23 Oct
2016
23 Oct
'16
11:04 p.m.
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
24 Oct
24 Oct
1:13 p.m.
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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
1:18 p.m.
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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
9:18 p.m.
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(a)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:
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com
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(a)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(a)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>
2951
days inactive
2952
days old
3 comments
2 participants
participants (2)
-
Daniel-Constantin Mierla -
Serhat Guler