Hello,
I made lots of tests in order to make Kamailio act as a topology hiding proxy, but i always got problems when using topos to store needed Route and Record-Route informations.
I thought that was because of possible error(s) playing with : - socket binding - destination tweaking that may have lead to transaction breaking, so i decided to use NAT mode (even if that only can be temporary).
My problem is the same. I have not found any way to influence TOPOS on the domain it use on the « Contact » header.
Everything is getting well, iPs on Contact Header are toggled between external and internal sides, until RE-INVITE. From there, Contact domain are no more toggled, and so: - Public IP appear on Private side : i can manage my divide to force it to reply to the sender of the request instead of the contact - Private IP appear on Public side : i can’t do anything to make the external side ignore contact’s domain
I wonder if that is a bug, related to RE-INVITE, or if there is any variable i can change to say topos which domain to be advertised on Contact headers?
PUBLIC NETWORK SIDE RFC6598 NETWORK SIDE ————————— | | PROVIDER Y.Y.Y.Y———————X.X.X.X—| KAMAILIO ]—100.64.10.1———————— 100.64.10.2 ASTERISK (or anything else) | | —————————
20:58:46.215911 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 1004 INVITE sip:+0598@100.64.10.1;user=phone SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport Max-Forwards: 70 From: "FAX" sip:+9997@100.64.10.2;tag=as4cfa9905 To: sip:+0598@100.64.10.1;user=phone Contact: sip:+9997@100.64.10.2:5060 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: sip:+9997@voip-connect.fr;user=phone Content-Type: application/sdp Content-Length: 246
v=0 o=root 79038203 79038203 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=audio 13564 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv
20:58:46.216642 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 375 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK38ea6539;rport=5060;received=100.64.10.2 From: "FAX" sip:+9997@100.64.10.2;tag=as4cfa9905 To: sip:+0598@100.64.10.1;user=phone Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 0
20:58:46.222391 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 1040 INVITE sip:+0598@Y.Y.Y.Y:5060 SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 Max-Forwards: 69 From: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 To: sip:+0598@Y.Y.Y.Y;user=phone Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.13.1 Date: Thu, 29 Dec 2016 19:59:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Access-Network-Info: GSTN;operator-specific-GI=320615200;network-provided P-Asserted-Identity: sip:+9997@voip-connect.fr;user=phone Content-Type: application/sdp Content-Length: 264 Contact: sip:btpsh-58656af1-769f-1@X.X.X.X
v=0 o=root 79038203 79038203 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=audio 57810 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv a=nortpproxy:yes
20:58:46.234592 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 290 SIP/2.0 100 Trying Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 To: sip:+0598@Y.Y.Y.Y;user=phone Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE
20:58:46.548771 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 718 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 To: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp Content-Disposition: session;handling=required Content-Type: application/sdp
v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - -
20:58:46.554903 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 719 SIP/2.0 183 Session Progress From: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 To: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: sip:atpsh-58656af1-769f-2@100.64.10.1
v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes
20:58:47.735919 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 To: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp Content-Disposition: session;handling=required Content-Type: application/sdp
v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - -
20:58:47.741229 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 To: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: sip:atpsh-58656af1-769f-2@100.64.10.1
v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes
20:58:48.050762 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 709 SIP/2.0 180 Ringing Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 To: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Contact: sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp Content-Disposition: session;handling=required Content-Type: application/sdp
v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - -
20:58:48.056175 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 710 SIP/2.0 180 Ringing From: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 To: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: sip:atpsh-58656af1-769f-2@100.64.10.1
v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes
20:58:52.811066 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 774 SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.841e0d4a4e7897fd734b985f81a15580.0 From: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 To: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 216 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp
v=0 o=- 1510281183 168103904 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=audio 17818 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - -
