Hello,
I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says:
8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed
I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5.
The ACK package:
ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"sip:bob@net1.test;tag=GxzKy1nCMEI1mR0RztrB To: sip:alice@net1.test;tag=18823 Contact: "Bob"sip:bob@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp Max-Forwards: 69 Route: sip:mo@192.168.0.11:880 ;transport=ws;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3 Route: sip:mo@192.168.0.11:4060 ;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3 Route: sip:mo@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062 Route: sip:mt@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062 Route: sip:mt@192.168.0.11:4060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.1c3 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom
Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached.
Thanks in advance, Serhat
Hello,
can you get all the log messages for ACK but with debug=3 in the kamailio.cfg?
Cheers, Daniel
On 23/10/16 22:04, Serhat Guler wrote:
Hello,
I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says:
8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed
I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5.
The ACK package:
ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"sip:bob@net1.test;tag=GxzKy1nCMEI1mR0RztrB To: sip:alice@net1.test;tag=18823 Contact: "Bob"sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp Max-Forwards: 69 Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3 Route: sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3 Route: sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062 Route: sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062 Route: sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.1c3 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom
Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached.
Thanks in advance, Serhat
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
Hi Daniel,
Thanks for your reply. I actually attached a log file with debug level 3, consisting ACK related messages. If you would like to see more logs, I'll send a new log file in the evening.
Cheers, Serhat
On 24 October 2016 at 12:13, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
can you get all the log messages for ACK but with debug=3 in the kamailio.cfg?
Cheers, Daniel
On 23/10/16 22:04, Serhat Guler wrote:
Hello,
I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says:
8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed
I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5.
The ACK package:
ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS; rport From: "Bob"sip:bob@net1.test;tag=GxzKy1nCMEI1mR0RztrB To: sip:alice@net1.test;tag=18823 Contact: "Bob"sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no; click2call=no;transport=ws;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp Max-Forwards: 69 Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag= GxzKy1nCMEI1mR0RztrB;did=e82.0c3 Route: sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag= GxzKy1nCMEI1mR0RztrB;did=e82.0c3 Route: sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB; did=e82.f062 Route: sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB; did=e82.f062 Route: sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB; did=e82.1c3 User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom
Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached.
Thanks in advance, Serhat
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 Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
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
Hello,
I haven't noticed the log files, it's ok.
From the Route header, I see that there is a proxy that uses WS:
Route: sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3
That is the address of the next hop and typically a proxy doesn't use websocket connection to another proxy. Can you show a diagram with the sip server nodes in your network and what protocols are used between them?
Are you simply use record_route() function, or some other function or different parameters to it?
Cheers, Daniel
On 24/10/16 12:18, Serhat Guler wrote:
Hi Daniel,
Thanks for your reply. I actually attached a log file with debug level 3, consisting ACK related messages. If you would like to see more logs, I'll send a new log file in the evening.
Cheers, Serhat
On 24 October 2016 at 12:13, Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com> wrote:
Hello, can you get all the log messages for ACK but with debug=3 in the kamailio.cfg? Cheers, Daniel On 23/10/16 22:04, Serhat Guler wrote:Hello, I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says: 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5. The ACK package: ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB To: <sip:alice@net1.test>;tag=18823 Contact: "Bob"<sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> Max-Forwards: 69 Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.1c3> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached. Thanks in advance, Serhat _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>-- Daniel-Constantin Mierla http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org <mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
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Daniel-Constantin Mierla -
Serhat Guler