Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-03-02 Thread Olivier 
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 :

> Olivier wrote:
>
>>
>>
>> 2016-02-19 12:01 GMT+01:00 Marek Červenka > >:
>>
>> 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
>
-- 
_
-- 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

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-03-02 Thread Joshua Colp 

Olivier wrote:



2016-02-19 12:01 GMT+01:00 Marek Červenka >:

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

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-29 Thread Olivier 
2016-02-19 12:01 GMT+01:00 Marek Červenka :

> 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 :
>
>> 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.hohb...@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
>
-- 
_
-- 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

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-19 Thread Marek Červenka 

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 >:


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
>:


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

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-18 Thread Olivier 
2016-02-18 15:42 GMT+01:00 Marek Červenka :

> 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 :
>
>>
>> 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
>
-- 
_
-- 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

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-18 Thread Marek Červenka 

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 
>:



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

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-18 Thread Olivier 
2016-02-18 14:57 GMT+01:00 Simon Hohberg :

>
> 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
>
-- 
_
-- 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

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-18 Thread Simon Hohberg 


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.


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

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-18 Thread Olivier 
Thank you much for yor reply.

2016-02-18 13:30 GMT+01:00 Simon Hohberg :

> Hi Oliver,
>
> On 02/18/2016 12:10 PM, Olivier wrote:
>
> Hello,
>
> I'm trying to have my first calls with WebRTC.
> My server has asterisk 13.7.0.
>
> I'm following the instructions from the wiki [1].
> So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie
> station.
>
> Whenever I type something like ws://123.123.123.123:8088/ws in Expert
> Mode form (see [1]), I'm getting this error :
> *2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket
> connection may not be initiated from a page loaded over HTTPS.*
> If I replace ws://123.123.123.123:8088/ws with wss://
> 123.123.123.123:8088/ws, this error message becomes with
> *Disconnected: Failed to connet to the server*
>
> My questions are:
> 1. Is wss now required by sipml5 live demo (implying wiki page is not
> up-to-date) ?
>
> Yes, like the error says, you have to use wss on pages served via https.
> Furthermore, Chrome requires the use of https when you want to use
> getUserMedia.
> See here:
> https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en.
> It says: " Starting with Chrome 47, getUserMedia() requests are only
> allowed from secure origins: HTTPS or localhost."
>

Is it implied here that both HTTPS and WSS must also come from the same
server (Same Origin Policy) ?

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.


>
> The solution for development is, to host the webrtc client locally, so
> that you load the page from localhost. In that case getUserMedia is allowed
> with http, too (as the quote says). That means you have to download the
> dubango client and run a webserver on your dev machine.
>
> 2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ?
>
> Unfortunately, there is not much documentation about this, as far as I can
> tell.
>

I didn't find any.

>
>
> Regards
>
> [1]
> https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
> [2] https://www.doubango.org/sipml5/
>
>
>
>
> Regards,
>
> 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
>
-- 
_
-- 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

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-18 Thread Simon Hohberg 

Hi Oliver,

On 02/18/2016 12:10 PM, Olivier wrote:

Hello,

I'm trying to have my first calls with WebRTC.
My server has asterisk 13.7.0.

I'm following the instructions from the wiki [1].
So I'm using [2] live demo from a Chrome navigator (v48) on Debian 
Jessie station.


Whenever I type something like ws://123.123.123.123:8088/ws 
 in Expert Mode form (see [1]), I'm 
getting this error :
*2:SecurityError: Failed to construct 'WebSocket': An insecure 
WebSocket connection may not be initiated from a page loaded over HTTPS.*
If I replace ws://123.123.123.123:8088/ws 
 with wss://123.123.123.123:8088/ws 
, this error message becomes with

/Disconnected:*Failed to connet to the server*/

My questions are:
1. Is wss now required by sipml5 live demo (implying wiki page is not 
up-to-date) ?
Yes, like the error says, you have to use wss on pages served via https. 
Furthermore, Chrome requires the use of https when you want to use 
getUserMedia.
See here: 
https://developers.google.com/web/updates/2015/10/chrome-47-webrtc?hl=en. It 
says: " Starting with Chrome 47, getUserMedia() requests are only 
allowed from secure origins: HTTPS or localhost."


The solution for development is, to host the webrtc client locally, so 
that you load the page from localhost. In that case getUserMedia is 
allowed with http, too (as the quote says). That means you have to 
download the dubango client and run a webserver on your dev machine.



2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ?
Unfortunately, there is not much documentation about this, as far as I 
can tell.




Regards

[1] 
https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5

[2] https://www.doubango.org/sipml5/





Regards,

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

[asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

2016-02-18 Thread Olivier 
Hello,

I'm trying to have my first calls with WebRTC.
My server has asterisk 13.7.0.

I'm following the instructions from the wiki [1].
So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie
station.

Whenever I type something like ws://123.123.123.123:8088/ws in Expert Mode
form (see [1]), I'm getting this error :
*2:SecurityError: Failed to construct 'WebSocket': An insecure WebSocket
connection may not be initiated from a page loaded over HTTPS.*
If I replace ws://123.123.123.123:8088/ws with wss://123.123.123.123:8088/ws,
this error message becomes with
*Disconnected: Failed to connet to the server*

My questions are:
1. Is wss now required by sipml5 live demo (implying wiki page is not
up-to-date) ?
2. Do you have any pointer for WebRTC with Asterisk 13 and PJSIP ?

Regards

[1] https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5
[2] https://www.doubango.org/sipml5/
-- 
_
-- 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