Hello,
We have a setup where Kamailio 4.2 is used in front of Asterisk as WebRTC Proxy doing the encryption and NAT Traversal.
Everything works as expected, except that BYE Requests sent by the WebRTC Client are not forwarded by Kamailio to Asterisk. We use record routing. Instead Kamailio responds with a "404 Not here".
INVITE Headers: Kamailio --> Browser:
sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com SIP/2.0 Record-Route: sip:104.155.11.255:5060;nat=yes;lr=on Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 Via: SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK2d4f70db Max-Forwards: 70 From: "Anonymous" sip:anonymous@anonymous.invalid;tag=as32dd9e24 To: sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com Contact: sip:anonymous@10.240.177.13:5060 Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.1 Date: Wed, 29 Oct 2014 10:04:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 676
BYE Headers Browser --> Kamailio:
BYE sip:anonymous@anonymous.invalid SIP/2.0 Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 , SIP/2.0/UDP 10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK2d4f70db From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com
;tag=2TQ878KMAVLA43TXVZHNAWCWVKU6BLPBURF3
To: "Anonymous" sip:anonymous@anonymous.invalid;tag=as32dd9e24 Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 0 BYE Record-Route: sip:104.155.11.255:5060;nat=yes;lr=on Reason: Q.850;cause=16
BYE Response Kamailio --> Browser:
SIP/2.0 404 Not here Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bK65df.6fd637d055286a45aa6f3e12c5ac873c.0 , SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK51c69f33;received=80.255.2.37 From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com
;tag=47J6E76F5EVB583A682FQR799J6XDSU46MW8
To: "Anonymous" sip:anonymous@anonymous.invalid;tag=as0b12a022 Call-ID: 683def08655393622c55299c1a7c92d5@10.240.177.13:5060 CSeq: 0 BYE Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0
In the logs I can see the following messages: [loose.c:113]: find_first_route(): No Route headers found [loose.c:929]: loose_route(): There is no Route HF
Does anybody know, why Kamailio may respond with 404 Not here? Do I have to send the BYE request directly to Asterisk?
Kind regards, Marko
Hello,
the BYE is coming with a rather strange R-URI. That should be taken from INVITE contact. Also, apparently the INVITE comes from behind NAT, you should use nat traversal logic to update the contact (e.g., add/set alias parameter).
See default configuration file for nat traversal, same should be applied here.
Cheers, Daniel
On 29/10/14 11:55, Marko Seidenglanz wrote:
Hello,
We have a setup where Kamailio 4.2 is used in front of Asterisk as WebRTC Proxy doing the encryption and NAT Traversal.
Everything works as expected, except that BYE Requests sent by the WebRTC Client are not forwarded by Kamailio to Asterisk. We use record routing. Instead Kamailio responds with a "404 Not here".
INVITE Headers: Kamailio --> Browser:
sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com mailto:sip%3Ae3W5LffMkg0PZjP3SIGf6@wh2.24dial.com SIP/2.0 Record-Route: sip:104.155.11.255:5060;nat=yes;lr=on Via: SIP/2.0/UDP 104.155.14.169:5064;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 Via: SIP/2.0/UDP 10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK2d4f70db Max-Forwards: 70 From: "Anonymous" sip:anonymous@anonymous.invalid;tag=as32dd9e24 To: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com mailto:sip%3Ae3W5LffMkg0PZjP3SIGf6@wh2.24dial.com> Contact: <sip:anonymous@10.240.177.13:5060 http://sip:anonymous@10.240.177.13:5060> Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 http://21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.1 Date: Wed, 29 Oct 2014 10:04:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 676
BYE Headers Browser --> Kamailio:
BYE sip:anonymous@anonymous.invalid SIP/2.0 Via: SIP/2.0/UDP 104.155.14.169:5064;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 , SIP/2.0/UDP 10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK2d4f70db From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com mailto:sip%3Ae3W5LffMkg0PZjP3SIGf6@wh2.24dial.com>;tag=2TQ878KMAVLA43TXVZHNAWCWVKU6BLPBURF3 To: "Anonymous" sip:anonymous@anonymous.invalid;tag=as32dd9e24 Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 http://21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 0 BYE Record-Route: sip:104.155.11.255:5060;nat=yes;lr=on Reason: Q.850;cause=16
BYE Response Kamailio --> Browser:
SIP/2.0 404 Not here Via: SIP/2.0/UDP 104.155.14.169:5064;branch=z9hG4bK65df.6fd637d055286a45aa6f3e12c5ac873c.0 , SIP/2.0/UDP 10.240.177.13:5060;received=127.0.0.1;branch=z9hG4bK51c69f33;received=80.255.2.37 From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com mailto:sip%3Ae3W5LffMkg0PZjP3SIGf6@wh2.24dial.com>;tag=47J6E76F5EVB583A682FQR799J6XDSU46MW8 To: "Anonymous" sip:anonymous@anonymous.invalid;tag=as0b12a022 Call-ID: 683def08655393622c55299c1a7c92d5@10.240.177.13:5060 http://683def08655393622c55299c1a7c92d5@10.240.177.13:5060 CSeq: 0 BYE Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0
In the logs I can see the following messages: [loose.c:113]: find_first_route(): No Route headers found [loose.c:929]: loose_route(): There is no Route HF
Does anybody know, why Kamailio may respond with 404 Not here? Do I have to send the BYE request directly to Asterisk?
Kind regards, Marko
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
Thank you for the fast reply,
I have enabled NAT Traversal like in the default config. The problem seems to be that the cannot be assigned to any transaction.
2014-10-29 13:08 GMT+01:00 Daniel-Constantin Mierla miconda@gmail.com:
Hello,
the BYE is coming with a rather strange R-URI. That should be taken from INVITE contact. Also, apparently the INVITE comes from behind NAT, you should use nat traversal logic to update the contact (e.g., add/set alias parameter).
