search for: jzeoet2v3qb0y27grntt2u2paya

...ork as the Asterisk server, it's *possible* you > might need a true TURN server as well, instead of just an ICE server. > > Matthew Fredrickson > Matthew when I set the following in rtp.conf : turnaddr=192.158.29.39:3478?transport=udp turnusername=28224511:1379330808 turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA then Asterisk 12 gets really slow and sometimes unresponsive. Calls result in 480 request timeout (possibly due to the freeze of Asterisk). So this is also no solution. Can not even test if it brings me some audio in my webRTC calls. (putting the above lines back in comment resolves the issu...

...ver. > > Matthew Fredrickson > > > Matthew > > when I set the following in rtp.conf : > > turnaddr=192.158.29.39:3478?transport=udp > <http://192.158.29.39:3478?transport=udp> > turnusername=28224511:1379330808 > turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA > > > then Asterisk 12 gets really slow and sometimes unresponsive. > Calls result in 480 request timeout (possibly due to the freeze of > Asterisk). > > So this is also no solution. > > Can not even test if it brings me some audio in my webRTC calls....

My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache :-) I will do so if there is no other option. But still, I don't see why Ast 13 would differ so much in this case ? If ICE and NAT is working (not causing problems) why should Ast 13 bring me audio and Ast 12 don't