Hello everyone!
I have my system based on ser 0.9.6. Internal calls work just fine. I am now trying to interop with another system. It has a sort of asterisk functionality, but it is not asterisk, it is a private software company product.
my.phone ---------> my.ser.proxy -------------> external.proxy -------------> external phone 10.111.0.119 10.111.0.50 10.111.0.20 10.111.0.144
There is no firewall or nat in the way.
The problem, i think, is that the external.proxy is buggy. I told that to the company, but who knows when they will fix this.
* Look at the OK (message #9 and #10). My.ser.proxy record-routes all invites. The Record-route headers reach the external.phone, which copies them in the OK message, sends the OK to the external.proxy ... and when this forwards it to my.ser.proxy, they are gone! The OK reaches my.phone, but then it generates my problem: the ACK. It contains NO ROUTE headers and the r-uri is also simply pointing to the external.proxy. If the ACK had the ROUTE headers, my.ser.proxy would loose_route the message and voila! But as loose_route() returns false, my (maybe bad) config file gets confused and treats it like a "new" call ... so it lookup("location") of the ACK r-ruri fails, and the ACK is dropped. Should i modify the config file so that ACK, if not loose_route'd, are simply t_relay'd?
* They also modify the contact field. The reason is because behind the external.proxy could be H323 or SIP phones, so they sort of want to be a termination as far as signalling is concerned.
I attach the message flow, hope it helps ... Thanks!
Cesc
Short answer: yes. Slightly longer: I have seen the same behavior. Without lr|lr=on the UA will (correctly) go over to strict routing and use contact in r-uri for the ACK. As long as only your record-route disappeared (i.e. you are the only hop in-between the UAS and UAC) this works. As your external proxy is B2BUA (at least for signalling), your fine if you relay the ACK. I'm not sure exactly when this happens and why, I seem to remember I posted something on this behavior for sjphone a while back, but I'm not capable of finding it. g-)
Cesc wrote:
Hello everyone!
I have my system based on ser 0.9.6. Internal calls work just fine. I am now trying to interop with another system. It has a sort of asterisk functionality, but it is not asterisk, it is a private software company product.
my.phone ---------> my.ser.proxy -------------> external.proxy -------------> external phone 10.111.0.119 10.111.0.50 10.111.0.20 10.111.0.144
There is no firewall or nat in the way.
The problem, i think, is that the external.proxy is buggy. I told that to the company, but who knows when they will fix this.
- Look at the OK (message #9 and #10). My.ser.proxy record-routes all
invites. The Record-route headers reach the external.phone, which copies them in the OK message, sends the OK to the external.proxy ... and when this forwards it to my.ser.proxy, they are gone! The OK reaches my.phone, but then it generates my problem: the ACK. It contains NO ROUTE headers and the r-uri is also simply pointing to the external.proxy. If the ACK had the ROUTE headers, my.ser.proxy would loose_route the message and voila! But as loose_route() returns false, my (maybe bad) config file gets confused and treats it like a "new" call ... so it lookup("location") of the ACK r-ruri fails, and the ACK is dropped. Should i modify the config file so that ACK, if not loose_route'd, are simply t_relay'd?
- They also modify the contact field. The reason is because behind the
external.proxy could be H323 or SIP phones, so they sort of want to be a termination as far as signalling is concerned.
I attach the message flow, hope it helps ... Thanks!
