Re: [Freeswitch-users] No audio on caller side when both side support speex/8000 only

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

Could be, anyhow, note that this happens only both side using speex/8000. If one party uses a different codec the problem does not exists. Moreover, these same two clients (with speex/8000) works fine when connected to iptel.org. The most concerning fact is that a=rtpmap:98 SPEEX/8000 sent by FS to the callee even though the caller sent a=rtpmap:102 SPEEX/8000. Does this make any sense?