[asterisk-dev] Asterisk forwards Audio without early session

[Johansson Olle E]

22 jan 2008 kl. 18.50 skrev Kevin P. Fleming:

> Sebastian Damm wrote:
>
>> Asterisk gets a call from PSTN (via Sangoma PRI card). It sends out  
>> an
>> INVITE to the appropriate SIP peer. The SIP peer answers with 180
>> without SDP, but still sends some RTP packets (silent). I don't  
>> know why
>> it does so, but it does. Asterisk now forwards the RTP packets to  
>> PSTN,
>> thus does not generate a Ringing itself. So for the caller the line  
>> is
>> silent until the callee picks up.
>>
>> According to the developers of the SIP PBX, Asterisk should ignore  
>> the
>> RTP packets, because there is no early media session established.  
>> To me,
>> this sounds correct.
>>
>> Is this a bug in Asterisk? Or is the behavior desired as it is? If  
>> not,
>> should I raise a bug in bugtracker?
>
> No, this is incorrect. Asterisk sent an INVITE to the phone that
> included SDP, which means that Asterisk is willing to receive media  
> from
> the phone and the phone is free to send it.
>
> The phone's response without SDP means Asterisk should not send audio
> *to* the phone, but it does not impact Asterisk receiving audio.
Kevin,
I think you misunderstand.

We send INVITE with SDP - we should be ready to receive.
They send 180 ringing without SDP - they are not ready to receive
They start sending RTP to us, which is fine and should be ignored.

The issue here is that Asterisk forwards the media to the PSTN
as early media without getting any indication from the device that
we have a situation with early media. I guess we should ignore incoming
audio until we have 183, 180 or 200 with SDP.

I think I've seen this in SIP2SIP situations too, and then Asterisk
generates an 183 on the caller's side, without an 183 on the callee
call leg.

/O