After struggling with one way audio issues as a result of STUN binding
errors on both the Asterisk side and the Chrome side, we've decided to just
simply go with a TURN relay for RTP packets until the issues are resolved.
I configured rtp.conf so that all of the STUN related entries are commented
out, and I use the following TURN configuration instead:
turnaddr=numb.viagenie.ca:3478
;
; Username used to authenticate with TURN relay server.
turnusername=myusername%40gmail.com
;
; Password used to authenticate with TURN relay server.
turnpassword=p@ssw0rd
I also use the same configuration on the client side. When running a
tcpdump, I see that there is traffic to/from the TURN relay:
10.0.1.18.53875 > blues.viagenie.ca.nat-stun-port: UDP, length 20
blues.viagenie.ca.nat-stun-port > 10.0.1.18.53875: UDP, length 56
10.0.1.18.51435 > blues.viagenie.ca.nat-stun-port: UDP, length 28
blues.viagenie.ca.nat-stun-port > 10.0.1.18.51435: UDP, length 100
10.0.1.18.51435 > blues.viagenie.ca.nat-stun-port: UDP, length 144
10.0.1.18.51435 > blues.viagenie.ca.nat-stun-port: UDP, length 144
blues.viagenie.ca.nat-stun-port > 10.0.1.18.51435: UDP, length 100
But it's dead silent when doing a tcpdump on the Asterisk server side.
The candidates on both sides don't contain relay candidates. Oddly, the
client side still has srflx candidates, suggesting STUN is still at work,
but the Asterisk side only contains host candidates.
Is TURN fully enabled in Asterisk 11? If so, how does one enable it and
make it the priority?
Thank you,
--
James Mortensen
Project Manager, VoiceCurve, Inc.
866-707-4590
james.morten...@voicecurve.com
--
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