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@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@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
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On Wed, Sep 11, 2013 at 6:13 AM, Brian West <brian@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@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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