[asterisk-users] Early audio SIP sequence order question

[Kevin P. Fleming]

On 02/10/2011 06:15 AM, Benoit Panizzon wrote:
> Hi Faisal
>
>> http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
>
> Thank you, but I seem to miss the option which tells asterisk to pass audio
> even if no 183 or 200 is received.
>
> No, we don't set the 'r' Flag while dialing out.
>
> So, my question ist sill the same.
>
> Sould asterisk pass audio of it didn't yet receive a 183 or 200 message, or is
> the carrier doing wrong in sending early audio without 183?

This does indeed sound like an Asterisk bug; Asterisk should be ready 
and willing to accept audio from the called SIP endpoint as soon as the 
INVITE is sent out with an SDP offer to receive audio.

Now the real issue here may be the Dial() application not forwarding 
that audio to the caller, rather than Asterisk not 'accepting' the audio 
and turning it into internal media frames. The net result for you is the 
same, but the source of the problem is quite different.

This can of course cause complications if Dial() is used to dial 
multiple endpoints... because then there could be multiple audio streams 
received from them as the call proceeds towards one of them answering.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kfleming at digium.com
Check us out at www.digium.com & www.asterisk.org