= webrtc2sip =

UPDATE: As of May 30, 2014 (with the 1.4.4 version of Freeswitch) we are no longer using webrtc2sip because freeswitch now offers native webrtc support.

== Overview ==

The [http://www.webrtc2sip.org/ webrtc2sip project] provides server components necessary to work with [https://code.google.com/p/sipml5/ sipml5] (the html5 sip library). sipml5 allows us to have a fully functional SIP client built in a web page using html5. We use that library to provide our [wiki:freeswitch-mexcla-html5-admin mexcla interpretation conference call system].

In particular, we depend on it to provide:

 * [https://code.google.com/p/webrtc2sip/#SIP_Proxy sip proxy] which converts communication from [https://en.wikipedia.org/wiki/Websocket Websocket] (which is how all html5 audio/video apps communicate) into [https://en.wikipedia.org/wiki/Session_Initiation_Protocol SIP] (which is the protocol used by our [wiki:freeswitch freeswitch] server, that handles the conference call).
 * [https://code.google.com/p/webrtc2sip/#RTCWeb_Breaker RTC Web breaker] which provides server side support for [https://en.wikipedia.org/wiki/Interactive_Connectivity_Establishment ICE] (the way clients get around NAT) and DTLS/SRTP (secure real time protocol and secure UDP). These protocols are mandatory in html5 applications, but are not supported natively by freeswitch, so webrtc2sip handles that part of the communication.
 * [https://code.google.com/p/webrtc2sip/#Media_Coder Media coder] which provides codec support and conversion for opus and g.711 (freeswitch actually supports both codecs so we don't really need this one).

== Installation ==

webrtc2sip is not [http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=711863 yet] in Debian.

The following packages are not required for building webrtc2sip, but is required to properly run a client:

{{{
apt-get install  resiprocate-turn-server
}}}

To install the components:

 * Ensure you are running wheezy (squeeze would probably work but I didn't try)

 * Install debian dependencies. As root:
{{{
apt-get install make libtool autoconf subversion git wget libogg-dev gcc gobjc++  libsrtp0-dev libsrtp0 srtp-utils openssl libspeex-dev libspeexdsp-dev libspeex1 libspeexdsp1 yasm libvpx-dev libvpx1 libopus-dev libopus0 libopencore-amrnb-dev libopencore-amrwb-dev libopencore-amrwb0 libopencore-amrnb0 libgsmme-dev libgsmme1c2a libgsm1-dev libgsm1 libgsm0710-dev libgsm0710-0 libgsm0710mux-dev libxml2-dev libxml2 libopal3.10.4 libavcodec-dev
}}}

Note: we are not supporting codecs/libraries not shipped with debian (libyuv, g.729, ilbc, h264, h263, MP4V-ES)

 * As root, create the target directory and chown to your non-priv user:
{{{
mkdir /usr/local/src/doubango
chown <user>:<user> /usr/local/src/doubango
}}}
 * As non-priv user, Checkout the framework source code:
{{{
cd /usr/local/src/
svn checkout http://doubango.googlecode.com/svn/branches/2.0/doubango doubango
}}}
 * As non-priv user, build the framework source code:
{{{
cd doubango
./autogen.sh && ./configure --with-ssl --with-srtp
make
}}}
 * As root, install (will install in /usr/local) 
{{{
make install
ldconfig
}}}
 * As root user, create the target directories:
{{{
mkdir /usr/local/src/webrtc2sip
chown <user>:<user> /usr/local/src/webrtc2sip
}}}
 * As non-priv user, checkout the webrtc2sip code:
{{{
cd /usr/local/src/
svn checkout http://webrtc2sip.googlecode.com/svn/trunk webrtc2sip
}}}
 * As non-priv user, build: 
{{{
cd webrtc2sip
./autogen.sh && ./configure --with-doubango=/usr/local --prefix=/usr/local
make clean && make
}}}
 * As root, install:
{{{
make install
}}}
 * As root, copy config file to /etc/ and create symlink:
{{{
mkdir /etc/webrtc2sip
cp -f ./config.xml /etc/webrtc2sip
}}}
 * To run:
{{{
export LD_LIBRARY_PATH=/usr/local/lib
/usr/local/sbin/webrtc2sip --config=/etc/webrtc2sip/config.xml
}}}
 * To run in a more permanent way... I've setup it up to be managed by runit via /etc/sv/webrtc2sip.