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
<http://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
<http://123.123.123.123:8088/ws> with wss://123.123.123.123:8088/ws
<http://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."
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.
Regards
[1]
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
[2] https://www.doubango.org/sipml5/
Regards,
Simon
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