Olivier
2016-Feb-18 14:36 UTC
[asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?
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 my own private >> Asterisk server and access this demo page through HTTPS ? >> If I'm not mistaken, this should fulfill all requirements. >> > It doesn't matter where the asterisk server is hosted. It is important > where the web application comes from. If you don't want to use https and > wss you only have the option to host the web app locally (on the same > machine as the browser that loads the page), which probably makes sense > only for development. Otherwise you have to use https and wss for the > reasons discussed earlier. > > Hope it helps.At least, it helped me to realize I still have several more things to learn ;-) My setup is the following: - an asterisk server, - a PC, - asterisk server and PC are installed on the same LAN - sipM5 live demo outside my LAN - no NAT/PAT configuration allowing incoming communications from the outside. Is using sipML live demo as a way to rapidly test private Asterisk WebRTC capabilies, something achievable ? What would keep this from working ?> > > > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160218/45d20774/attachment.html>
Marek Červenka
2016-Feb-18 14:42 UTC
[asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?
my experience with pjsip for webrtc http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html 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 <mailto: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 my own > private Asterisk server and access this demo page through HTTPS ? > If I'm not mistaken, this should fulfill all requirements. > > It doesn't matter where the asterisk server is hosted. It is > important where the web application comes from. If you don't want > to use https and wss you only have the option to host the web app > locally (on the same machine as the browser that loads the page), > which probably makes sense only for development. Otherwise you > have to use https and wss for the reasons discussed earlier. > > Hope it helps. > > > > At least, it helped me to realize I still have several more things to > learn ;-) > > My setup is the following: > - an asterisk server, > - a PC, > - asterisk server and PC are installed on the same LAN > - sipM5 live demo outside my LAN > - no NAT/PAT configuration allowing incoming communications from the > outside. > > Is using sipML live demo as a way to rapidly test private Asterisk > WebRTC capabilies, something achievable ? > What would keep this from working ? > >-- --------------------------------------- Marek Cervenka ====================================== -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160218/559cb918/attachment.html>
Olivier
2016-Feb-18 15:01 UTC
[asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?
2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>:> my experience with pjsip for webrtc > http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html > > > 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 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. >>> >> It doesn't matter where the asterisk server is hosted. It is important >> where the web application comes from. If you don't want to use https and >> wss you only have the option to host the web app locally (on the same >> machine as the browser that loads the page), which probably makes sense >> only for development. Otherwise you have to use https and wss for the >> reasons discussed earlier. >> >> Hope it helps. > > > > At least, it helped me to realize I still have several more things to > learn ;-) > > My setup is the following: > - an asterisk server, > - a PC, > - asterisk server and PC are installed on the same LAN > - sipM5 live demo outside my LAN > - no NAT/PAT configuration allowing incoming communications from the > outside. > > Is using sipML live demo as a way to rapidly test private Asterisk WebRTC > capabilies, something achievable ? > What would keep this from working ? > > > > > -- > --------------------------------------- > Marek Cervenka > ======================================> > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160218/2c9eefb2/attachment.html>