On May 18, 2014, at 2:34 AM, MrIhaveAnOpinionOnEverything <melry...@gmail.com> wrote:
> Hi guys: > > I am a R&D engineer trying to learn kamailio. After following some > tutorials and reading the thread in this mailing list I was able to setup a > voip backend with this configuration > > > XLITE/LINPHONE ---> KAMAILIO ----> FREESWITCH > > I am using Freeswitch as a media server. After configuring RTP Proxy and > kamailio to use bridged mode. I was able to successfully setup a voip backend > like the one above. > > I encountered a problem when the UAC I am using is a webclient like sipml5. > > I noticed that when SIP INVITES from KAMAILIO to FREESWITCH are being > passed when a INVITE transaction is initiated from a sipml5 client FREESWITCH > is trying to use the public ip of webrtc server of the sipml5 backend. > Unfortunately, I am using private ip/LAN IP between kamailio and freeswitch. > As a result calls are established but there is no audio that is happening. > I think you're confused, unless I'm confused. What I see from reading the traces is that freeswitch is offering media on a rfc1918 address. You either need to static NAT a non rfc1918 address to freeswitch or allow it to bind one directly. You can use the ext-rtp-ip sofia parameter on your profile if you aren't binding directly. HTH --FC _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
