Regovich, Timothy
2004-Feb-10 12:06 UTC
[Fwd: [Asterisk-Users] Having problems with RTP packets and H old]
Why does the FXO gateway treat a lack of RTP packets as a dropped call (and what heuristic does it use to determine?) Until the SIP UA sends an actual BYE message, the Dialog should still be considered active, regardless of the RTP that may or may not be happening. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Clif Jones Sent: Tuesday, February 10, 2004 1:33 PM To: asterisk-dev@lists.digium.com; asterisk users Subject: [Fwd: [Asterisk-Users] Having problems with RTP packets and Hold] If anyone is familiar with the SIP SDP handling routines I would appreciate some insight. The following problem that I found using Asterisk appears to be improper handling of a call put on hold when there is no music on hold: [FXO gateway] [Asterisk] [IP phone] |-------[INVITE s/SDP]---------------->|-------[INVITE s/SDP]---------------->| | | | |<--------[180 Ringing]----------------|<--------[180 Ringing]----------------| | | | |<----[183 Session Progress]-----------|<-----------[200 OK/SDP]--------------| | | | |<--------[200 OK/SDP]-----------------|------------[ACK]-------------------->| | |<=========== RTP ====================>| |------------[ACK]-------------------->| | |<=========== RTP ====================>| | {IP phone puts caller on hold} | |<-----[INVITE/held SDP]---------------| | | | | |-----------[200 OK/SDP]-------------->| | | | | |<------------[ACK]--------------------| |============ RTP (one-way)===========>| | | | | |----------[BYE]---------------------->| | | | | |<------------[200 OK]-----------------| | When the IP phone puts the gateway on hold, Asterisk gets the re-INVITE with held media but Asterisk doesn't re-INVITE the gateway. The RTP traffic to the gateway stops so the gateway handles the condition as a lost connection. Shouldn't asterisk be forwarding the re-INVITE to the gateway unless MOH is enabled? ------------------------------------------------------------------------------ Notice: This e-mail message, together with any attachments, contains information of Merck & Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. ------------------------------------------------------------------------------
Clif Jones
2004-Feb-10 12:15 UTC
[Fwd: [Asterisk-Users] Having problems with RTP packets and H old]
Good question. If you look at my original post, you will see that this problem was discovered after this "feature" was evidently added to our AudioCodes gateway GA firmware. The beta code didn't do this. They are probably trying to solve the problem of detecting dropped calls from the IP side but if this "feature" is not selectable you run into problems like this. I'm actually beating them up over this but I have not been impressed with their support as a company. I have still failed to get DTMF bridging via RFC2833 working 100%. If anyone has had success with Audiocodes FXO SIP gateways and Asterisk, I would like to know the magic formula that makes all this work. :) Regovich, Timothy wrote:>Why does the FXO gateway treat a lack of RTP packets as a dropped call (and >what heuristic does it use to determine?) >Until the SIP UA sends an actual BYE message, the Dialog should still be >considered active, regardless of the RTP that may or may not be happening. > > > >-----Original Message----- >From: asterisk-users-admin@lists.digium.com >[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Clif Jones >Sent: Tuesday, February 10, 2004 1:33 PM >To: asterisk-dev@lists.digium.com; asterisk users >Subject: [Fwd: [Asterisk-Users] Having problems with RTP packets and Hold] > > >If anyone is familiar with the SIP SDP handling routines I would appreciate >some > >insight. The following problem that I found using Asterisk appears to be >improper > >handling of a call put on hold when there is no music on hold: > >[FXO gateway] [Asterisk] >[IP phone] > > |-------[INVITE s/SDP]---------------->|-------[INVITE >s/SDP]---------------->| > | | >| > |<--------[180 Ringing]----------------|<--------[180 >Ringing]----------------| > | | >| > |<----[183 Session Progress]-----------|<-----------[200 >OK/SDP]--------------| > | | >| > |<--------[200 >OK/SDP]-----------------|------------[ACK]-------------------->| > | |<=========== RTP >====================>| > |------------[ACK]-------------------->| >| > |<=========== RTP ====================>| >| > > {IP phone puts caller on >hold} > > | |<-----[INVITE/held >SDP]---------------| > | | >| > | |-----------[200 >OK/SDP]-------------->| > | | >| > | >|<------------[ACK]--------------------| > |============ RTP (one-way)===========>| >| > | | >| > |----------[BYE]---------------------->| >| > | | >| > |<------------[200 OK]-----------------| >| > >When the IP phone puts the gateway on hold, Asterisk gets the re-INVITE with >held >media but Asterisk doesn't re-INVITE the gateway. The RTP traffic to the >gateway >stops so the gateway handles the condition as a lost connection. Shouldn't >asterisk >be forwarding the re-INVITE to the gateway unless MOH is enabled? > > > > >------------------------------------------------------------------------------ >Notice: This e-mail message, together with any attachments, contains >information of Merck & Co., Inc. (One Merck Drive, Whitehouse Station, New >Jersey, USA 08889), and/or its affiliates (which may be known outside the >United States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan, as >Banyu) that may be confidential, proprietary copyrighted and/or legally >privileged. It is intended solely for the use of the individual or entity >named on this message. If you are not the intended recipient, and have >received this message in error, please notify us immediately by reply e-mail >and then delete it from your system. >------------------------------------------------------------------------------ >_______________________________________________ >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 > > >
