Excellent !


On Sun, Oct 30, 2016 at 8:33 PM, Serhat Guler <srtguler@gmail.com> wrote:
Hi Alberto,

Removing the outbound proxy solved the problem. Thanks for your help.

Cheers,
Serhat

On 30 October 2016 at 10:52, Alberto Llamas <albertollamaso@gmail.com> wrote:
Hi Serhat,

I am not sure how is the setup of your network, but you should remove the outbound proxy setting from sipml5 (SIP outbound Proxy URL: udp://192.168.0.11:4060).

Test it and let us know.

Regards,

On Sat, Oct 29, 2016 at 9:38 PM, Serhat Guler <srtguler@gmail.com> wrote:
Hi all,

I am still stuck with the ACK message not being forwarded by the originating PCSCF. Any advice would be great.

Thanks,
Serhat

On 24 October 2016 at 21:00, Serhat Guler <srtguler@gmail.com> wrote:
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@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=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> 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:

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
Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6
CSeq: 3887 ACK
Content-Length: 
Max-Forwards: 69
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

-- 
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/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



_______________________________________________
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




--
Alberto Llamas
Telecommunications Engineer
Digium Certified Asterisk Professional (dCap)


_______________________________________________
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



_______________________________________________
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




--
Alberto Llamas
Phone: +1-786-805-6003
Telecommunications Engineer
Digium Certified Asterisk Professional (dCap)