Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-14 Thread Jonas Kellens 

Hello


I've succeeded in installing Asterisk 13 and more important : I can make 
webRTC call and I have audio !!


For those on the search like myself, I want to spare some weeks of headache.

My steps (CentOS 6.8) :

yum install uuid-devel libuuid-devel autoconf patch automake 
libcurl-devel libogg-devel libvorbis-devel speex-devel popt-devel 
libtool-ltdl-devel libresample-devel gsm-devel libedit-devel 
python-devel jansson-devel binutils-devel


wget http://www.pjsip.org/release/2.5.5/pjproject-2.5.5.tar.bz2
tar -xjvf pjproject-2.5.5.tar.bz2
./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr 
--libdir=/usr/lib64 --enable-shared --disable-video --disable-sound 
--disable-opencore-amr

make dep
make
make install
ldconfig -p | grep pj
ldconfig

wget 
http://downloads.asterisk.org/pub/telephony/certified-asterisk/asterisk-certified-13.8-current.tar.gz
[root@siptest asterisk-certified-13.8-cert1]# ./configure 
--libdir=/usr/lib64

[root@siptest asterisk-certified-13.8-cert1]# make menuselect
[root@siptest asterisk-certified-13.8-cert1]# make && make install


Forget the option "--with-pjproject-bundled" I would say. Did not work 
for me on : CentOS release 6.8 (Final)




Kind regards.


On 12-08-16 17:22, Jonas Kellens wrote:

Hello


running into several problems when installing 
asterisk-certified-13.8-cert1 (more then I ever had in Asterisk 11 and 
12).


I compile : ./configure --libdir=/usr/lib64 --with-pjproject-bundled

First, I do not seem to have res_srtp module available, although all 
necessary libs are present on the system


Second, I am not able to start Asterisk with following error : 
"/usr/sbin/asterisk: error while loading shared libraries: libpj.so.2: 
cannot open shared object file: No such file or directory"





Help appreciated.

Kind regards.




On 12-08-16 16:58, Jonas Kellens wrote:


On 12-08-16 16:38, Joshua Colp wrote:

Jonas Kellens wrote:

Question : I noticed I received an error when installing pjproject
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm 
having ??


WebRTC requires SRTP and Asterisk has to be built with it enabled. 
It's okay if pjproject doesn't as we don't use their media layer. Do 
you have the res_srtp module in Asterisk?




Hello

Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest 
version

Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version

However, I am not able to select res_srtp module in menuselect. It 
says XXX res_srtp module




Kind regards.







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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens 

Hello


running into several problems when installing 
asterisk-certified-13.8-cert1 (more then I ever had in Asterisk 11 and 12).


I compile : ./configure --libdir=/usr/lib64 --with-pjproject-bundled

First, I do not seem to have res_srtp module available, although all 
necessary libs are present on the system


Second, I am not able to start Asterisk with following error : 
"/usr/sbin/asterisk: error while loading shared libraries: libpj.so.2: 
cannot open shared object file: No such file or directory"





Help appreciated.

Kind regards.




On 12-08-16 16:58, Jonas Kellens wrote:


On 12-08-16 16:38, Joshua Colp wrote:

Jonas Kellens wrote:

Question : I noticed I received an error when installing pjproject
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm 
having ??


WebRTC requires SRTP and Asterisk has to be built with it enabled. 
It's okay if pjproject doesn't as we don't use their media layer. Do 
you have the res_srtp module in Asterisk?




Hello

Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest 
version

Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version

However, I am not able to select res_srtp module in menuselect. It 
says XXX res_srtp module




Kind regards.





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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens 


On 12-08-16 16:38, Joshua Colp wrote:

Jonas Kellens wrote:

Question : I noticed I received an error when installing pjproject
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm 
having ??


WebRTC requires SRTP and Asterisk has to be built with it enabled. 
It's okay if pjproject doesn't as we don't use their media layer. Do 
you have the res_srtp module in Asterisk?




Hello

Package libsrtp-devel-1.5.4-3.el6.x86_64 already installed and latest 
version

Package libsrtp-1.5.4-3.el6.x86_64 already installed and latest version

However, I am not able to select res_srtp module in menuselect. It says 
XXX res_srtp module




Kind regards.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Joshua Colp 

Jonas Kellens wrote:

Question : I noticed I received an error when installing pjproject
--with-external-srtp

I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm having ??