20:58:52.818381 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 775 SIP/2.0 200 OK From: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 To: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 INVITE Content-Length: 232 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Via: SIP/2.0/UDP 100.64.10.2:5060;received=100.64.10.2;branch=z9hG4bK38ea6539;rport=5060 Contact: sip:atpsh-58656af1-769f-2@100.64.10.1
v=0 o=- 1510281183 168103904 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=audio 62130 RTP/AVP 8 101 a=sendrecv a=ptime:20 a=rtpmap:101 telephone-event/8000 a=maxptime:30 a=silenceSupp:off - - - - a=nortpproxy:yes
20:58:52.819315 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 446 ACK sip:atpsh-58656af1-769f-2@100.64.10.1 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.2:5060;branch=z9hG4bK18c31028;rport Max-Forwards: 70 From: "FAX" sip:+9997@100.64.10.2;tag=as4cfa9905 To: sip:+0598@100.64.10.1;user=phone;tag=SDc8s1299-016500890000adc2 Contact: sip:+9997@100.64.10.2:5060 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0
20:58:52.825894 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 491 ACK sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;branch=z9hG4bK0cec.d149220aa711a5230da51bfea457d8bc.0 Max-Forwards: 69 From: "FAX" sip:+9997@100.64.10.2;tag=as4cfa9905 To: sip:+0598@100.64.10.1;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 13.13.1 Content-Length: 0 Contact: sip:btpsh-58656af1-769f-1@X.X.X.X
20:59:00.388481 IP Y.Y.Y.Y.5060 > X.X.X.X.5060: SIP, length: 910 INVITE sip:btpsh-58656af1-769f-1@X.X.X.X sip:btpsh-58656af1-769f-1@X.X.X.X SIP/2.0 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 To: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 From: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 273 Max-Forwards: 69 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Contact: sip:+0598@Y.Y.Y.Y:5060;user=phone;transport=udp Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp
v=0 o=- 1510281183 168103905 IN IP4 Y.Y.Y.Z s=- c=IN IP4 Y.Y.Y.Z t=0 0 m=image 17818 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy
20:59:00.390679 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 420 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP Y.Y.Y.Y:5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1;rport=5060 To: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 From: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 0
20:59:00.395723 IP 100.64.10.1.5060 > 100.64.10.2.5060: SIP, length: 890 INVITE sip:+9997@100.64.10.2:5060 SIP/2.0 Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0 To: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 From: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Content-Length: 289 Max-Forwards: 68 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO Accept-Encoding: identity Content-Disposition: session;handling=required Content-Type: application/sdp Contact: sip:btpsh-58656af1-769f-1@X.X.X.X
v=0 o=- 1510281183 168103905 IN IP4 100.64.10.1 s=- c=IN IP4 100.64.10.1 t=0 0 m=image 62130 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:1800 a=T38FaxMaxDatagram:260 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes
20:59:00.397160 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 580 SIP/2.0 100 Trying Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 To: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: sip:+9997@100.64.10.2:5060 Content-Length: 0
20:59:00.397344 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 To: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: sip:+9997@100.64.10.2:5060 Content-Type: application/sdp Content-Length: 266
v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy
20:59:00.405004 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 SIP/2.0 200 OK From: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 To: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: sip:atpsh-58656af1-769f-2@100.64.10.1
v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes
20:59:00.896864 IP 100.64.10.2.5060 > 100.64.10.1.5060: SIP, length: 875 SIP/2.0 200 OK Via: SIP/2.0/UDP 100.64.10.1;branch=z9hG4bK720b.88d9cf77da5d6f5fa4aac2c5623d8ac4.0;received=100.64.10.1;rport=5060 From: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 To: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: sip:+9997@100.64.10.2:5060 Content-Type: application/sdp Content-Length: 266
v=0 o=root 79038203 79038204 IN IP4 100.64.10.2 s=Asterisk PBX 13.13.1 c=IN IP4 100.64.10.2 t=0 0 m=image 4882 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy
20:59:00.904049 IP X.X.X.X.5060 > Y.Y.Y.Y.5060: SIP, length: 874 TSIP/2.0 200 OK From: sip:+0598@Y.Y.Y.Y;user=phone;tag=SDc8s1299-016500890000adc2 To: "FAX" sip:+9997@X.X.X.X;tag=as4cfa9905 Call-ID: 1c9fb2447f3f72732a83aafa6a3e3017@100.64.10.2:5060 CSeq: 1 INVITE Server: Asterisk PBX 13.13.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 285 Via: SIP/2.0/UDP Y.Y.Y.Y:5060;rport=5060;branch=z9hG4bKm3trae309oacrlkm13h0sb0000g00.1 Contact: sip:atpsh-58656af1-769f-2@100.64.10.1
v=0 o=root 79038203 79038204 IN IP4 X.X.X.X s=Asterisk PBX 13.13.1 c=IN IP4 X.X.X.X t=0 0 m=image 57810 udptl t38 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:1400 a=T38FaxUdpEC:t38UDPRedundancy a=nortpproxy:yes
And so on until timeout and BYE
On Thu, Dec 29, 2016 at 09:45:39PM +0100, Daren FERREIRA wrote:
I wonder if that is a bug, related to RE-INVITE, or if there is any variable i can change to say topos which domain to be advertised on Contact headers?
I encountered the same problem. http://lists.sip-router.org/pipermail/sr-users/2016-September/094403.html
The topos module is a very nice feature of 4.4 (and the only reason I consider updating) but it is to new and buggy for production.
