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 <srtgu...@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 <mico...@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 <mico...@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:b...@net1.test>;tag=GxzKy1nCMEI1mR0RztrB
>>> To: <sip:al...@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-us...@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@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
>>
>>
>
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