PhoneGap/Cordova 3 Plugin for PJSIP

I need a developer that can build a plugin to make PJSIP useable in PhoneGap/Cordova 3.x on Android and iOS. PJSip ([login to view URL]) is available for Android and iOS. Requirements: Javascript interface to PJSIP in PhoneGap/Cordova to realize the following app: - register with existing JSON/plain API with given token, session id is returned, intialization only possible if login was successful) - register with existing sip server - make calls - receive calls - send dtmf signals - optionally: make/receive video calls I'll provide you with free minutes, required api token and sip credentials during project phase. I'm going to build a cordova app for Android and iOS that should have the ability to make and receive SIP calls, interface design/implementation is done by me, you provide the javascript interface. More detailed specs: - PJSIP 2.1 ([login to view URL]) plugin will be useable with Cordova 3.x ([login to view URL]) on Android platform - all codecs that are provided by PJSIP have to be useable (incl. Intel IPP: G723.1, G726, G728, G729A) - Short description of JavaScript-Interface CordovaPJSIP: - register to VoIP service, username, password and registrar address will be provided - codec preferences have to be available, i.e. define that GSM has higher priority than uLaw or G729 - determine current registration status (registered, registering, unregistered) - JavaScript-Function to establish a new call, destination address will be provided as sip:username(AT)[login to view URL]" - JavaScript-Function to send DTMF tones (0-9, *, #), mute/unmute a call, activate/deactivate speaker, hold/unhold the call, volume control - JavaScript-Function to hangup the call - JavaScript-Function to enable/disable video, camera has to be selectable - certain events have to be generated: - incoming call, call details have to be provided - call end/hangup, incl. call details, release cause - when call has been muted/unmuted, speaker enabled/disabled, hold/unhold, volume change - debug messages / sip messages should be sent to [login to view URL]() if debug has been enabled