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

[Faisal Hanif]

hi,

you can simply avoid this by using local ring r option in dial command. azterisk pass local ring voice to caller and will not bridge b leg audio until b leg is answered.iin 
Regards,

Faisal Hanif
(sent from phone)

Steve Davies <davies147 at gmail.com> wrote:

>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
>
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