Cesc
No. Time Source Destination Protocol Info 1 0.000000000 10.111.0.119 10.111.0.50 SIP/SDP Request: INVITE sip:6007@10.111.0.20, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Content-Length: 337 Contact: sip:7005@10.111.0.119:5060 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 70 To: sip:6007@10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 3 0.001209000 10.111.0.50 10.111.0.20 SIP/SDP Request: INVITE sip:6007@10.111.0.20:5060, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20:5060 SIP/2.0 Message Header Record-Route: sip:10.111.0.50;ftag=12083790462495;lr=on Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Content-Length: 337 Contact: sip:7005@10.111.0.119:5060 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 16 To: sip:6007@10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 5 0.040629000 10.111.0.20 10.111.0.144 SIP/SDP Request: INVITE sip:6007@10.111.0.144:5060;transport=UDP, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.144:5060;transport=UDP SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e RECORD-ROUTE: sip:10.111.0.50;ftag=12083790462495;lr=on From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Max-Forwards: 16 Contact: sip:10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Content-Type: application/sdp Content-Length: 337 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 gsm/8000 Media Attribute (a): rtpmap:97 ilbc/8000 Media Attribute (a): rtpmap:98 ilbc/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 pcma/8000 Media Attribute (a): rtpmap:0 pcmu/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 9 3.736665000 10.111.0.144 10.111.0.20 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Record-Route: sip:10.111.0.50;ftag=12083790462495;lr=on Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Contact: sip:6007@10.111.0.144:5060 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE Supported: 100rel, replaces Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 PCMA/8000
No. Time Source Destination Protocol Info 10 3.751306000 10.111.0.20 10.111.0.50 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: sip:10.111.0.20:5060 Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info 11 3.751589000 10.111.0.50 10.111.0.119 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: sip:10.111.0.20:5060 Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info 12 3.756127000 10.111.0.119 10.111.0.50 SIP Request: ACK sip:10.111.0.20:5060 Session Initiation Protocol Request-Line: ACK sip:10.111.0.20:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb81d00000c5f00001922 Content-Length: 0 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 ACK From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 70 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 User-Agent: SJphone/1.60.289a (SJ Labs)
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Tks ... i guess i will try to modify the config file so this ack problem is taken care of in the most generic way i can think of ...
What i tried and it seems it didn't work is to enter the "sip:10.111.0.20:5060" as a permanent entry in the location table (using serctl). This way my config, which does a lookup(location) and fails and drops the ACK, it would then return true and simply t_relay it ...
as for the sjphone, i don't blame the phone. I think it does reasonably good, given how bad the OK from the external.proxy gets to him ...
Cesc
On 10/11/06, Greger V. Teigre greger@teigre.com wrote:
Short answer: yes. Slightly longer: I have seen the same behavior. Without lr|lr=on the UA will (correctly) go over to strict routing and use contact in r-uri for the ACK. As long as only your record-route disappeared (i.e. you are the only hop in-between the UAS and UAC) this works. As your external proxy is B2BUA (at least for signalling), your fine if you relay the ACK. I'm not sure exactly when this happens and why, I seem to remember I posted something on this behavior for sjphone a while back, but I'm not capable of finding it. g-)
Cesc wrote: Hello everyone!
I have my system based on ser 0.9.6. Internal calls work just fine. I am now trying to interop with another system. It has a sort of asterisk functionality, but it is not asterisk, it is a private software company product.
my.phone ---------> my.ser.proxy -------------> external.proxy -------------> external phone 10.111.0.119 10.111.0.50 10.111.0.20 10.111.0.144
There is no firewall or nat in the way.
The problem, i think, is that the external.proxy is buggy. I told that to the company, but who knows when they will fix this.
- Look at the OK (message #9 and #10). My.ser.proxy record-routes all
invites. The Record-route headers reach the external.phone, which copies them in the OK message, sends the OK to the external.proxy ... and when this forwards it to my.ser.proxy, they are gone! The OK reaches my.phone, but then it generates my problem: the ACK. It contains NO ROUTE headers and the r-uri is also simply pointing to the external.proxy. If the ACK had the ROUTE headers, my.ser.proxy would loose_route the message and voila! But as loose_route() returns false, my (maybe bad) config file gets confused and treats it like a "new" call ... so it lookup("location") of the ACK r-ruri fails, and the ACK is dropped. Should i modify the config file so that ACK, if not loose_route'd, are simply t_relay'd?
- They also modify the contact field. The reason is because behind the
external.proxy could be H323 or SIP phones, so they sort of want to be a termination as far as signalling is concerned.
