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