Regovich, Timothy
2004-Feb-10 12:58 UTC
[Fwd: [Asterisk-Users] Having problems with RTP packets and H old]
Can you send the sip debug messages along? That would help. I am interested in what the original invites looked like dialog information) and what the subsequent invite looks like. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Clif Jones Sent: Tuesday, February 10, 2004 2:16 PM To: asterisk-users@lists.digium.com Cc: asterisk-dev@lists.digium.com Subject: Re: [Fwd: [Asterisk-Users] Having problems with RTP packets and H old] Good question. If you look at my original post, you will see that this problem was discovered after this "feature" was evidently added to our AudioCodes gateway GA firmware. The beta code didn't do this. They are probably trying to solve the problem of detecting dropped calls from the IP side but if this "feature" is not selectable you run into problems like this. I'm actually beating them up over this but I have not been impressed with their support as a company. I have still failed to get DTMF bridging via RFC2833 working 100%. If anyone has had success with Audiocodes FXO SIP gateways and Asterisk, I would like to know the magic formula that makes all this work. :) Regovich, Timothy wrote:>Why does the FXO gateway treat a lack of RTP packets as a dropped call (and >what heuristic does it use to determine?) >Until the SIP UA sends an actual BYE message, the Dialog should still be >considered active, regardless of the RTP that may or may not be happening. > > > >-----Original Message----- >From: asterisk-users-admin@lists.digium.com >[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Clif Jones >Sent: Tuesday, February 10, 2004 1:33 PM >To: asterisk-dev@lists.digium.com; asterisk users >Subject: [Fwd: [Asterisk-Users] Having problems with RTP packets and Hold] > > >If anyone is familiar with the SIP SDP handling routines I would appreciate >some > >insight. The following problem that I found using Asterisk appears to be >improper > >handling of a call put on hold when there is no music on hold: > >[FXO gateway] [Asterisk] >[IP phone] > > |-------[INVITE s/SDP]---------------->|-------[INVITE >s/SDP]---------------->| > | | >| > |<--------[180 Ringing]----------------|<--------[180 >Ringing]----------------| > | | >| > |<----[183 Session Progress]-----------|<-----------[200 >OK/SDP]--------------| > | | >| > |<--------[200 >OK/SDP]-----------------|------------[ACK]-------------------->| > | |<=========== RTP >====================>| > |------------[ACK]-------------------->| >| > |<=========== RTP ====================>| >| > > {IP phone puts caller on >hold} > > | |<-----[INVITE/held >SDP]---------------| > | | >| > | |-----------[200 >OK/SDP]-------------->| > | | >| > | >|<------------[ACK]--------------------| > |============ RTP (one-way)===========>| >| > | | >| > |----------[BYE]---------------------->| >| > | | >| > |<------------[200 OK]-----------------| >| > >When the IP phone puts the gateway on hold, Asterisk gets the re-INVITEwith>held >media but Asterisk doesn't re-INVITE the gateway. The RTP traffic to the >gateway >stops so the gateway handles the condition as a lost connection. Shouldn't >asterisk >be forwarding the re-INVITE to the gateway unless MOH is enabled? > > > > >------------------------------------------------------------------------------>Notice: This e-mail message, together with any attachments, contains >information of Merck & Co., Inc. (One Merck Drive, Whitehouse Station, New >Jersey, USA 08889), and/or its affiliates (which may be known outside the >United States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan, as >Banyu) that may be confidential, proprietary copyrighted and/or legally >privileged. It is intended solely for the use of the individual or entity >named on this message. If you are not the intended recipient, and have >received this message in error, please notify us immediately by replye-mail>and then delete it from your system. >------------------------------------------------------------------------------>_______________________________________________ >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 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 ------------------------------------------------------------------------------ Notice: This e-mail message, together with any attachments, contains information of Merck & Co., Inc. (One Merck Drive, Whitehouse Station, New Jersey, USA 08889), and/or its affiliates (which may be known outside the United States as Merck Frosst, Merck Sharp & Dohme or MSD and in Japan, as Banyu) that may be confidential, proprietary copyrighted and/or legally privileged. It is intended solely for the use of the individual or entity named on this message. If you are not the intended recipient, and have received this message in error, please notify us immediately by reply e-mail and then delete it from your system. ------------------------------------------------------------------------------