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)
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