[OpenSIPS-Users] ERROR:rtpproxy:force_rtp_proxy: Unable to parse body

Nabeel nabeelshikder at gmail.com
Wed Aug 19 17:48:13 CEST 2015


Well, at least the call connects when the other SIP server accepts the
port.  But OpenSIPS fails to connect.    Is there a  way to relax the port
criteria in OpenSIPS to accept such ports?  Even if the port is in a
'bogus' format?
On 19 Aug 2015 16:17, "Bogdan-Andrei Iancu" <bogdan at opensips.org> wrote:

> Make a network capture - if the incoming CANCEL has that broken VIA port,
> simply report it to the vendor. The fact the bogus port is accepted by
> other SIP server implementations is not really relevant.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 19.08.2015 17:49, Nabeel wrote:
>
> I used that same client 'Lumicall' (lumicall.org) with Repro SIP server
> and the calls worked, so I'm not sure how to fix it.
> On 19 Aug 2015 15:24, "Bogdan-Andrei Iancu" <bogdan at opensips.org> wrote:
>
>> Fix the SIP client sending you the bogus SIP messages :). You cannot fix
>> anything on OpenSIPS side.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 19.08.2015 17:16, Nabeel wrote:
>>
>> Okay, so what caused the bogus port nunber and how can I fix it?
>> On 19 Aug 2015 14:54, "Bogdan-Andrei Iancu" <bogdan at opensips.org> wrote:
>>
>>> Hi Nabeel,
>>>
>>> This error has nothing to do with the rtpproxy . It is a parsing error
>>> of an incoming SIP request. The VIA header in the received CANCEL has a
>>> bogus port number:
>>>     Via: <SIP/2.0/TCP 10.61.208.143:*46032:-1*;rport;branch=z9hG4bK78006
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 19.08.2015 09:02, Nabeel wrote:
>>>
>>> I'm using port 12221 for rtpproxy (from default script) and port 12223
>>> for OpenSIPS.  Are those 'bad ports' like the error says?
>>> On 19 Aug 2015 06:48, "Nabeel" <nabeelshikder at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> The rtpproxy error is resolved with your suggestion, but I still get
>>>> the 'bad port' errors in the log followed by CANCEL of the call:
>>>>
>>>> ERROR:core:parse_via: bad port
>>>>> ERROR:core:parse_via:  <SIP/2.0/TCP 10.61.208.143:46032:-1;rport;branch=z9hG4bK78006#015#012Max-Forwards:
>>>>> 70#015#012To: <sip:+447984977774 at sipdomain.com;transport=tcp>#015#012From:
>>>>> <sip:+447479189410 at sipdomain.com>;tag=z9hG4bK93013084#015#012Call-ID:
>>>>> 358919227046 at 10.61.208.143#015#012CSeq
>>>>> <http://358919227046@10.61.208.143#015%23012CSeq>: 1
>>>>> CANCEL#015#012Contact: <sip:+447479189410 at 10.61.208.143:46032;transport=tcp>#015#012Expires:
>>>>> 3600#015#012User-Agent: Agent/1.3.4/MP-S168#015#012Content-Length:
>>>>> 0#015#012#015#012>
>>>>> ERROR:core:parse_via: parsed so far:<SIP/2.0/TCP 10.61.208.143:46032>
>>>>> ERROR:core:get_hdr_field: bad via
>>>>> DBG:core:set_err_info: ec: 1, el: 3, ei: 'error parsing Via'
>>>>> DBG:core:get_hdr_field: error exit
>>>>> INFO:core:parse_headers: bad header field
>>>>> ERROR:core:parse_msg: message=<CANCEL sip:+447984977774 at sipdomain.com;transport=tcp
>>>>> SIP/2.0#015#012Via: SIP/2.0/TCP 10.61.208.143:46032:-1;rport;branch=z9hG4bK78006#015#012Max-Forwards:
>>>>> 70#015#012To: <sip:+447984977774 at sipdomain.com;transport=tcp>#015#012From:
>>>>> <sip:+447479189410 at sipdomain.com>;tag=z9hG4bK93013084#015#012Call-ID:
>>>>> 358919227046 at 10.61.208.143#015#012CSeq
>>>>> <http://358919227046@10.61.208.143#015%23012CSeq>: 1
>>>>> CANCEL#015#012Contact: <sip:+447479189410 at 10.61.208.143:46032;transport=tcp>#015#012Expires:
