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

[James Mortensen]

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?

[image: Happy] <https://secure.teamhively.com/welcome/rate/735/3>[image:
Satisfied] <https://secure.teamhively.com/welcome/rate/735/2>[image:
Unhappy] <https://secure.teamhively.com/welcome/rate/735/1>

Click on a face to provide feedback on my
performance!<https://secure.teamhively.com/welcome/rate/735>

Synclio Extension:*  110627*
Phone: * 866-707-4590*
Skype: *No!* *Call me on Synclio! :)*


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?
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130911/b0848d83/attachment-0001.html