Try delete nat from 770000wrtc settings ice should do the same On Aug 11, 2016 10:00 PM, "Jonas Kellens" <jonas.kell...@telenet.be> wrote:
> On 11-08-16 18:03, Matt Fredrickson wrote: > >> On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kell...@telenet.be> >> wrote: >> >>> 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 ?? >>> >> If you want to minimize grief, start with 13 - WebRTC has been a >> moving target for the last 5 years, it is not an old, mature standard >> like ISDN or SIP. If you find interop problems in an older version of >> Asterisk with WebRTC, it's likely that it has been fixed in 13, and if >> it hasn't the most likely place to obtain the fix will be in 13. >> >> After you get the WebRTC part working, then you can move back the >> versions of Asterisk you're using to see if it still works. >> >> As far as ICE not working goes, if the browser you're talking to is >> not on the same network 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 issue of Asterisk > freeze. This is all EXTREMELY BUGGY !) > > > Asterisk 13 here I come (with very high expectations). > > > Kind regards. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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