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 8. Asterisk replies with a 488 Not acceptable here. 9. Asterisk begins transmitting RTP G711U again 10. SIP/PSTN gateway response with 200 OK 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. 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
