Unfortunately we are using SER 0.10 Sent from my iPhone
> On Apr 27, 2017, at 5:50 AM, Bogdan-Andrei Iancu <bog...@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 <bog...@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 >>>> <bog...@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 >>>>>> <abalas...@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 >>>>>>>> <abalas...@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 >>>>>>>>>> <abalas...@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@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@lists.opensips.org >>>>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>>>>>> _______________________________________________ >>>>>>>>>> Users mailing list >>>>>>>>>> Users@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@lists.opensips.org >>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>>>> _______________________________________________ >>>>>>>> Users mailing list >>>>>>>> Users@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@lists.opensips.org >>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>>> _______________________________________________ >>>>>> Users mailing list >>>>>> Users@lists.opensips.org >>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users