2016-02-19 12:01 GMT+01:00 Marek Červenka <[email protected]>: > on my own server >
Today, I'm back from holidays trip. First of all, thanks for replying ! I'll try to use jssip as you suggested. Anyway, I'm still failing to understand if wiki's page [1] is still valid with Asterisk 13, and if it's not valid anymore, which is the main change that prevent things to work. [1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 > > i want try jssip > https://github.com/versatica/JsSIP > it looks like a lot "livelier" than sipml5 > > any experience with jssip? > > > Dne 18.2.2016 v 16:01 Olivier napsal(a): > > > > 2016-02-18 15:42 GMT+01:00 Marek Červenka <[email protected]>: > >> 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 < >> <[email protected]>[email protected]>: >> >>> >>> 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 >
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