asterisk users - Aug 2016 - Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

My main reason not to upgrade to Ast 13 is because I'm afraid of losing 
functionality as there are certain functions deprecated/replaced. This 
can also cause headache :-)

I will do so if there is no other option.

But still, I don't see why Ast 13 would differ so much in this case ? If 
ICE and NAT is working (not causing problems) why should Ast 13 bring me 
audio and Ast 12 don't ??



I indeed use SIPML5 demo as quick test-case. So do many tutorials on the 
web.

Self-signed certificates should be OK as long as they are imported in 
the browser. Never knew this could cause audio problems ?




Kind regards.



On 11-08-16 16:25, Jonathan H wrote:
> I'm genuinely fascinated why you are insisting on using a version of 
> Asterisk almost 3 years old, for which EOL support ended last year.
>
> Is there any particular reason you cannot or will not use the current 
> version as others have suggested?
>
> Also, I see you are using Doubango and WebRTC, but in the logs, I see 
> WS and WSS.
>
> You NEED to be using 100% WSS otherwise you've not got a hope in hell 
> of anything working with WEBRTC.
> Check the console of the web browser you are trying to make the call 
> from (CTRL-SHIFT-I in Chrome on Windows, for example).
>
> Also, you'll need to be using valid certificates - self-signed 
> certificates won't work for any current implementation of WebRTC that 
> I know of, certainly not if anything involves current versions of 
> Chrome or Firefox. That said, LetsEncrypt certs work fine for this, so 
> no need to spend out on one.
>
> Switch to Asterisk 13.10 and save yourself a whole lotta headache.
>
> On 11 August 2016 at 15:09, Jonas Kellens <jonas.kellens at telenet.be 
> <mailto:jonas.kellens at telenet.be>> wrote:
>
>     Hello
>
>     Using Asterisk 12.8.2.
>
>
>     On 10-08-16 22:03, Matt Fredrickson wrote:
>
>         My suggestion is to verify and debug against Asterisk 13
>         first, and
>         then you can try backing down versions, rather than reverse. 
>         WebRTC
>         is a rapidly moving target, and has required ongoing changes
>         that may
>         not have made it into older and feature frozen versions of
>         Asterisk.
>
>
>
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On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at
telenet.be> wrote:
> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
> functionality as there are certain functions deprecated/replaced. This can
> also cause headache :-)
>
> I will do so if there is no other option.
>
> But still, I don't see why Ast 13 would differ so much in this case ?
If ICE
> and NAT is working (not causing problems) why should Ast 13 bring me audio
> and Ast 12 don't ??
If you want to minimize grief, start with 13 - WebRTC has been a
moving target for the last 5 years, it is not an old, mature standard
like ISDN or SIP.  If you find interop problems in an older version of
Asterisk with WebRTC, it's likely that it has been fixed in 13, and if
it hasn't the most likely place to obtain the fix will be in 13.

After you get the WebRTC part working, then you can move back the
versions of Asterisk you're using to see if it still works.

As far as ICE not working goes, if the browser you're talking to is
not on the same network as the Asterisk server, it's *possible* you
might need a true TURN server as well, instead of just an ICE server.

Matthew Fredrickson
>
>
>
> On 11-08-16 16:25, Jonathan H wrote:
>
> I'm genuinely fascinated why you are insisting on using a version of
> Asterisk almost 3 years old, for which EOL support ended last year.
>
> Is there any particular reason you cannot or will not use the current
> version as others have suggested?
>
> Also, I see you are using Doubango and WebRTC, but in the logs, I see WS
and
> WSS.
>
> You NEED to be using 100% WSS otherwise you've not got a hope in hell
of
> anything working with WEBRTC.
> Check the console of the web browser you are trying to make the call from
> (CTRL-SHIFT-I in Chrome on Windows, for example).
>
> Also, you'll need to be using valid certificates - self-signed
certificates
> won't work for any current implementation of WebRTC that I know of,
> certainly not if anything involves current versions of Chrome or Firefox.
> That said, LetsEncrypt certs work fine for this, so no need to spend out on
> one.
>
> Switch to Asterisk 13.10 and save yourself a whole lotta headache.
>
> On 11 August 2016 at 15:09, Jonas Kellens <jonas.kellens at
telenet.be> wrote:
>>
>> Hello
>>
>> Using Asterisk 12.8.2.
>>
>
>
>>
>> On 10-08-16 22:03, Matt Fredrickson wrote:
>>>
>>> My suggestion is to verify and debug against Asterisk 13 first, and
>>> then you can try backing down versions, rather than reverse. 
WebRTC
>>> is a rapidly moving target, and has required ongoing changes that
may
>>> not have made it into older and feature frozen versions of
Asterisk.
>
>
>
>
>
>
> --
> _____________________________________________________________________
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-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at
telenet.be> wrote:
> My main reason not to upgrade to Ast 13 is because I'm afraid of losing
> functionality as there are certain functions deprecated/replaced. This can
> also cause headache :-)
What in particular?

Any longer, Asterisk is *very* conservative with functionality that is
removed. Given that Asterisk 13 is simply the evolution and refinement
of the architecture introduced in Asterisk 12, I would not expect
there to be any major differences moving from 12 to 13.
> I will do so if there is no other option.
>
> But still, I don't see why Ast 13 would differ so much in this case ?
If ICE
> and NAT is working (not causing problems) why should Ast 13 bring me audio
> and Ast 12 don't ??
Asterisk 13 has a lot more bug fixes than Asterisk 12. Asterisk 12 is
no longer actively supported.

Supported timelines for versions are available on the wiki:

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

-- 
Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

On 11-08-16 18:03, Matt Fredrickson wrote:
> On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens <jonas.kellens at
telenet.be> wrote:
>> My main reason not to upgrade to Ast 13 is because I'm afraid of
losing
>> functionality as there are certain functions deprecated/replaced. This
can
>> also cause headache :-)
>>
>> I will do so if there is no other option.
>>
>> But still, I don't see why Ast 13 would differ so much in this case
? If ICE
>> and NAT is working (not causing problems) why should Ast 13 bring me
audio
>> and Ast 12 don't ??
> If you want to minimize grief, start with 13 - WebRTC has been a
> moving target for the last 5 years, it is not an old, mature standard
> like ISDN or SIP.  If you find interop problems in an older version of
> Asterisk with WebRTC, it's likely that it has been fixed in 13, and if
> it hasn't the most likely place to obtain the fix will be in 13.
>
> After you get the WebRTC part working, then you can move back the
> versions of Asterisk you're using to see if it still works.
>
> As far as ICE not working goes, if the browser you're talking to is
> not on the same network as the Asterisk server, it's *possible* you
> might need a true TURN server as well, instead of just an ICE server.
>
> Matthew Fredrickson
>
Matthew

when I set the following in rtp.conf :

turnaddr=192.158.29.39:3478?transport=udp
turnusername=28224511:1379330808
turnpassword=JZEOEt2V3Qb0y27GRntt2u2PAYA


then Asterisk 12 gets really slow and sometimes unresponsive. Calls 
result in 480 request timeout (possibly due to the freeze of Asterisk).

So this is also no solution.

Can not even test if it brings me some audio in my webRTC calls.


(putting the above lines back in comment resolves the issue of Asterisk 
freeze. This is all EXTREMELY BUGGY !)


Asterisk 13 here I come (with very high expectations).


Kind regards.