Hello there, 

I am trying to setup a Kamailio (3.3) + Linphone + Jitsi based private calling 
system (users can call each other but not call out of the network). 

Agents can be behind NAT (think Verizon cellphone or home-users behind their 
router). I started with the default setup and have tried different settings for 
NAT/rtpproxy. Currently, all my users register onto the Kamailio server. Users 
who are not behind a NAT (Jitsi on the internet) i.e. with Public IPs are able 
to successfully make audio/video calls. 

I have RTP proxy setup / running as well, and calls to rtpproxy_manage() as 
required. The process is running and Kamailio is configured to know that the 
RTPproxy is listening.

I had to switch to using TCP on some of the my UAs because of provider blocking 
UDP traffic. So they are able to register, yet calls are not completed. The 
call looks like it connects but no media (audio or video goes through). 

I have read a lot of the forums on NAT - and am trying to solve for a couple of 
big issues:
1. Figuring out when a UA is behind a NAT, and handling the RTPProxying 
appropriately. 
2. while ngrep-ing UDP packets occassionally I see "Warning: 399 sipalg 
"Unauthorized" or Bad request.  based on what I saw on the internet, it has to 
do with the service provider (Verizon) messing with the packet. Is there a way 
arount this?

Any help is greatly appreciated! 

Thanks
Pranv
_______________________________________________
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

Reply via email to