I attach the message flow, hope it helps ... Thanks!
Cesc ________________________________
No. Time Source Destination Protocol Info 1 0.000000000 10.111.0.119 10.111.0.50 SIP/SDP Request: INVITE sip:6007@10.111.0.20, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Content-Length: 337 Contact: sip:7005@10.111.0.119:5060 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 70 To: sip:6007@10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 3 0.001209000 10.111.0.50 10.111.0.20 SIP/SDP Request: INVITE sip:6007@10.111.0.20:5060, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20:5060 SIP/2.0 Message Header Record-Route: sip:10.111.0.50;ftag=12083790462495;lr=on Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Content-Length: 337 Contact: sip:7005@10.111.0.119:5060 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 16 To: sip:6007@10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 5 0.040629000 10.111.0.20 10.111.0.144 SIP/SDP Request: INVITE sip:6007@10.111.0.144:5060;transport=UDP, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.144:5060;transport=UDP SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e RECORD-ROUTE: sip:10.111.0.50;ftag=12083790462495;lr=on From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Max-Forwards: 16 Contact: sip:10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Content-Type: application/sdp Content-Length: 337 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 gsm/8000 Media Attribute (a): rtpmap:97 ilbc/8000 Media Attribute (a): rtpmap:98 ilbc/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 pcma/8000 Media Attribute (a): rtpmap:0 pcmu/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 9 3.736665000 10.111.0.144 10.111.0.20 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Record-Route: sip:10.111.0.50;ftag=12083790462495;lr=on Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Contact: sip:6007@10.111.0.144:5060 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE Supported: 100rel, replaces Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 PCMA/8000
No. Time Source Destination Protocol Info 10 3.751306000 10.111.0.20 10.111.0.50 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: sip:10.111.0.20:5060 Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info 11 3.751589000 10.111.0.50 10.111.0.119 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: sip:10.111.0.20:5060 Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info 12 3.756127000 10.111.0.119 10.111.0.50 SIP Request: ACK sip:10.111.0.20:5060 Session Initiation Protocol Request-Line: ACK sip:10.111.0.20:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb81d00000c5f00001922 Content-Length: 0 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 ACK From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 70 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 User-Agent: SJphone/1.60.289a (SJ Labs)
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
Yes, when I come to think of it, I think the sjphone thing was something different.. g-)
Cesc wrote:
Tks ... i guess i will try to modify the config file so this ack problem is taken care of in the most generic way i can think of ...
What i tried and it seems it didn't work is to enter the "sip:10.111.0.20:5060" as a permanent entry in the location table (using serctl). This way my config, which does a lookup(location) and fails and drops the ACK, it would then return true and simply t_relay it ...
as for the sjphone, i don't blame the phone. I think it does reasonably good, given how bad the OK from the external.proxy gets to him ...
Cesc
On 10/11/06, Greger V. Teigre greger@teigre.com wrote:
Short answer: yes. Slightly longer: I have seen the same behavior. Without lr|lr=on the UA will (correctly) go over to strict routing and use contact in r-uri for the ACK. As long as only your record-route disappeared (i.e. you are the only hop in-between the UAS and UAC) this works. As your external proxy is B2BUA (at least for signalling), your fine if you relay the ACK. I'm not sure exactly when this happens and why, I seem to remember I posted something on this behavior for sjphone a while back, but I'm not capable of finding it. g-)
Cesc wrote: Hello everyone!
I have my system based on ser 0.9.6. Internal calls work just fine. I am now trying to interop with another system. It has a sort of asterisk functionality, but it is not asterisk, it is a private software company product.
my.phone ---------> my.ser.proxy -------------> external.proxy -------------> external phone 10.111.0.119 10.111.0.50 10.111.0.20 10.111.0.144
There is no firewall or nat in the way.
The problem, i think, is that the external.proxy is buggy. I told that to the company, but who knows when they will fix this.
