[OpenSIPS-Users] SIP SDP rfc1918 address fix

Bogdan-Andrei Iancu bogdan at opensips.org
Thu Apr 27 05:50:02 EDT 2017


Satish,

I do not fine the err log you mentioned ("extract_mediaip: no `c=' in 
SDP") in the code of OpenSIPS - what version are you using ??

Also I tried to to inject your SDP into OpenSIPS 2.3 and I do not get 
the any errors.

Regards,

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

OpenSIPS Summit May 2017 Amsterdam
   http://www.opensips.org/events/Summit-2017Amsterdam.html

On 04/27/2017 03:43 AM, Satish Patel wrote:
> Yes, whenever fix_nated_sdp() fiction run it produce that error which I mentioned in my previous email. Every single time.
>
> Sent from my iPhone
>
>> On Apr 26, 2017, at 4:52 PM, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:
>>
>> So below is the SDP OpenSIPS receives (from network) and when doing fix_nated_sdp() on that SDP leads to the "c=" errors ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>   OpenSIPS Founder and Developer
>>   http://www.opensips-solutions.com
>>
>> OpenSIPS Summit May 2017 Amsterdam
>>   http://www.opensips.org/events/Summit-2017Amsterdam.html
>>
>>> On 04/26/2017 08:44 PM, Satish Patel wrote:
>>> Here is my payload again we have custom application which is using SER
>>> so some of them are custom values, This is the payload after i apply
>>> fix_nated_sdp() function.
>>>
>>>
>>> Max-Forwards: 16.
>>> Content-Type: application/sdp.
>>> Content-Length: 418.
>>> Supported: path, 100rel.
>>> P-hint: LOCAL.
>>> P-hint: ALIASED OUTBOUND.
>>> P-hint: DIRECT-RTP.
>>> .
>>> v=0.
>>> o=user1 53655765 2353687637 IN IP4 192.168.1.8.
>>> s=-.
>>> c=IN IP4 173.71.121.4.
>>> t=0 0.
>>> m=audio 6000 RTP/AVP 0.
>>> a=rtpmap:127 VANI/32000.
>>> a=fmtp:127 ver=3;mode=3;sub-types=1,7;codecs=0x26.
>>> a=rtpmap:111 SIREN14-3D/32000.
>>> a=fmtp:111 bitrate=32000.
>>> a=vx_payload_hdr_ver:2.
>>> a=rtpmap:0 PCMU/8000.
>>> a=vx_join_audio:1.
>>> a=vx_join_text:0.
>>> a=vx_jc:60.
>>> a=setup:both.
>>> a=vx_rtcp:0.
>>> a=direction:active.
>>> a=oldmediaip:192.168.1.8.
>>>
>>> On Wed, Apr 26, 2017 at 6:18 AM, Bogdan-Andrei Iancu
>>> <bogdan at opensips.org> wrote:
>>>> Hi Satish,
>>>>
>>>> For the mime test, you can use the has_body() function:
>>>> http://www.opensips.org/html/docs/modules/2.2.x/sipmsgops.html#idp3886992
>>>>
>>>> About the error - could you post the actual SDP payload generating those
>>>> errors ?
>>>>
>>>> Regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>>    OpenSIPS Founder and Developer
>>>>    http://www.opensips-solutions.com
>>>>
>>>> OpenSIPS Summit May 2017 Amsterdam
>>>>    http://www.opensips.org/events/Summit-2017Amsterdam.html
>>>>
>>>>
>>>>> On 04/25/2017 10:35 PM, Satish Patel wrote:
>>>>> We have some custome Voice solution and in-house media server so right
>>>>> now i don't care about PORT all i need correct IP address.
>>>>>
>>>>> I have tried following and it fixed issue but i am seeing following
>>>>> error in logs
>>>>>
>>>>> if (method=="INVITE") {
>>>>>           if(search("^Content-Type:.*application/sdp")) {
>>>>>           fix_nated_sdp("3");
>>>>>           };
>>>>> };
>>>>>
>>>>>
>>>>> Error:
>>>>>
>>>>> ERROR: extract_mediaip: no `c=' in SDP
>>>>> ERROR: extract_mediaip: no `c=' in SDP
>>>>>
>>>>> Do you know what does that means and how to fix that issue?
>>>>>
>>>>> On Mon, Apr 24, 2017 at 11:41 PM, Alex Balashov
