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

[Ray Van Dolson]

On Tue, Oct 04, 2005 at 09:59:56PM +0200, Olle E. Johansson wrote:
> > 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.

Actually, it appears that between steps 4 & 6 somewhere, the SIP/PSTN gateway
sends a 200 OK with SDP body -- specifying G711u.  This happens above 10.5
seconds after the 183 message was received.

> 
> > 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?

Actually, step 10 was an ACK and contained no 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.

> 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

Will open a bug with the requested info and also a full tcpdump showing the
SIP streams and RTP streams (they go to different servers in this case).

Thanks for the response.

Ray