- Look at the OK (message #9 and #10). My.ser.proxy record-routes all
invites. The Record-route headers reach the external.phone, which copies them in the OK message, sends the OK to the external.proxy ... and when this forwards it to my.ser.proxy, they are gone! The OK reaches my.phone, but then it generates my problem: the ACK. It contains NO ROUTE headers and the r-uri is also simply pointing to the external.proxy. If the ACK had the ROUTE headers, my.ser.proxy would loose_route the message and voila! But as loose_route() returns false, my (maybe bad) config file gets confused and treats it like a "new" call ... so it lookup("location") of the ACK r-ruri fails, and the ACK is dropped. Should i modify the config file so that ACK, if not loose_route'd, are simply t_relay'd?
- They also modify the contact field. The reason is because behind the
external.proxy could be H323 or SIP phones, so they sort of want to be a termination as far as signalling is concerned.
I attach the message flow, hope it helps ... Thanks!
Cesc ________________________________
No. Time Source Destination Protocol Info 1 0.000000000 10.111.0.119 10.111.0.50 SIP/SDP Request: INVITE sip:6007@10.111.0.20, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
Content-Length: 337 Contact: sip:7005@10.111.0.119:5060 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 70 To: sip:6007@10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 3 0.001209000 10.111.0.50 10.111.0.20 SIP/SDP Request: INVITE sip:6007@10.111.0.20:5060, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20:5060 SIP/2.0 Message Header Record-Route: sip:10.111.0.50;ftag=12083790462495;lr=on Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
Content-Length: 337 Contact: sip:7005@10.111.0.119:5060 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 16 To: sip:6007@10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 5 0.040629000 10.111.0.20 10.111.0.144 SIP/SDP Request: INVITE sip:6007@10.111.0.144:5060;transport=UDP, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.144:5060;transport=UDP SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
RECORD-ROUTE: sip:10.111.0.50;ftag=12083790462495;lr=on From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Max-Forwards: 16 Contact: sip:10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Content-Type: application/sdp Content-Length: 337 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 gsm/8000 Media Attribute (a): rtpmap:97 ilbc/8000 Media Attribute (a): rtpmap:98 ilbc/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 pcma/8000 Media Attribute (a): rtpmap:0 pcmu/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 9 3.736665000 10.111.0.144 10.111.0.20 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
Record-Route: sip:10.111.0.50;ftag=12083790462495;lr=on Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Contact: sip:6007@10.111.0.144:5060 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE Supported: 100rel, replaces Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 PCMA/8000
No. Time Source Destination Protocol Info 10 3.751306000 10.111.0.20 10.111.0.50 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: sip:10.111.0.20:5060 Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info 11 3.751589000 10.111.0.50 10.111.0.119 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: sip:10.111.0.20:5060 Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info 12 3.756127000 10.111.0.119 10.111.0.50 SIP Request: ACK sip:10.111.0.20:5060 Session Initiation Protocol Request-Line: ACK sip:10.111.0.20:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb81d00000c5f00001922
Content-Length: 0 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 ACK From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 70 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 User-Agent: SJphone/1.60.289a (SJ Labs)
Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
If you take a look at 200 OK you will notice that the value of Contact header is changed. The first 200 OK contains
sip:6007@10.111.0.144:5060
which I assume is correct. The second one contains:
sip:10.111.0.20:5060
But that (according to the dumps below) points to SER -- and that's the problem. Do you rewrite Contact header in SER ? If so then it should not point back to the SER instance.
SER treats the ACK correctly -- it has no Route headers and the Request-URI points to SER -- hence it will process it as local message.
What you need to have fixed is the IP and port in rewritten contact.
Jan.
Cesc wrote:
Tks ... i guess i will try to modify the config file so this ack problem is taken care of in the most generic way i can think of ...
What i tried and it seems it didn't work is to enter the "sip:10.111.0.20:5060" as a permanent entry in the location table (using serctl). This way my config, which does a lookup(location) and fails and drops the ACK, it would then return true and simply t_relay it ...
as for the sjphone, i don't blame the phone. I think it does reasonably good, given how bad the OK from the external.proxy gets to him ...
