[Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.

[Olle E. Johansson]

Ray Van Dolson wrote:
> On Thu, Sep 29, 2005 at 08:54:42PM -0500, Kevin P. Fleming wrote:
> 
>>Ray Van Dolson wrote:
>>
>>
>>>Our SIP/PSTN gateway provider seems to think that Asterisk should initiate 
>>>a
>>>renegotiation to G711 when it sends the 488 message rejecting T38.
>>
>>This is not correct. The 488 response 'cancels' the INVITE, so no codec 
>>change was ever actually involved. The gateway should continue sending 
>>G711 since the other device (Asterisk) did not accept the change.
> 
> 
> I agree.
> 
> However, my provider is telling me that they need Asterisk to send them a
> re-INVITE with G711u requested in order to re-establish the RTP stream in both
> directions.
> 
> Their gateway appears to be an Audiocodes-Sip-Gateway-TrunkPack
> 1610/v.4.40.211.387.
> 
> What seems to happen is this:
> 
> 1. Asterisk sends the initial INVITE (requesting G711u)
> 2. SIP/PSTN gateway says it's trying (100) and its media server begins sending
>    G711U RTP traffic.
> 3. SIP/PSTN gateway sends a 183 session progress message with an SDP payload
>    (carrying G711)
> 4. Asterisk begins sending RTP data (G711).  RTP continues in both directions
>    for 10 seconds or so.
> 5. Fax negotiation tone occurs.
> 6. SIP/PSTN gateway stops transmitting RTP
> 7. SIP/PSTN gateway sends an INVITE requesting T38
Before the session is established? Interesting.

> 8. Asterisk replies with a 488 Not acceptable here.
> 9. Asterisk begins transmitting RTP G711U again
> 10. SIP/PSTN gateway response with 200 OK
With what SDP?
> 11. Asterisk continues transmitting RTP for another 30 seconds or so.
> 12. Asterisk sends BYE
> 13. SIP/PSTN gateway response OK and the call is terminated.
> 
> Since their SIP/PSTN gateway doesn't appear to restart G711u transmission at
> step 10, I either need to talk to the manufacturer of this device directly to
> confirm that this is how it is supposed to behave, or look into either getting
> asterisk to send another INVITE or to include a session description for G711
> in its 488 message in step 8 (which appears to be a valid thing to do according 
> to the RFC).
> 
> Anyone out there used an Audiocodes Gateway before?  Our provider tells us
> it's not possible to turn T38 support off on a per-customer basis on this
> gateway.
> 
> Any advice would be appreciated.  I guess I'll try to get an SDP payload into
> the 488 message, but I just feel like the Audiocodes isn't doing the right
> thing here.
> 
I think this is a bug. Please open a report in the bug tracker,
attaching all the requested information. If a re-invite fails, we should
not cancel the call. I am afraid that is exactly what is happening here
and would like to investigate this issue further. It is indeed an
interesting call flow that we have to prepared for as we are
implementing T.38.

Don't forget to add a full SIP debug with verbose 4, debug 4 and sip
debug turned on. Make sure your debug log channel goes to the console
together with verbose and the rest of logging.

I might choose to postpone the actual work with this until after
Astricon, there's quite a lot to work with right now. New registrations
come in every hour and we're going to be more than 300 persons in
California!

Meet you there!
/O