>>>>> 3600#015#012User-Agent: Agent/1.3.4/MP-S168#015#012Content-Length:
>>>>> 0#015#012#015#012>
>>>>> ERROR:core:receive_msg: Unable to parse msg received from [
>>>>> 188.29.165.141:46315]
>>>>> ERROR:core:tcp_handle_req: receive_msg failed
>>>>
>>>>
>>>>
>>>> The SIP trace of the call is here:
>>>>
>>>> http://pastebin.com/dNpau6GT
>>>>
>>>> I am using the default config scripts.  Please advise how to fix this.
>>>>
>>>>
>>>>
>>>>
>>>> On 14 August 2015 at 18:36, Bogdan-Andrei Iancu <bogdan at opensips.org>
>>>> wrote:
>>>>
>>>>> yes, you should.
>>>>>
>>>>> Regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>
>>>>> On 14.08.2015 18:06, Nabeel wrote:
>>>>>
>>>>> Should I make the same change to rtpproxy_offer?
>>>>>
>>>>>     if (is_method("INVITE")) {
>>>>>                if (isflagset(NAT)) {
>>>>>                       rtpproxy_offer("ro");
>>>>>                }
>>>>> On 14 Aug 2015 13:09, "Bogdan-Andrei Iancu" <bogdan at opensips.org>
>>>>> wrote:
>>>>>
>>>>>> I see....you have a onreply_route[handle_nat] in your script, doing:
>>>>>>
>>>>>>     if ( isflagset(NAT) )
>>>>>>         rtpproxy_answer("ro");
>>>>>>
>>>>>> Change that to :
>>>>>>
>>>>>>     if ( isflagset(NAT) && has_body("application/sdp") )
>>>>>>         rtpproxy_answer("ro");
>>>>>>
>>>>>> Regards,
>>>>>>
>>>>>> Bogdan-Andrei Iancu
>>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>>
>>>>>> On 14.08.2015 15:04, Nabeel wrote:
>>>>>>
>>>>>> The 'ringing' stage of a call is when the error occurs.  Why should
>>>>>> 180 Ringing lead to an error?
>>>>>> On 14 Aug 2015 12:47, "Bogdan-Andrei Iancu" <bogdan at opensips.org>
>>>>>> wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> ACK request or 180 ringing reply are part of a call and they do not
>>>>>>> have a body.
>>>>>>>
>>>>>>> It depends on your scripting to see when the rtpproxy are called
>>>>>>> (for what sip message).
>>>>>>>
>>>>>>> You may use the script_trace() function :
>>>>>>>
>>>>>>> http://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc42
>>>>>>> to see which sip messages to generate the err log.
>>>>>>>
>>>>>>> Regards,
>>>>>>>
>>>>>>> Bogdan-Andrei Iancu
>>>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>>>
>>>>>>> On 14.08.2015 14:15, Nabeel wrote:
>>>>>>>
>>>>>>> Hi Bogdan,
>>>>>>>
>>>>>>> Thanks, but I had no intentions of using a SIP message without a
>>>>>>> body; all I'm trying to do is make a normal call with OpenSIPS.
>>>>>>>
>>>>>>> Please explain why I'm getting a SIP message without a body and how
>>>>>>> do I fix it?
>>>>>>> On 14 Aug 2015 11:12, "Bogdan-Andrei Iancu" <bogdan at opensips.org>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Hi Nabeel,
>>>>>>>>
>>>>>>>> You may get this error when calling the rtpproxy functions for a
>>>>>>>> SIP message without a body.
>>>>>>>>
>>>>>>>> Regards,
>>>>>>>>
>>>>>>>> Bogdan-Andrei Iancu
>>>>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>>>>
>>>>>>>> On 14.08.2015 09:58, Nabeel wrote:
>>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>> I am getting this error when making some calls:
>>>>>>>>
>>>>>>>> ERROR:rtpproxy:force_rtp_proxy: Unable to parse body
>>>>>>>>
>>>>>>>> I think this may be related to 'rtpproxy_offer' and
>>>>>>>> 'rtpproxy_answer' in the config file but I don't know how to fix it.  I am
>>>>>>>> using the default OpenSIPS config.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
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