asterisk users - Feb 2016 - 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
>
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
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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 ?ervenka <cervajs at fpf.slu.cz 
> <mailto: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
>>     <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
======================================
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2016-02-19 12:01 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>:
> 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 ?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>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
>
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