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 _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
