What parts of the SDP payload are most important when manipulating "Far End" NAT traversal. How/where do we manipulate the Contact header as such that it does not effect preceding loose routes.
A lot of examples cover an OpenSIPS+RTPProxy on a public IP that must handle UA behind NAT. The problem we are having is that even our OpenSIPS+RTPProxy server is behind a NAT, and getting dead ear when using RTP proxy, and one way (outbound) audio when not using RTP proxy implementation. Please help.... Nick. On 3/15/13, Ovidiu Sas <o...@voipembedded.com> wrote: > Contact is just a SIP header that you can manipulate via > transformations and sipmsgops functions. > You can even remove the received header and build/append a new one. > > Regards, > Ovidiu Sas > > On Wed, Mar 13, 2013 at 1:13 PM, Nick Khamis <sym...@gmail.com> wrote: >> Hello Ovidiu, >> >> Thank you so much for your response. I got blind sided when we took on >> this new DID service provider. What is strange is that the same >> configuration works fine with three other providers.. Anyhow, c and o >> are squared away however, I would also like to fix the contact using >> fix_contact(). Is there any way I can pass the public IP to the >> function? Almost there I hope.... :) >> >> Nicholas. >> >> On 3/13/13, Ovidiu Sas <o...@voipembedded.com> wrote: >>> When calling rtpproxy_*, use flag 'c' along with the external IP as a >>> second parameter: >>> http://www.opensips.org/html/docs/modules/devel/rtpproxy#id292744 >>> >>> Regards, >>> Ovidiu Sas >>> >>> -- >>> VoIP Embedded, Inc. >>> http://www.voipembedded.com >>> >>> On Tue, Mar 12, 2013 at 11:38 PM, Nick Khamis <sym...@gmail.com> wrote: >>>> Hello Everyone, >>>> >>>> We recently took on a new DID provider that is complaining about the >>>> "c=IN IP4 192.168.2.5." advertised by our opensips server. They are >>>> telling us that is the reason we are not receiving any audio. For some >>>> reason this was not an issue with other service providers. >>>> Moving forward, is it possible to change the "c=in" of the opensips >>>> server before sending it to our service provider? >>>> >>>> I have seen how nat has been fixed for UA but no example for OpenSIPS >>>> herself. As I mentioned OpenSIPS and RTPProxy is behind the NAT box. >>>> >>>> Thanks in Advance, >>>> >>>> Nick. > > _______________________________________________ > 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