Hi Alberto,
Removing the outbound proxy solved the problem. Thanks for your help.
Cheers,
Serhat
On 30 October 2016 at 10:52, Alberto Llamas <albertollam...@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 <srtgu...@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 <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
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> 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)
>
>
> _______________________________________________
> 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
>
>
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