Thank you much for yor reply. 2016-02-18 13:30 GMT+01:00 Simon Hohberg <simon.hohb...@mcs-datalabs.com>:
> Hi Oliver, > > On 02/18/2016 12:10 PM, Olivier wrote: > > Hello, > > I'm trying to have my first calls with WebRTC. > My server has asterisk 13.7.0. > > I'm following the instructions from the wiki [1]. > So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie > station. > > Whenever I type something like ws://123.123.123.123:8088/ws in Expert > Mode form (see [1]), I'm getting this error : > *2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket > connection may not be initiated from a page loaded over HTTPS.* > If I replace ws://123.123.123.123:8088/ws with wss:// > 123.123.123.123:8088/ws, this error message becomes with > *Disconnected: Failed to connet to the server* > > My questions are: > 1. Is wss now required by sipml5 live demo (implying wiki page is not > up-to-date) ? > > Yes, like the error says, you have to use wss on pages served via https. > Furthermore, Chrome requires the use of https when you want to use > getUserMedia. > See here: > https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en. > It says: " Starting with Chrome 47, getUserMedia() requests are only > allowed from secure origins: HTTPS or localhost." > Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ? Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ? If I'm not mistaken, this should fulfill all requirements. > > The solution for development is, to host the webrtc client locally, so > that you load the page from localhost. In that case getUserMedia is allowed > with http, too (as the quote says). That means you have to download the > dubango client and run a webserver on your dev machine. > > 2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ? > > Unfortunately, there is not much documentation about this, as far as I can > tell. > I didn't find any. > > > Regards > > [1] > https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 > [2] https://www.doubango.org/sipml5/ > > > > > Regards, > > Simon > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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