On May 18, 2014, at 2:34 AM, MrIhaveAnOpinionOnEverything melryanf@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 configurationXLITE/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