WebRTC requires SRTP and Asterisk has to be built with it enabled. It's 
okay if pjproject doesn't as we don't use their media layer. Do you have 
the res_srtp module in Asterisk?


--
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


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens 
Question : I noticed I received an error when installing pjproject 
--with-external-srtp


I do not seems to have the srtp capability.
(However I can easily install with "yum install libsrtp-devel")

Can this have anything to do with the no-audio-problems that I'm having ??



Kind regards.


On 12-08-16 15:02, Jonas Kellens wrote:

Hello


setting "nat=no" or omitting "nat=" in peer definition does not help 
either. Still no audio.


Why do you think this is a NAT issue ? IP and port information in 
SDP-body is correct.





Kind regards.


On 12-08-16 09:25, Антон Сацкий wrote:


Try delete nat from 77wrtc settings ice should do the same


On Aug 11, 2016 10:00 PM, "Jonas Kellens" > wrote:


On 11-08-16 18:03, Matt Fredrickson wrote:

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens
>
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.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Jonas Kellens 

Hello


setting "nat=no" or omitting "nat=" in peer definition does not help 
either. Still no audio.


Why do you think this is a NAT issue ? IP and port information in 
SDP-body is correct.





Kind regards.


On 12-08-16 09:25, Антон Сацкий wrote:


Try delete nat from 77wrtc settings ice should do the same


On Aug 11, 2016 10:00 PM, "Jonas Kellens" > wrote:


On 11-08-16 18:03, Matt Fredrickson wrote:

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens
>
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.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-12 Thread Антон Сацкий 
Try delete nat from 77wrtc settings ice should do the same

On Aug 11, 2016 10:00 PM, "Jonas Kellens"  wrote:

> On 11-08-16 18:03, Matt Fredrickson wrote:
>
>> On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens 
>> 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.
>
>
> --
> _
> -- 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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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-11 Thread Jonas Kellens 

On 11-08-16 18:03, Matt Fredrickson wrote:

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens  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.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-11 Thread Matthew Jordan 
On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens  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

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

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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-11 Thread Matt Fredrickson 
On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens  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  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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> _
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Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-11 Thread Jonathan H 
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  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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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-11 Thread Jonas Kellens 

Hello

Using Asterisk 12.8.2.

I now have the "via ICE" messages in the RTP debug (see below).

If you look in the SIP debug (see below), you also now see the 
"ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from Asterisk to the 
webRTC client.



But still no audio ! None at all ! In both directions.


You can see in the SIP debug that the IP-address in de SDP-body is 
correctly set for sending audio. So I don't think it is a NAT/ICE problem.



Can anyone tell me then what is left that could be causing the 
'no-audio' problem ??




SIP debug :


[Aug 11 15:53:47] <--- SIP read from WS:178.119.146.190:60191 --->
[Aug 11 15:53:47] INVITE sip:419@178.18.90.230 SIP/2.0
[Aug 11 15:53:47] Via: SIP/2.0/WSS 
df7jal23ls0d.invalid;branch=z9hG4bKSqKu6K3uxr3dOFdU5WAtPM5tKKA5yzAq;rport
[Aug 11 15:53:47] From: 
;tag=SGUVL1LMdvxQrUfxprZJ

[Aug 11 15:53:47] To: 
[Aug 11 15:53:47] Contact: 
;+g.oma.sip-im;language="en,fr"

[Aug 11 15:53:47] Call-ID: 47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:47] CSeq: 58874 INVITE
[Aug 11 15:53:47] Content-Type: application/sdp
[Aug 11 15:53:47] Content-Length: 2301
[Aug 11 15:53:47] Max-Forwards: 70
[Aug 11 15:53:47] Authorization: Digest 
username="77wrtc",realm="178.18.90.230",nonce="5d5c700b",uri="sip:419@178.18.90.230",response="ca118222a4674b4c6dcc19dd95e00c15",algorithm=MD5

