Johansson Olle E wrote: > Kevin, > I think you misunderstand. > > We send INVITE with SDP - we should be ready to receive. > They send 180 ringing without SDP - they are not ready to receive > They start sending RTP to us, which is fine and should be ignored.
Actually, I never finished my message :-) I was responding to his point that the phone shouldn't be sending audio because Asterisk wasn't supposed to be receiving it, but in fact we are 'ready to receive' at that point. In fact, the whole requirement that the other endpoint be ready to receive before it can send you audio, as already commented on, is very confusing and easy to misunderstand. I can certainly agree though that if we haven't been told the other end will accept audio, that we should drop any audio we receive from them. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk Experience" (TM) _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
