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: 4060The 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,SerhatOn 24 October 2016 at 20:18, Daniel-Constantin Mierla <miconda@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;trans
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?port=ws;r2=on;lr=on;ftag=GxzKy 1nCMEI1mR0RztrB;did=e82.0c3>
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> 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 found8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header8(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;rpor t 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 ;+g.oma.siall=no;transport=ws> p-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACKContent-Length:Max-Forwards: 69User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04Organization: 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/cg i-bin/mailman/listinfo/sr-user s -- 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_________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cg i-bin/mailman/listinfo/sr-user s -- 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