[Aug 11 15:53:47] User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
[Aug 11 15:53:47] Organization: Doubango Telecom
[Aug 11 15:53:47]
[Aug 11 15:53:47] v=0
[Aug 11 15:53:47] o=- 5876454736929512000 2 IN IP4 127.0.0.1
[Aug 11 15:53:47] s=Doubango Telecom - chrome
[Aug 11 15:53:47] t=0 0
[Aug 11 15:53:47] a=group:BUNDLE audio
[Aug 11 15:53:47] a=msid-semantic: WMS kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka
[Aug 11 15:53:47] m=audio 63897 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 
105 13 126

[Aug 11 15:53:47] c=IN IP4 178.119.146.190
[Aug 11 15:53:47] a=rtcp:63899 IN IP4 178.119.146.190
[Aug 11 15:53:47] a=candidate:2999745851 1 udp 2122260223 192.168.56.1 
63896 typ host generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:3378846520 1 udp 2122194687 192.168.1.120 
63897 typ host generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:2999745851 2 udp 2122260222 192.168.56.1 
63898 typ host generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:3378846520 2 udp 2122194686 192.168.1.120 
63899 typ host generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:1210916236 1 udp 1685987071 
178.119.146.190 63897 typ srflx raddr 192.168.1.120 rport 63897 
generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:1210916236 2 udp 1685987070 
178.119.146.190 63899 typ srflx raddr 192.168.1.120 rport 63899 
generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:4233069003 1 tcp 1518280447 192.168.56.1 9 
typ host tcptype active generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:2280056776 1 tcp 1518214911 192.168.1.120 
9 typ host tcptype active generation 0 network-id 2
[Aug 11 15:53:47] a=candidate:4233069003 2 tcp 1518280446 192.168.56.1 9 
typ host tcptype active generation 0 network-id 1
[Aug 11 15:53:47] a=candidate:2280056776 2 tcp 1518214910 192.168.1.120 
9 typ host tcptype active generation 0 network-id 2

[Aug 11 15:53:47] a=ice-ufrag:TxJQpv1i5O04Q+Kw
[Aug 11 15:53:47] a=ice-pwd:LvfUjrDPbY/np215T3+6Sy03
[Aug 11 15:53:47] a=fingerprint:sha-256 
EF:A4:78:E4:C1:33:5A:F5:36:6B:C5:DF:C7:D9:10:44:FD:96:5D:88:79:AB:8C:A0:E2:71:66:DA:6D:2C:30:84

[Aug 11 15:53:47] a=setup:actpass
[Aug 11 15:53:47] a=mid:audio
[Aug 11 15:53:47] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[Aug 11 15:53:47] a=extmap:3 
http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time

[Aug 11 15:53:47] a=sendrecv
[Aug 11 15:53:47] a=rtcp-mux
[Aug 11 15:53:47] a=rtpmap:111 opus/48000/2
[Aug 11 15:53:47] a=rtcp-fb:111 transport-cc
[Aug 11 15:53:47] a=fmtp:111 minptime=10;useinbandfec=1
[Aug 11 15:53:47] a=rtpmap:103 ISAC/16000
[Aug 11 15:53:47] a=rtpmap:104 ISAC/32000
[Aug 11 15:53:47] a=rtpmap:9 G722/8000
[Aug 11 15:53:47] a=rtpmap:0 PCMU/8000
[Aug 11 15:53:47] a=rtpmap:8 PCMA/8000
[Aug 11 15:53:47] a=rtpmap:106 CN/32000
[Aug 11 15:53:47] a=rtpmap:105 CN/16000
[Aug 11 15:53:47] a=rtpmap:13 CN/8000
[Aug 11 15:53:47] a=rtpmap:126 telephone-event/8000
[Aug 11 15:53:47] a=ssrc:54412034 cname:H2asKiJklFa9L3Xw
[Aug 11 15:53:47] a=ssrc:54412034 
msid:kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka 
f25030f2-3e48-4180-aea4-4edec3e67410
[Aug 11 15:53:47] a=ssrc:54412034 
mslabel:kBwsfCPizGNiVjZS23dGoTNcUubDMMhxXrka

[Aug 11 15:53:47] a=ssrc:54412034 label:f25030f2-3e48-4180-aea4-4edec3e67410
[Aug 11 15:53:47] <->
[Aug 11 15:53:47] --- (13 headers 44 lines) ---
[Aug 11 15:53:47] Using INVITE request as basis request - 
47ca4cc9-9dce-4449-d58f-e069a67061ec
[Aug 11 15:53:47] Found peer '77wrtc' for '77wrtc' from 
178.119.146.190:60191

[Aug 11 15:53:47]   == Using SIP RTP TOS bits

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-11 Thread Jonas Kellens 
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 > 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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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-10 Thread Matt Fredrickson 
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.

