[Freeswitch-users] Configuring Freeswitch 1.4b for WebRTC Peer to Peer and to the PSTN

[James Mortensen]

Hi Anthony,

I have a Chrome client behind a NAT'd network, and I have a Freeswitch 1.4b
instance sitting behind a NAT'd EC2 server.  I enabled ws-binding in
internal.xml. It's the only area where I see anything related to
Websockets.

I also verified my firewall is opened on port 5066. I can see requests
coming in from Chrome when I try to register, but Freeswitch's logs don't
even twitch.  I followed the instructions here to enable DEBUG logging:
http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch

I've also grepped the configuration files and internal.xml is the only
location of "websocket", other than the htdocs.... which seem to relate to
getting a GUI running.

Perhaps there's something I'm misunderstanding about logging in Freeswitch?
 I verified from the console that debug logging is enabled.

Hope this helps and thank you!
James



On Wed, Sep 11, 2013 at 9:15 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Are you testing by connecting to internal or external IP?
> Uncommenting the param is really the only thing you have to do.
>
>
>
>
> On Wed, Sep 11, 2013 at 10:28 AM, James Mortensen <
> james.mortensen at synclio.com> wrote:
>
>> Hi Brian,
>>
>> I didn't stray too far from the default configuration.  I just
>> uncommented the param and left the default port in place, 5066.  Is there a
>> way to tell if the WS/HTTP server is running from within Freeswitch?  I
>> didn't see a command for that.
>>
>> I also replaced the ext-sip-ip and ext-rtp-ip with my stun server and my
>> server's public IP in internal.xml:
>>
>>     <param name="ext-rtp-ip" value="stun:stun.l.google.com:19302"/>
>>     <param name="ext-sip-ip" value="MY_PUBLIC_IP" />
>>
>> The server starts up, but doesn't accept websocket connections.  What
>> information can I get you so this is more helpful?
>>
>> Thank you!
>> James
>>
>>
>> James
>>
>> *"Every Call, Every Time!"*
>>
>> How did I do?
>>
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>>
>>
>> On Wed, Sep 11, 2013 at 6:13 AM, Brian West <brian at freeswitch.org> wrote:
>>
>>> What exactly did you put into the ws-binding param?
>>>
>>> /b
>>>
>>>
>>> On Sep 10, 2013, at 8:48 PM, James Mortensen <
>>> james.mortensen at synclio.com> wrote:
>>>
>>> > Hello,
>>> >
>>> > I'm working on a one-way audio bug that may or may not be either
>>> Asterisk or Chrome WebRTC related. Details are here:
>>> https://code.google.com/p/webrtc/issues/detail?id=2347, but basically
>>> the problem is that I get about 1 in 10 calls returning one way audio where
>>> audio flows from the PSTN to Chrome but not the other way.
>>> >
>>> > I believe the issue may be with Asterisk 11, based on Wireshark traces
>>> and other troubleshooting, and if that's the case, I'm happy to try
>>> Freeswitch as an alternative.
>>> > I tried the demo here:
>>> https://webrtc.freeswitch.org/webrtc/portal.html, and the 20 test calls
>>> to the PSTN worked perfect with two way audio.
>>> >
>>> > So, now, onto the Freeswitch question:  Where is all the documentation
>>> for configuring the Websocket server on Freeswitch 1.4b?  Where is the
>>> documentation that explains how to get ICE enabled?  Why, when I enabled
>>> the ws-bind port in internal.xml does the client fail to establish a WS
>>> connection?
>>> >
>>> > Basically, I'd love to get my hands on the docs for setting this up,
>>> but the Internet is largely silent on this matter?  Is it possible for the
>>> Freeswitch team to share the configuration they used to get that
>>> JsSIP/Freeswitch demo setup?
>>> >
>>> > If not the configuration, where would I find the docs for Freeswitch
>>> 1.4b?
>>> >
>>> > Thank you!!
>>> >
>>> > James
>>> >
>>> > "Every Call, Every Time!"
>>> >
>>> > How did I do?
>>> >
>>>
>>> _________________________________________________________________________
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>>>
>>> 
>>> 
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>>
>> _________________________________________________________________________
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>>
>> 
>> 
>>
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>>
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>>
>
>
> --
> Anthony Minessale II
>
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>
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