[SR-Users] PCSCF cannot create via header for sipml5 ACK package
Alberto Llamas
albertollamaso at gmail.com
Sun Oct 30 21:35:44 CET 2016
Excellent !
On Sun, Oct 30, 2016 at 8:33 PM, Serhat Guler <srtguler at 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 at 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 at 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 at 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 at .*") {
>>>> 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 at 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 at 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 at 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 at 192.168.0.10:49794;transport=udp SIP/2.
>>>>>> Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9
>>>>>> hG4bKvuly7bmxnN4aqM4zZTIS;rport
>>>>>> From: "Bob"<sip:bob at net1.test>;tag=GxzKy1nCMEI1mR0RztrB
>>>>>> To: <sip:alice at net1.test>;tag=18823
>>>>>> Contact: "Bob"<sip:bob at 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 at 192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy
>>>>>> 1nCMEI1mR0RztrB;did=e82.0c3>
>>>>>> Route: <sip:mo at 192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz
>>>>>> trB;did=e82.0c3>
>>>>>> Route: <sip:mo at 192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di
>>>>>> d=e82.f062>
>>>>>> Route: <sip:mt at 192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di
>>>>>> d=e82.f062>
>>>>>> Route: <sip:mt at 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 at 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 at 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 at 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 at 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 at 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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