[OpenSIPS-Users] SIP SDP rfc1918 address fix

Wed Apr 26 13:44:16 EDT 2017

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
>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>> --
>>>>>>> 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
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>> --
>>>>> 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
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> --
>>> 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
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>