Cesc
On 10/11/06, Greger V. Teigre greger@teigre.com wrote:
Short answer: yes. Slightly longer: I have seen the same behavior. Without lr|lr=on the UA will (correctly) go over to strict routing and use contact in r-uri for the ACK. As long as only your record-route disappeared (i.e. you are the only hop in-between the UAS and UAC) this works. As your external proxy is B2BUA (at least for signalling), your fine if you relay the ACK. I'm not sure exactly when this happens and why, I seem to remember I posted something on this behavior for sjphone a while back, but I'm not capable of finding it. g-)
Cesc wrote: Hello everyone!
I have my system based on ser 0.9.6. Internal calls work just fine. I am now trying to interop with another system. It has a sort of asterisk functionality, but it is not asterisk, it is a private software company product.
my.phone ---------> my.ser.proxy -------------> external.proxy -------------> external phone 10.111.0.119 10.111.0.50 10.111.0.20 10.111.0.144
There is no firewall or nat in the way.
The problem, i think, is that the external.proxy is buggy. I told that to the company, but who knows when they will fix this.
- Look at the OK (message #9 and #10). My.ser.proxy record-routes all
invites. The Record-route headers reach the external.phone, which copies them in the OK message, sends the OK to the external.proxy ... and when this forwards it to my.ser.proxy, they are gone! The OK reaches my.phone, but then it generates my problem: the ACK. It contains NO ROUTE headers and the r-uri is also simply pointing to the external.proxy. If the ACK had the ROUTE headers, my.ser.proxy would loose_route the message and voila! But as loose_route() returns false, my (maybe bad) config file gets confused and treats it like a "new" call ... so it lookup("location") of the ACK r-ruri fails, and the ACK is dropped. Should i modify the config file so that ACK, if not loose_route'd, are simply t_relay'd?
- They also modify the contact field. The reason is because behind the
external.proxy could be H323 or SIP phones, so they sort of want to be a termination as far as signalling is concerned.
I attach the message flow, hope it helps ... Thanks!
Cesc ________________________________
No. Time Source Destination Protocol Info 1 0.000000000 10.111.0.119 10.111.0.50 SIP/SDP Request: INVITE sip:6007@10.111.0.20, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Content-Length: 337 Contact: sip:7005@10.111.0.119:5060 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 70 To: sip:6007@10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 3 0.001209000 10.111.0.50 10.111.0.20 SIP/SDP Request: INVITE sip:6007@10.111.0.20:5060, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20:5060 SIP/2.0 Message Header Record-Route: sip:10.111.0.50;ftag=12083790462495;lr=on Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
Content-Length: 337 Contact: sip:7005@10.111.0.119:5060 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 16 To: sip:6007@10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 5 0.040629000 10.111.0.20 10.111.0.144 SIP/SDP Request: INVITE sip:6007@10.111.0.144:5060;transport=UDP, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.144:5060;transport=UDP SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
RECORD-ROUTE: sip:10.111.0.50;ftag=12083790462495;lr=on From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Max-Forwards: 16 Contact: sip:10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Content-Type: application/sdp Content-Length: 337 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 gsm/8000 Media Attribute (a): rtpmap:97 ilbc/8000 Media Attribute (a): rtpmap:98 ilbc/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 pcma/8000 Media Attribute (a): rtpmap:0 pcmu/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 9 3.736665000 10.111.0.144 10.111.0.20 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
Record-Route: sip:10.111.0.50;ftag=12083790462495;lr=on Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Contact: sip:6007@10.111.0.144:5060 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE Supported: 100rel, replaces Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 PCMA/8000
No. Time Source Destination Protocol Info 10 3.751306000 10.111.0.20 10.111.0.50 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: sip:10.111.0.20:5060 Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info 11 3.751589000 10.111.0.50 10.111.0.119 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: sip:10.111.0.20:5060 Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info 12 3.756127000 10.111.0.119 10.111.0.50 SIP Request: ACK sip:10.111.0.20:5060 Session Initiation Protocol Request-Line: ACK sip:10.111.0.20:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb81d00000c5f00001922 Content-Length: 0 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 ACK From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 70 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 User-Agent: SJphone/1.60.289a (SJ Labs)
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Hi,
The external.proxy (not ser) rewrites the contact, as a way to be a b2bua in terms of signalling: 10.111.0.20 is the external.proxy ... 10.111.0.50 is my.ser.proxy. In terms of ACK processing, no doubt ser does its best ... it is just the external.proxy not processing correctly the record-route, but this shall be fixed soon ... Meanwhile, i modify my config so i detect this kind of ACK and simply t_relay them, nothing else.
