on my own server 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] <mailto:[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] <mailto:[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 =======================================
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