Hi Daniel,
I am using only record_route() without any parameters. I do not have a
proper computer atm to draw the network diagram, but I can tell you shortly
about the network setup.
I have only enabled websockets for the pcscf to allow ws and wss
connections. In that case there is a ws connection that uses UDP protocol.
This is the ACK to complete the session setup.
the sipml5 client is configured as follows:
WebSocket Server URL: ws://192.168.0.11:880
SIP outbound Proxy URL: udp://192.168.0.11:4060
Mercuro IMS client: uses UDP port as well: 4060
The call is made from sipml5 client. The Mercuro phone rings, and when I
reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from
sipml5 doesn't pass the PCSCF as I explained in the previous message.
A part of PCSCF cfg file:
# Check for Subsequent requests:
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
if (loose_route()) {
if ($route_uri =~ "sip:mo@.*") {
setflag(FLT_MO);
}
if(!isdsturiset()) {
handle_ruri_alias();
}
# RTP-Relay, if necessary
route(RTPPROXY);
t_relay();
} else {
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
t_relay();
exit;
} else {
xlog("L_INFO", "ACK without matching transaction ...
ignore and discard!!!!!\n");
# ACK without matching transaction ... ignore and
discard
exit;
}
}
sl_send_reply("404","Not here");
}
exit;
}
Cheers,
Serhat
On 24 October 2016 at 20:18, Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
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=GxzKy
1nCMEI1mR0RztrB;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
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=z9
hG4bKvuly7bmxnN4aqM4zZTIS;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;click2c
all=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=GxzKy
1nCMEI1mR0RztrB;did=e82.0c3>
Route: <sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz
trB;did=e82.0c3>
Route: <sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di
d=e82.f062>
Route: <sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di
d=e82.f062>
Route: <sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di
d=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
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--
Daniel-Constantin
Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Oct 24-26, 2016 -
http://www.asipto.com
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--
Daniel-Constantin
Mierlahttp://twitter.com/#!/miconda -
http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Oct 24-26, 2016 -
http://www.asipto.com