Thanks everyone!
Cesc
On 10/12/06, Jan Janak jan@iptel.org wrote:
If you take a look at 200 OK you will notice that the value of Contact header is changed. The first 200 OK contains
sip:6007@10.111.0.144:5060
which I assume is correct. The second one contains:
sip:10.111.0.20:5060
But that (according to the dumps below) points to SER -- and that's the problem. Do you rewrite Contact header in SER ? If so then it should not point back to the SER instance.
SER treats the ACK correctly -- it has no Route headers and the Request-URI points to SER -- hence it will process it as local message.
What you need to have fixed is the IP and port in rewritten contact.
Jan.
Cesc wrote:
Tks ... i guess i will try to modify the config file so this ack problem is taken care of in the most generic way i can think of ...
What i tried and it seems it didn't work is to enter the "sip:10.111.0.20:5060" as a permanent entry in the location table (using serctl). This way my config, which does a lookup(location) and fails and drops the ACK, it would then return true and simply t_relay it ...
as for the sjphone, i don't blame the phone. I think it does reasonably good, given how bad the OK from the external.proxy gets to him ...
Cesc
On 10/11/06, Greger V. Teigre greger@teigre.com wrote:
Short answer: yes. Slightly longer: I have seen the same behavior. Without lr|lr=on the UA will (correctly) go over to strict routing and use contact in r-uri for the ACK. As long as only your record-route disappeared (i.e. you are the only hop in-between the UAS and UAC) this works. As your external proxy is B2BUA (at least for signalling), your fine if you relay the ACK. I'm not sure exactly when this happens and why, I seem to remember I posted something on this behavior for sjphone a while back, but I'm not capable of finding it. g-)
Cesc wrote: Hello everyone!
I have my system based on ser 0.9.6. Internal calls work just fine. I am now trying to interop with another system. It has a sort of asterisk functionality, but it is not asterisk, it is a private software company product.
my.phone ---------> my.ser.proxy -------------> external.proxy -------------> external phone 10.111.0.119 10.111.0.50 10.111.0.20 10.111.0.144
There is no firewall or nat in the way.
The problem, i think, is that the external.proxy is buggy. I told that to the company, but who knows when they will fix this.
- Look at the OK (message #9 and #10). My.ser.proxy record-routes all
invites. The Record-route headers reach the external.phone, which copies them in the OK message, sends the OK to the external.proxy ... and when this forwards it to my.ser.proxy, they are gone! The OK reaches my.phone, but then it generates my problem: the ACK. It contains NO ROUTE headers and the r-uri is also simply pointing to the external.proxy. If the ACK had the ROUTE headers, my.ser.proxy would loose_route the message and voila! But as loose_route() returns false, my (maybe bad) config file gets confused and treats it like a "new" call ... so it lookup("location") of the ACK r-ruri fails, and the ACK is dropped. Should i modify the config file so that ACK, if not loose_route'd, are simply t_relay'd?
- They also modify the contact field. The reason is because behind the
external.proxy could be H323 or SIP phones, so they sort of want to be a termination as far as signalling is concerned.
I attach the message flow, hope it helps ... Thanks!
