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2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > > Is it implied here that both HTTPS and WSS must also come from the same >> server (Same Origin Policy) ? >> > No, the same origin policy does not apply to web sockets. > > Then, can I also install my own WebRTC demo page on...

...t; Yes I saw this post earlier today. Having to fight 14 days scared me a bit ! Did you set sipml5 on your own server or did you use Live demo ( https://www.doubango.org/sipml5/call.htm?svn=241) ? > Dne 18.2.2016 v 15:36 Olivier napsal(a): > > > > 2016-02-18 14:57 GMT+01:00 Simon Hohberg <simon.hohberg at mcs-datalabs.com>: > >> >> Is it implied here that both HTTPS and WSS must also come from the same >>> server (Same Origin Policy) ? >>> >> No, the same origin policy does not apply to web sockets. >> >> Then, can I also ins...

Thank you much for yor reply. 2016-02-18 13:30 GMT+01:00 Simon Hohberg <simon.hohberg at 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...

...o fight 14 days scared me a bit ! > > Did you set sipml5 on your own server or did you use Live demo ( > https://www.doubango.org/sipml5/call.htm?svn=241) ? > > > >> Dne 18.2.2016 v 15:36 Olivier napsal(a): >> >> >> >> 2016-02-18 14:57 GMT+01:00 Simon Hohberg < >> <simon.hohberg at mcs-datalabs.com>simon.hohberg at mcs-datalabs.com>: >> >>> >>> Is it implied here that both HTTPS and WSS must also come from the same >>>> server (Same Origin Policy) ? >>>> >>> No, the same origin p...

Hi, I want to pass a part of a SIP INVITE multipart body. I found a quite old patch here: https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22 But this patch is for the SIP channel driver not PJSIP, right? Is it even possible without a patch? What do I have to put in the dialplan then? Thanks in advance, Simon

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