[asterisk-users] Wanted: WebRTC tutorial

[Bruce Ferrell]

On 04/24/2018 09:08 AM, Matt Fredrickson wrote:
> On Tue, Apr 24, 2018 at 10:54 AM, Bruce Ferrell <bferrell at baywinds.org> 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.