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?
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! -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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.
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! >OK, I've gone back and refreshed myself;? When I try to access cybermega in /var/lib/asterisk/static-http at port 8088 the asterisk debug shows: [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP opening session.? Top level [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP Request URI is /cyber/index.html [Apr 24 20:34:48] DEBUG[17170] http.c: Requested URI [/cyber/index.html] has no handler [Apr 24 20:34:48] DEBUG[17170] http.c: HTTP keeping session open. status_code:404 When I serve it from apache, the web ui appears, but never connects. Using the firefox dev tools/console I see firefox can't establish a connection the server at wss://<IP address>:8089/ws The asterisk debug log shows: [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP opening session.? Top level [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP Request URI is /ws [Apr 24 20:39:21] DEBUG[19041] http.c: Requested URI [/ws] has no handler [Apr 24 20:39:21] DEBUG[19041] http.c: HTTP keeping session open. status_code:404 Suggestions?