On 04/24/2018 09:08 AM, Matt Fredrickson wrote:
On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell <[email protected]> wrote:
A while back (last year maybe?), there was a Digium blog post on setting up
WebRTC.
I was never able to get that working.
I was working with Asterisk 15 on a RHEL derived distro and had no idea of
where to go to shoot the failure.
Has anyone got a tutorial with trouble shooting?
Great question! I'm assuming you're talking about the SFU blog post -
http://blogs.asterisk.org/2017/09/20/asterisk-15-multi-stream-media-sfu/
?
I'd be curious as to what difficulties you ran into. We actually need
to try to consolidate the information in that post with the webrtc
setup page on the wiki -
https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support
You might try those two pages if you haven't yet. If you have
already, perhaps posting your specific challenges that you encountered
here might be helpful.
Thanks!
Matt,
That is indeed the post. I could get as far as the second web page screenshot
and nothing past that login/connect screen... and no meaningful logs on the
asterisk instance.
I tried serving the client via the Apache instance on the server (2.2 and 2.4)
and from the asterisk built in http(s) server.
I'm basically a hobbyist and occasional contractor. The paying job beckoned,
so...
I'd REALLY like to get it working. And for the record, I REALLY HATE pjsip.
I've been twiddling Asterisk (and other VOIP systems) since 2002; Linux since '93 and telecom since 1980. The config is so opaque, poorly documented and error prone I, to this
day, use the legacy sip config wherever I can. No one has ever been able to show me an advantage for it and it doesn't seem to use realtime configuration (even more of a
drawback). I much prefer realtime for my configuration on Asterisk; Having configuration picked up from a DB is far preferable to reloading flat files.
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