2013/5/15 Emil Ivov <email@example.com>
Can you show us the wireshark dumps?
On 15.05.13, 00:41, Oleg Odinec wrote:
> I use Jitsi as sip phone.
> I receive a Call with first audio codec g711A, and I send response OK
> with first audio codec G711A,
> but when I enable video - it send REINVITE with first codec G722 and I
> get OK from SIP Server with first codec G722, whereupon I can not hear
> received audio.
> Wireshark shows: that after REINVITE rtpstreams are transmitting in
> My interlocutor hears me perfectly , but I do not hear him.
> Follows from the above that Jitsi handles sending RTP in changed codec,
> but it does not work with received audio.
> As well: After change the codec, SIP Server does not change SSRC and
> sends stream in new codec with old SSRC. maybe it is cause of the issue?
> Best regards,