Question : I noticed I received an error when installing pjproject
--with-external-srtp
I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")
Can this have anything to do with the no-audio-problems that I'm having ??
Kind regards.
On 12-08-16 15:02, Jonas Kellens wrote:
Hello
setting "nat=no" or omitting "nat=" in peer definition does not help
either. Still no audio.
Why do you think this is a NAT issue ? IP and port information in
SDP-body is correct.
Kind regards.
On 12-08-16 09:25, Антон Сацкий wrote:
Try delete nat from 770000wrtc settings ice should do the same
On Aug 11, 2016 10:00 PM, "Jonas Kellens" <[email protected]
<mailto:[email protected]>> wrote:
On 11-08-16 18:03, Matt Fredrickson wrote:
On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens
<[email protected] <mailto:[email protected]>>
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
<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.
(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.
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