[asterisk-users] SIP Question - Early audio one-way or 2-way?

[Steve Davies]

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