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

Bogdan-Andrei Iancu bogdan at opensips.org
Wed Aug 19 19:23:36 CEST 2015


Nabeel,

What is broken is broken and cannot be unbroken :)

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 19.08.2015 18:48, Nabeel wrote:
>
> 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 
> <mailto: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 Developer
>     http://www.opensips-solutions.com
>
>     On 19.08.2015 17:49, Nabeel wrote:
>>
>>     I used that same client 'Lumicall' (lumicall.org
>>     <http://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
>>     <mailto: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 Developer
>>         http://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 <mailto: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
>>>             <http://10.61.208.143>:*46032:-1*;rport;branch=z9hG4bK78006
>>>
>>>             Regards,
>>>
>>>             Bogdan-Andrei Iancu
>>>             OpenSIPS Founder and Developer
>>>             http://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
>>>>             <mailto: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
>>>>                     <mailto:sip%3A%2B447984977774 at sipdomain.com>;transport=tcp>#015#012From:
>>>>                     <sip:+447479189410 at sipdomain.com
>>>>                     <mailto:sip%3A%2B447479189410 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
>>>>                     <tel:%2B447479189410>@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
>>>>                     <http://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
>>>>                     <mailto:sip%3A%2B447984977774 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
>>>>                     <mailto:sip%3A%2B447984977774 at sipdomain.com>;transport=tcp>#015#012From:
>>>>                     <sip:+447479189410 at sipdomain.com
>>>>                     <mailto:sip%3A%2B447479189410 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
>>>>                     <tel:%2B447479189410>@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
>>>>                     <http://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 <mailto:bogdan at opensips.org>>
>>>>                 wrote:
>>>>
>>>>                     yes, you should.
>>>>
>>>>                     Regards,
>>>>
>>>>                     Bogdan-Andrei Iancu
>>>>                     OpenSIPS Founder and Developer
>>>>                     http://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
>>>>>                     <mailto: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 Developer
>>>>>                         http://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
>>>>>>                         <mailto: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 Developer
>>>>>>                             http://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
>>>>>>>                             <mailto: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 Developer
>>>>>>>                                 http://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 list
>>>>>>>>                                 Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>>>>>>>                                 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>
>>
>

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