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!).
1) Asterisk block outbound audio between the 183 Progress and the 200
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.
Many thanks, Steve