new problem since (+- ? - maybe from begin of time) > > snapshot20110119 version. (Now I'm using 30.1.2011 version) > > Even if both (input/output) RTP streams seems to be OK, the other > site can't hear me (in most cases). > > Interesting is, that if I record this call, then in recorded WAV are > both sides to hear. > > After many tests I thing, the problem is in codecs mishmash. > > What ever I set by my account as allowed codecs, SFLPhone uses as > output codec PCMU (= 711u). > > e.g. > 1. I set as only codec GSM > 2. I make call (in INVITE is only GSM codec listed - OK) > 3. The other side accepts this codes (in ACCEPT is also only this > GSM codec - OK) > 4. the other side sends to me RTP stream in GSM codec - OK, but > I send RTP in 711u (PCMU) codec ! - ERR -> The other side do not > hear me. > > Tested by 2 different providers - iptel.org and mikrotech.cz. > > The same problem is if I allow more codecs (inclusive 711u) - if the > 711u is not selected as first, the other side accepts my list and > sends to me in 1. listed codec (e.g. 711a, GSM, ...), but I send > again in 711u. And other side do not hear me. > > Also - SFLPhone MUST by outgoing call send RTP stream in 1.codec > listed in ACCEPT session description (which is in ideal the same as > the 1. in INVITE list). > > --kapetr > > P.S.: is there any chance to repair the "missing output RTP stream > persistent problem" ? > > > > _______________________________________________ > SFLphone mailing list > [email protected] > http://lists.savoirfairelinux.net/mailman/listinfo/sflphone _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone _______________________________________________ SFLphone mailing list [email protected] http://lists.savoirfairelinux.net/mailman/listinfo/sflphone