Cesc ________________________________
No. Time Source Destination Protocol Info 1 0.000000000 10.111.0.119 10.111.0.50 SIP/SDP Request: INVITE sip:6007@10.111.0.20, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e Content-Length: 337 Contact: sip:7005@10.111.0.119:5060 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 70 To: sip:6007@10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 3 0.001209000 10.111.0.50 10.111.0.20 SIP/SDP Request: INVITE sip:6007@10.111.0.20:5060, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.20:5060 SIP/2.0 Message Header Record-Route: sip:10.111.0.50;ftag=12083790462495;lr=on Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
Content-Length: 337 Contact: sip:7005@10.111.0.119:5060 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 Content-Type: application/sdp CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 16 To: sip:6007@10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:98 iLBC/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 5 0.040629000 10.111.0.20 10.111.0.144 SIP/SDP Request: INVITE sip:6007@10.111.0.144:5060;transport=UDP, with session description Session Initiation Protocol Request-Line: INVITE sip:6007@10.111.0.144:5060;transport=UDP SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
RECORD-ROUTE: sip:10.111.0.50;ftag=12083790462495;lr=on From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Max-Forwards: 16 Contact: sip:10.111.0.20 User-Agent: SJphone/1.60.289a (SJ Labs) Content-Type: application/sdp Content-Length: 337 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3369481881 3369481881 IN IP4 10.111.0.119 Session Name (s): SJphone Connection Information (c): IN IP4 10.111.0.119 Time Description, active time (t): 0 0 Session Attribute (a): direction:active Media Description, name and address (m): audio 49248 RTP/AVP 3 97 98 8 0 101 Media Attribute (a): rtpmap:3 gsm/8000 Media Attribute (a): rtpmap:97 ilbc/8000 Media Attribute (a): rtpmap:98 ilbc/8000 Media Attribute (a): fmtp:98 mode=20 Media Attribute (a): rtpmap:8 pcma/8000 Media Attribute (a): rtpmap:0 pcmu/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16
No. Time Source Destination Protocol Info 9 3.736665000 10.111.0.144 10.111.0.20 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.20:5060;branch=z9hG4bKm27749469 Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
Record-Route: sip:10.111.0.50;ftag=12083790462495;lr=on Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Contact: sip:6007@10.111.0.144:5060 Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE Supported: 100rel, replaces Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 PCMA/8000
No. Time Source Destination Protocol Info 10 3.751306000 10.111.0.20 10.111.0.50 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.50;branch=z9hG4bK2eff.75fbc652.0 Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: sip:10.111.0.20:5060 Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info 11 3.751589000 10.111.0.50 10.111.0.119 SIP/SDP Status: 200 OK, with session description Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.111.0.119;rport=5060;branch=z9hG4bK0a6f0077000004b2452bb819000009f00000191e
From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 INVITE Contact: sip:10.111.0.20:5060 Content-Type: application/sdp Content-Length: 140 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): 6007 14194531 23224216 IN IP4 10.111.0.144 Session Name (s): SIP CALL Connection Information (c): IN IP4 10.111.0.144 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 10000 RTP/AVP 8 Media Attribute (a): rtpmap:8 pcma/8000
No. Time Source Destination Protocol Info 12 3.756127000 10.111.0.119 10.111.0.50 SIP Request: ACK sip:10.111.0.20:5060 Session Initiation Protocol Request-Line: ACK sip:10.111.0.20:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.111.0.119;rport;branch=z9hG4bK0a6f0077000004b2452bb81d00000c5f00001922 Content-Length: 0 Call-ID: D95A2208-3031-44E8-862E-2877A042900D@10.111.0.119 CSeq: 1 ACK From: "7005"sip:7005@10.111.0.50:5060;tag=12083790462495 Max-Forwards: 70 To: sip:6007@10.111.0.20;tag=0rxFXxkiza46soF7 User-Agent: SJphone/1.60.289a (SJ Labs)
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-
Cesc -
Greger V. Teigre -
Jan Janak