Matthew Fredrickson

On Wed, Aug 10, 2016 at 3:01 PM, Jonas Kellens  wrote:
> Hello
>
> thank you for your answer.
>
> I don't understand how there are many tutorials and examples on the web
> where every time the outcome is a working setup. Very strange I feel now
> after my personal experience with Asterisk 11 and webRTC.
>
> You also say Asterisk 13. How about Asterisk 12 then ??
>
>
>
> Kind regards.
>
>
>
> On 10-08-16 21:53, Matt Fredrickson wrote:
>
> I don't see an ice-ufrag or ice-pwd line in the response from
> Asterisk, correlating with your suspicion that there is no ICE.  Are
> you sure that the stun server you're using (the google one) still
> works?  I haven't tried that server in a while, but I distantly seem
> to recall that maybe they shut it down.
>
> Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
> feature updated in a while, and it could be that it could be a number
> of patches/fixes behind with regards to webrtc support, particularly
> with regards to interoperating with a modern browser version.
>
> Hope that helps,
> Matthew Fredrickson
>
> On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens 
> wrote:
>
> On 10-08-16 08:52, Ludovic Gasc wrote:
>
> For WebRTC, I recommend you to use Asterisk 13+.
>
> Have a nice day.
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
>
>
>
> Hello
>
> then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
>
> This is no answer to my question.
>
> So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??
>
>
>
> Kind regards.
>
>
>
> --
> _
> -- 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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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-10 Thread Jonas Kellens 

Hello

thank you for your answer.

I don't understand how there are many tutorials and examples on the web 
where every time the outcome is a working setup. Very strange I feel now 
after my personal experience with Asterisk 11 and webRTC.


You also say Asterisk 13. How about Asterisk 12 then ??



Kind regards.



On 10-08-16 21:53, Matt Fredrickson wrote:

I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE.  Are
you sure that the stun server you're using (the google one) still
works?  I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it down.

Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
feature updated in a while, and it could be that it could be a number
of patches/fixes behind with regards to webrtc support, particularly
with regards to interoperating with a modern browser version.

Hope that helps,
Matthew Fredrickson

On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens  wrote:

On 10-08-16 08:52, Ludovic Gasc wrote:

For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/




Hello

then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??

This is no answer to my question.

So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??



Kind regards.



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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-10 Thread Matt Fredrickson 
I don't see an ice-ufrag or ice-pwd line in the response from
Asterisk, correlating with your suspicion that there is no ICE.  Are
you sure that the stun server you're using (the google one) still
works?  I haven't tried that server in a while, but I distantly seem
to recall that maybe they shut it down.

Asterisk 13 is a better place to be as well.  Asterisk 11 hasn't been
feature updated in a while, and it could be that it could be a number
of patches/fixes behind with regards to webrtc support, particularly
with regards to interoperating with a modern browser version.

Hope that helps,
Matthew Fredrickson

On Wed, Aug 10, 2016 at 5:02 AM, Jonas Kellens  wrote:
>
> On 10-08-16 08:52, Ludovic Gasc wrote:
>
> For WebRTC, I recommend you to use Asterisk 13+.
>
> Have a nice day.
>
> Ludovic Gasc (GMLudo)
> http://www.gmludo.eu/
>
>
>
>
> Hello
>
> then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??
>
> This is no answer to my question.
>
> So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??
>
>
>
> Kind regards.
>
>
>
> --
> _
> -- 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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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-10 Thread Jonas Kellens 


On 10-08-16 08:52, Ludovic Gasc wrote:


For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/





Hello

then why is there an option in sip.conf and rtp.conf " icesupport=yes" ??

This is no answer to my question.

So again : what am I missing to get ICE support on my Asterisk 11.23.0 ??



Kind regards.


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Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

2016-08-10 Thread Ludovic Gasc 
For WebRTC, I recommend you to use Asterisk 13+.

Have a nice day.

Ludovic Gasc (GMLudo)
http://www.gmludo.eu/
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