See default configuration file for nat traversal, same should be applied here.
Cheers, Daniel
On 29/10/14 11:55, Marko Seidenglanz wrote:
Hello,
We have a setup where Kamailio 4.2 is used in front of Asterisk as WebRTC Proxy doing the encryption and NAT Traversal.
Everything works as expected, except that BYE Requests sent by the WebRTC Client are not forwarded by Kamailio to Asterisk. We use record routing. Instead Kamailio responds with a "404 Not here".
INVITE Headers: Kamailio --> Browser:
sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com SIP/2.0 Record-Route: sip:104.155.11.255:5060;nat=yes;lr=on Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 Via: SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK2d4f70db Max-Forwards: 70 From: "Anonymous" sip:anonymous@anonymous.invalid;tag=as32dd9e24 To: sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com Contact: sip:anonymous@10.240.177.13:5060 Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.1 Date: Wed, 29 Oct 2014 10:04:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 676
BYE Headers Browser --> Kamailio:
BYE sip:anonymous@anonymous.invalid SIP/2.0 Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 , SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK2d4f70db From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com
;tag=2TQ878KMAVLA43TXVZHNAWCWVKU6BLPBURF3
To: "Anonymous" sip:anonymous@anonymous.invalid;tag=as32dd9e24 Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 0 BYE Record-Route: sip:104.155.11.255:5060;nat=yes;lr=on Reason: Q.850;cause=16
BYE Response Kamailio --> Browser:
SIP/2.0 404 Not here Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bK65df.6fd637d055286a45aa6f3e12c5ac873c.0 , SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK51c69f33;received=80.255.2.37 From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com
;tag=47J6E76F5EVB583A682FQR799J6XDSU46MW8
To: "Anonymous" sip:anonymous@anonymous.invalid;tag=as0b12a022 Call-ID: 683def08655393622c55299c1a7c92d5@10.240.177.13:5060 CSeq: 0 BYE Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0
In the logs I can see the following messages: [loose.c:113]: find_first_route(): No Route headers found [loose.c:929]: loose_route(): There is no Route HF
Does anybody know, why Kamailio may respond with 404 Not here? Do I have to send the BYE request directly to Asterisk?
Kind regards, Marko
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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
The question is, how I can forward the BYE message back to asterisk?
2014-10-29 15:24 GMT+01:00 Marko Seidenglanz marko.seidenglanz@gmail.com:
Thank you for the fast reply,
I have enabled NAT Traversal like in the default config. The problem seems to be that the cannot be assigned to any transaction.
2014-10-29 13:08 GMT+01:00 Daniel-Constantin Mierla miconda@gmail.com:
Hello,
the BYE is coming with a rather strange R-URI. That should be taken from INVITE contact. Also, apparently the INVITE comes from behind NAT, you should use nat traversal logic to update the contact (e.g., add/set alias parameter).
See default configuration file for nat traversal, same should be applied here.
Cheers, Daniel
On 29/10/14 11:55, Marko Seidenglanz wrote:
Hello,
We have a setup where Kamailio 4.2 is used in front of Asterisk as WebRTC Proxy doing the encryption and NAT Traversal.
Everything works as expected, except that BYE Requests sent by the WebRTC Client are not forwarded by Kamailio to Asterisk. We use record routing. Instead Kamailio responds with a "404 Not here".
INVITE Headers: Kamailio --> Browser:
sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com SIP/2.0 Record-Route: sip:104.155.11.255:5060;nat=yes;lr=on Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 Via: SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK2d4f70db Max-Forwards: 70 From: "Anonymous" sip:anonymous@anonymous.invalid;tag=as32dd9e24 To: sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com Contact: sip:anonymous@10.240.177.13:5060 Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.13.1 Date: Wed, 29 Oct 2014 10:04:22 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 676
BYE Headers Browser --> Kamailio:
BYE sip:anonymous@anonymous.invalid SIP/2.0 Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bKa938.2add79aa083cd5776556ff0813878415.0 , SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK2d4f70db From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com
;tag=2TQ878KMAVLA43TXVZHNAWCWVKU6BLPBURF3
To: "Anonymous" sip:anonymous@anonymous.invalid;tag=as32dd9e24 Call-ID: 21cdb2366ccaab492fd1e69f590620b4@10.240.177.13:5060 CSeq: 0 BYE Record-Route: sip:104.155.11.255:5060;nat=yes;lr=on Reason: Q.850;cause=16
BYE Response Kamailio --> Browser:
SIP/2.0 404 Not here Via: SIP/2.0/UDP 104.155.14.169:5064 ;branch=z9hG4bK65df.6fd637d055286a45aa6f3e12c5ac873c.0 , SIP/2.0/UDP 10.240.177.13:5060 ;received=127.0.0.1;branch=z9hG4bK51c69f33;received=80.255.2.37 From: <sip:e3W5LffMkg0PZjP3SIGf6@wh2.24dial.com
;tag=47J6E76F5EVB583A682FQR799J6XDSU46MW8
To: "Anonymous" sip:anonymous@anonymous.invalid;tag=as0b12a022 Call-ID: 683def08655393622c55299c1a7c92d5@10.240.177.13:5060 CSeq: 0 BYE Server: kamailio (4.2.0 (x86_64/linux)) Content-Length: 0
In the logs I can see the following messages: [loose.c:113]: find_first_route(): No Route headers found [loose.c:929]: loose_route(): There is no Route HF
Does anybody know, why Kamailio may respond with 404 Not here? Do I have to send the BYE request directly to Asterisk?
Kind regards, Marko
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
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
-
Daniel-Constantin Mierla -
Marko Seidenglanz