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 am attaching a snapshot of the ngrep that on kamailio and freeswitch
server for your reference.
I would like to know if there is a setting in kamailio that would allow
me to modify the IP in the "o" and "c" sdp parameter when forwarding an
invite to Freeswitch.
I did another test. XLITE ---> KAMAILIO ---> FREESWITCH ----> KAMAILIO
----> sipml5 And the call works. It has audio. I think it must have
something to do with the SDP header that is being generated by sip5ml UAC
that is conflicting with my setup.
Any help or advice will be greatly appreciated. Thanks.
--
"When I look at you I see two people, ther person you are
and the person you are supposed to be. Someday these two
people will meet. And when they do, they will achieve great things"
- Gene Hackman, The Replacements
Catch up on me and check my BLOG at
http://mrihaveanopiniononeverything.blogspot.com
freeswitch.log
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kamailio.log
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