[SR-Users] Modifying SDP in Kamailio

[Frank Carmickle]

On May 18, 2014, at 2:34 AM, MrIhaveAnOpinionOnEverything <melryanf at 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