This looks and sounds like a case where pjsip isn't listening to our SDP. If we 200 OK with speex on 102 and the far end starts sending it on 98 then I suspect the client is broken if I'm not mistaken. /b On Sep 9, 2009, at 6:19 AM, Tzury Bar Yochay wrote:
Hi,
Owe to the network bandwidth limitations (running on cellular phones ip link) we are using speex/8000 as our voice codec.
However, when both parties are using that codec the sound is not to be heard on the caller side.
looking at the log dumps one can see that
a) at the caller side, it supports speex/8000 in pt=102 and receives from the server speex/8000 in pt=102 b) at the callee side FreeSwitch supports support speex/8000 in pt=98 although it receives from the client speex/8000 in pt=102
When the voice starts caller sends RTP with pt=102 and expect to receive RTP with pt=102, while the callee sends RTP with pt=98 and expect to receive RTP with pt=102.
The RTP packets that received in the caller side are with pt=98 instead of 102 and thusly the client drops them.
Attached are the 2 files recorded from a call between 2 pjsip clients that support only speex/8000 codec.
un_FSCallerSide-speexClient.TXT – is the caller side SIP messages.
un_FSAnswerSide-speexClient.TXT – is the answer side of SIP messages.
Is there anything can be done at the configuration level to avoid this? Thanks in advance for your help
/tzury <un_FSAnswerSide-speexClient.TXT><un_FSCallerSide- speexClient.TXT>_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org