Hi SIP Gurus, I've tried to find the relevant RFCs, but am struggling. I can find the odd opinion online, but was wondering if anyone could give a definitive answer.
If a SIP call is initiated (INVITE) and receives either a "180 with SDP", or a "183 with SDP", then the remote party will start to send audio for the inband-ringing. Asterisk then passes this audio, and it is correctly heard by the caller. At present, Asterisk will also start to pass back any handset audio in return, in theory allowing a conversation to occur on an unanswered channel if an endpoint were designed to allow this (free phonecalls here we come!). My question: Should: 1) Asterisk block outbound audio between the 183 Progress and the 200 OK packets? 2) Replace any outbound audio with silence? 3) Replace outbound audio with a special NULL RTP of some sort? Does that exist? 4) Allow any audio to be sent regardless? I have implemented 1) at present on our test rig, but the lack of outbound RTP causes issues with firewall state not being set-up to allow the inbound audio. I am not sure how hard/easy it would be to do 2) as you'd need to create silence of the correct duration to replace each audio frame. Thoughts please? Many thanks, Steve -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
