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&gt; 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&gt; 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 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/cg >> i-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 > >