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Configure FreeSWITCH
SIP.js has been tested with FreeSWITCH 1.5.14 without any modification to the source code of SIP.js or FreeSWITCH. Later versions of FreeSWITCH will require similar configuration.
System Setup
FreeSWITCH and SIP.js were tested using the following setup:
CentOS 6.6 minimal (x86_64)
FreeSWITCH 1.5.14
A public IP address to avoid NAT scenarios on the server side. Required Packages
Install the following dependencies:
git
autoconf
automake
libtool
gcc-c++
libuuid-devel
zlib-devel
libjpeg-devel
ncurses-devel
openssl-devel
Using YUM, all dependencies can be installed with:
yum install git autoconf automake libtool gcc-c++ libuuid-devel zlib-devel libjpeg-devel ncurses-devel openssl-devel
Install FreeSWITCH
FreeSWITCH recommends using the latest version of FreeSWITCH from the FreeSWITCH git repo. This example uses FreeSWITCH tag v1.5.14.
```
cd /usr/local/src/
git clone https://freeswitch.org/stash/scm/fs/freeswitch.git
cd /usr/local/src/freeswitch
git checkout v1.5.14
./bootstrap.sh
./configure
make (This may take a few minutes.)
make install
```
Configure FreeSWITCH
The default configuration files for FreeSWITCH are located in /usr/local/freeswitch/conf.
Start by editing the internal SIP profile sip_profiles/internal.xml. Uncomment the line to allow web sockets to talk to FreeSWITCH. No other configuration changes are necessary to make FreeSWITCH work with WebRTC.
```
<!--internal.xml-->
<!-- Uncomment the following: -->
<param name="ws-binding" value=":5066"/>
```
If you’d like to enable video as well as audio, adjust FreeSWITCH’s codec preferences to include VP8.
<!--vars.xml-->
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,VP8">
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,VP8">
```
Start FreeSWITCH: /usr/local/freeswitch/bin/freeswitch.
Configure SIP.js
SIP.js works with FreeSWITCH without any special configuration parameters. The following UA is configured to connect to a default FreeSWITCH configuration. Replace 127.0.0.1 with the IP address of your FreeSWITCH server.
```
var config = {
// Replace this IP address with your FreeSWITCH IP address
uri: '1000@127.0.0.1',
// Replace this IP address with your FreeSWITCH IP address
// and replace the port with your FreeSWITCH port
ws_servers: 'ws://127.0.0.1:5066',
// FreeSWITCH Default Username
authorizationUser: '1000',
// FreeSWITCH Default Password
password: '1234'
};
var ua = new SIP.UA(config);
```
Troubleshooting
It is known that SIP.js and FreeSWITCH might not interop well if you have the following option enabled on FreeSWITCH:
```
<variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/>
```
Firefox 34+ requires SIP.js 0.6.4 or later to interop with FreeSWITCH or Asterisk.
FreeSWITCH has a wiki article on WebRTC support.