Your work firewall must be blocking packets when you test on 3G. The Wifi must be within your work network !! I hope you are using RTPProxy or MediaProxy to handle media when originated from NATed clients. If yes, you dont need STUN and TURN as of now.
--- Jayesh On Fri, Mar 7, 2014 at 5:01 PM, Rajesh Babu <rajesh.b...@goodcoresoft.com>wrote: > Hi All, > > > > I use Opensips 1.9.1 and have enabled RTP and Nating in the > configuration, Whenever I use to connect the calls using my 3G connection, > call gets connected but my voice is not being heard, whereas though wifi > everything is working fine. I tried connecting with Linphone I didn't face > any issue, where as whenever I try connecting using my app which on top of > CSip I am getting this issue. This issue is not getting replicated over > wifi, I am getting this issue only on 3G. My carrier is not blocking any > packets from my side as different opensource client is letting me make > calls over the SIP. > > > > Some blogs stated that configuring Stun will solve this issue, I tried > doing it but no luck. In some other blog they where stating I can go with > TURN Server, I need to know whether Turn servers solve these issues and > someone can put me over the installation and using guide for the same. > > > > Can someone please direct me on the right track please? > > > > -Thanks > > Rajesh > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
_______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users