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

Anthony Minessale anthony.minessale at gmail.com
Wed Sep 11 22:07:45 MSD 2013


Did you open all the necessary firewall ports?
Playing around on amazon as your first try complicates thing a bit for you.

You should be able to verify its listening on the port with netstat -an |
grep 5066



On Wed, Sep 11, 2013 at 12:55 PM, James Mortensen <
james.mortensen at synclio.com> wrote:

> Here's another update to my adventures in Freeswitch WebRTC.  I assume
> from the getting started documentation that there are users created by
> default with the password 1234, so I'm trying to create the ws 5066
> connection from the TryIt JsSIP demo:  http://tryit.jssip.net
>
> Name: James
> SIP URI: 1000 at Y.Y.Y.Y   <--- Local IP of EC2 server
> SIP password: 1234
> WS URI:  ws://X.X.X.X:5066   <--- Public IP of EC2 server
>
> Hope this helps!
> James
>
>
> On Wed, Sep 11, 2013 at 10:31 AM, James Mortensen <
> james.mortensen at synclio.com> wrote:
>
>> 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?
>>>>
>>>> [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
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>>>>>
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>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> 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
>>>>
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>>>>
>>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com
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>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
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>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _________________________________________________________________________
>>> 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
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>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _________________________________________________________________________
> 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
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> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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