asterisk users - Mar 2016 - Asterisk 13 and WebRTC. Is wiki page still valid ?

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 --
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Olivier wrote:
>
>
> 2016-02-19 12:01 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz
> <mailto: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.
If Chrome or the other browsers have changed things (or implemented new 
requirements, ala HTTPS for serving stuff up) then it may not be correct 
anymore. Chasing WebRTC is not currently something we currently do due 
to the resources involved, but if the community can provide any changes 
to the wiki page to help make it clearer or valid again they can be left 
as a comment and we can incorporate them. If code changes are required 
we do of course encourage those to be contributed[1].

Cheers,

[1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

I'm discovering WebRTC and I think it's a technology that is quite
difficult to integrate with so many changing interfaces.

I think this is typically the kind of subject where the community could
positively contribute to keep wiki pages updated.

As I'm quite interested in this topic, I'm assigning myself this task
for
the next weeks.


2016-03-02 12:40 GMT+01:00 Joshua Colp <jcolp at digium.com>:
> Olivier wrote:
>
>>
>>
>> 2016-02-19 12:01 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz
>> <mailto: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.
>>
>
> If Chrome or the other browsers have changed things (or implemented new
> requirements, ala HTTPS for serving stuff up) then it may not be correct
> anymore. Chasing WebRTC is not currently something we currently do due to
> the resources involved, but if the community can provide any changes to the
> wiki page to help make it clearer or valid again they can be left as a
> comment and we can incorporate them. If code changes are required we do of
> course encourage those to be contributed[1].
>
> Cheers,
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
>
>
> --
> _____________________________________________________________________
> -- 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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