Hello Daniel,
Thanks for your reply, i saw your thread, it seemed to be solved with last updates, so i was confident in keeping trying… :)
So i’m wondering what is the best solution for topology hiding.
It is changing from time to time:
- Kamailio + Freeswitch : but in this case, outgoing requests are done with freeswitch which seems much less performant than Kamailio for gateway dispatching according to their availability, their load and so on - Kamailio + SEMS SBC module : but it seems to be no more maintained (replaced by commercial ABC SBC from TRAFOS)
Kamalio is very good for a lot of things, and would be perfect if it might better manage topology hiding, as TOPOS begin to be able to do.
In my case i suppose that is related to topos not to manage multiple branches, or more precisely, not doing its direction detection, that seems to be the most influencing parameter for him to know which domain to use on forwarder requests. As, if i well understood, re-invite is an new branch, or maybe there is something to do in the routes to help TOPOS knowing what to do…
Maybe Daniel (miconda) has some ideas on that, as we already fixed some TOPOS problems in the past…
Best regards
Date: Fri, 30 Dec 2016 10:23:45 +0100 From: Daniel Tryba <d.tryba@pocos.nl mailto:d.tryba@pocos.nl> To: "Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org> Subject: Re: [SR-Users] TOPOS contact issue on RE-INVITE Message-ID: <20161230092344.GN31818@office.pocos.nl mailto:20161230092344.GN31818@office.pocos.nl> Content-Type: text/plain; charset=us-ascii
On Thu, Dec 29, 2016 at 09:45:39PM +0100, Daren FERREIRA wrote:
I wonder if that is a bug, related to RE-INVITE, or if there is any variable i can change to say topos which domain to be advertised on Contact headers?
I encountered the same problem. http://lists.sip-router.org/pipermail/sr-users/2016-September/094403.html http://lists.sip-router.org/pipermail/sr-users/2016-September/094403.html
The topos module is a very nice feature of 4.4 (and the only reason I consider updating) but it is to new and buggy for production.
On Fri, Dec 30, 2016 at 12:48:16PM +0100, Daren FERREIRA wrote:
Kamalio is very good for a lot of things, and would be perfect if it might better manage topology hiding, as TOPOS begin to be able to do.
Have you tried the older topoh module? It works fine on my standalone kamailio setups, but fails in my kamailio loadbalancer setup (can't remember what exactly went wrong).
Hello,
open an issue on bug tracker from the kamailio github.com project adding all the details of the problem you face. I will try investigate in few days, after the winter holidays.
Cheers, Daniel
On 30/12/2016 12:48, Daren FERREIRA wrote:
Hello Daniel,
Thanks for your reply, i saw your thread, it seemed to be solved with last updates, so i was confident in keeping trying… :)
So i’m wondering what is the best solution for topology hiding.
It is changing from time to time:
- Kamailio + Freeswitch : but in this case, outgoing requests are done
with freeswitch which seems much less performant than Kamailio for gateway dispatching according to their availability, their load and so on
- Kamailio + SEMS SBC module : but it seems to be no more maintained
(replaced by commercial ABC SBC from TRAFOS)
Kamalio is very good for a lot of things, and would be perfect if it might better manage topology hiding, as TOPOS begin to be able to do.
In my case i suppose that is related to topos not to manage multiple branches, or more precisely, not doing its direction detection, that seems to be the most influencing parameter for him to know which domain to use on forwarder requests. As, if i well understood, re-invite is an new branch, or maybe there is something to do in the routes to help TOPOS knowing what to do…
Maybe Daniel (miconda) has some ideas on that, as we already fixed some TOPOS problems in the past…
Best regards
Date: Fri, 30 Dec 2016 10:23:45 +0100 From: Daniel Tryba <d.tryba@pocos.nl mailto:d.tryba@pocos.nl> To: "Kamailio (SER) - Users Mailing List" <sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org> Subject: Re: [SR-Users] TOPOS contact issue on RE-INVITE Message-ID: <20161230092344.GN31818@office.pocos.nl mailto:20161230092344.GN31818@office.pocos.nl> Content-Type: text/plain; charset=us-ascii
On Thu, Dec 29, 2016 at 09:45:39PM +0100, Daren FERREIRA wrote:
I wonder if that is a bug, related to RE-INVITE, or if there is any variable i can change to say topos which domain to be advertised on Contact headers?
I encountered the same problem. http://lists.sip-router.org/pipermail/sr-users/2016-September/094403.html
The topos module is a very nice feature of 4.4 (and the only reason I consider updating) but it is to new and buggy for production.
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
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Daniel Tryba -
Daniel-Constantin Mierla -
Daren FERREIRA