Re: [Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.
On Tue, Oct 04, 2005 at 09:59:56PM +0200, Olle E. Johansson wrote: > 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. Well, looks like the Asterisk team did not consider it a bug :-) I kinda think they are right and Asterisk is doing the right thing. It's our ISP's gateway that is not performing according to the RFC. Only thing I can think of to try is to shove an SDP payload in the 488 message and see if the other side honors it. Ray ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.
Opened bug #5384. http://bugs.digium.com/view.php?id=5384 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.
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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.
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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.
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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.
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. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Gateway wants T38, Asterisk rejects but media path not established.
Disclaimer: Yes, I know faxing over G711 is unreliable. :-) We're running Asterisk 1.0.9 which talks to a Audiocodes SIP Gateway. We're running Sipura SPA-2002's as ATA's and faxing within our own voice network is working. If we try and fax out to the world however, we're running into a problem. When the call connects and the modem tones begin to negotiate, our SIP/PSTN Gateways's SIP server determines it's a fax call and sends an INVITE to our Asterisk box asking us to negotiate T38. To this, Asterisk replies with a 488: Not acceptable here message. There is no SDP payload in this 488 message. The SIP/PSTN gateway sends an ACK to this. At this point, nothing happens. Eventually Asterisk gets bored and sends a BYE to the SIP/PSTN gateway and the call ends. Our SIP/PSTN gateway provider seems to think that Asterisk should initiate a renegotiation to G711 when it sends the 488 message rejecting T38. Is this how it should work? I've asked them to look into disabling T38 support for us completely, but it seems like Asterisk and the SIP/PSTN gateway should be able to negotiate G711, I just don't know what the proper sequence of events should be. Any ideas? Ray ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
