Finally back to blog after a longtime. Continuing from the FreeSWITCH stuff. I learned about the RTMP module which enables us to connect to the FreeSWITCH server via web browser and make calls just like any other Soft-phone.
This is wonderful, I don't need to install any soft-phone or anything, just open up my FS server RTMP phone page - login using my SIP credentials and once authenticated make calls !! simple.
In order to get it working I went back to the source directory of Freeswitch and just compiled thertmp module. Learn more about the FreeSWITCH RTMP Module from this link.
root@FS_HA1#cd /usr/src/freeswitch
root@FS_HA1#make mod_rtmp && make mod_rtmp install
once compiled and installed successfully, enable the rtmp module to be loaded automatically by freeswitch.
root@FS_HA1#cd /usr/local/freeswitch/conf/autoload_configs/
root@FS_HA1#vim modules.conf.xml
Uncomment by removing the "<!--" and "-->"
<load module="mod_rtmp"/>
Save and exit from the file after uncommenting.
Goto FreeSwitch console.
root@FS_HA1#fs_cli
Since my freeswitch was already started so to load RTMP module either I need to restart my Freeswitch server or I can just load the module from the console just like this.freeswitch@internal> load mod_rtmp
It will print lots of output lines on console. Once loaded successfully edit the module settings to your choice.
root@FS_HA1#cd /usr/local/freeswitch/conf/autoload_configs/
root@FS_HA1#vim rtmp.conf.xml
I didn't need to change anything so I just simply reviewed the rtmp.conf.xml file and exited.
Now, time to copy the web-phone from the source directory to the web root.
root@FS_HA1#cd /usr/src/freeswitch/clients/
root@FS_HA1#cp -rf flex/ /var/www/html/
Now we need to edit the html file and change one small thing there.
root@FS_HA1#vim /var/www/html/flex/freeswitch.html
We need to change the rtmp url to that of our server's public IP.
var flashvars = { rtmp_url: 'rtmp://193.195.196.193/phone' };
Save and exit, now open your browser and type down this URL.http://193.195.196.193/flex/freeswitch.html
You;ll see a sample dialer. Login using your SIP credentials and if asked for allowing the browser to use your webcam/microphone/speaker accept it.
After successful login you should be able to make calls given the context your user is assigned to is configured likewise.
Thanks. This is very cool!
ReplyDeleteCan I call from SIP to "web-phone" on page?
Well Yeah. But I will suggest you to look into WebRTC for this purpose now.
DeleteWebRTC isn't ready yet. Its partially supported in Chrome/FF...why do you think hangouts still uses a plugin?
Delete