>>>>> <abalashov at evaristesys.com> wrote:
>>>>>> The intent of my questions was to get what you think about what you
>>>>>> actually want to accomplish. fix_nated_sdp() allows you to replace the
>>>>>> IP with the received signalling IP:
>>>>>>
>>>>>> http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899
>>>>>>
>>>>>> But what about the port?
>>>>>>
>>>>>>> On Mon, Apr 24, 2017 at 11:39:14PM -0400, Satish Patel wrote:
>>>>>>>
>>>>>>> after google found bunch of post where people suggesting use
>>>>>>> fix_nated_sdp()  is that right approach ?
>>>>>>>
>>>>>>> On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov
>>>>>>> <abalashov at evaristesys.com> wrote:
>>>>>>>> Yes, but RTP can come from a different address than the signalling
>>>>>>>> (SIP). What sense would there be in substituting the source of the SIP
>>>>>>>> message in there?
>>>>>>>>
>>>>>>>>> On Mon, Apr 24, 2017 at 11:23:30PM -0400, Satish Patel wrote:
>>>>>>>>>
>>>>>>>>> I meant "origin public address of client"  if c line isn't public then
>>>>>>>>> media never work.
>>>>>>>>>
>>>>>>>>> c=IN IP4 192.168.1.8.
>>>>>>>>>
>>>>>>>>> It should be
>>>>>>>>>
>>>>>>>>> c=IN IP4 <public_ip_of_client>
>>>>>>>>>
>>>>>>>>> On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov
>>>>>>>>> <abalashov at evaristesys.com> wrote:
>>>>>>>>>> Satish,
>>>>>>>>>>
>>>>>>>>>> When you say "origin public address", do you mean the external source
>>>>>>>>>> address and port of the SIP message, or the incoming RTP stream?
>>>>>>>>>>
>>>>>>>>>>> On Mon, Apr 24, 2017 at 11:00:40PM -0400, Satish Patel wrote:
>>>>>>>>>>>
>>>>>>>>>>> In my INVITE/SDP i am seeing sometime rfc1918 address which i want
>>>>>>>>>>> fix
>>>>>>>>>>> and replace it with origin public address. ex
>>>>>>>>>>>
>>>>>>>>>>> I am seeing following info in INVITE
>>>>>>>>>>>
>>>>>>>>>>> v=0.
>>>>>>>>>>> o=amsip 0 0 IN IP4 192.168.1.8.
>>>>>>>>>>> s= .
>>>>>>>>>>> c=IN IP4 192.168.1.8.
>>>>>>>>>>> t=0 0.
>>>>>>>>>>> m=audio 22530 RTP/AVP 127 111 0 101.
>>>>>>>>>>>
>>>>>>>>>>> _______________________________________________
>>>>>>>>>>> Users mailing list
>>>>>>>>>>> Users at lists.opensips.org
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>>>>>>>>>> --
>>>>>>>>>> Alex Balashov | Principal | Evariste Systems LLC
>>>>>>>>>>
>>>>>>>>>> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
>>>>>>>>>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>>>>>>>>>
>>>>>>>>>> _______________________________________________
>>>>>>>>>> Users mailing list
>>>>>>>>>> Users at lists.opensips.org
>>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>> _______________________________________________
>>>>>>>>> Users mailing list
>>>>>>>>> Users at lists.opensips.org
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>>>>>>>> --
>>>>>>>> Alex Balashov | Principal | Evariste Systems LLC
>>>>>>>>
>>>>>>>> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
>>>>>>>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing list
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>>>>>>> _______________________________________________
>>>>>>> Users mailing list
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>>>>>> --
>>>>>> Alex Balashov | Principal | Evariste Systems LLC
>>>>>>
>>>>>> Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
>>>>>> Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org
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