[OpenSIPS-Users] SIP SDP rfc1918 address fix
Satish Patel
satish.txt at gmail.com
Fri Apr 28 09:24:48 EDT 2017
Unfortunately we are using SER 0.10
Sent from my iPhone
> On Apr 27, 2017, at 5:50 AM, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:
>
> 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.
>>>>>>>>>>>>
>>>>>>>>>>>> _______________________________________________
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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
>>>>>>>>>> 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/
>>>>>>>>>
>>>>>>>>> _______________________________________________
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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/
>>>>>>>
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>>>>>>> Users mailing list
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