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

[Steve Davies]

On 24 August 2012 15:34, Faisal Hanif <faisal at vopium.com> wrote:
> 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
>>
> 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)

Nice thought, but what if there is a useful reason for the progress
audio? If it is sent we want to hono[u]r it, and if it is not, we
expect a "180 ringing", and let the SIP device generate the tone,
rather than send an unwanted audio stream to use up bandwidth :)

For example, some UK ISDN services will give a useful call failure
message by sending a progress-frame, followed by some audio. DAHDI and
SIP handle this nicely  with a 183 progress message, and pass on the
message on the un-answered SIP channel.

Regards,
Steve