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

Brian, The original "a=rtpmap:98 SPEEX/8000" is generated by FS and NOT by the client.