PhoneGap/Cordova 3 Plugin for PJSIP

This project received 17 bids from talented freelancers with an average bid price of €738 EUR.

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Project Budget
€30 - €250 EUR
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Project Description

I need a developer that can build a plugin to make PJSIP useable in PhoneGap/Cordova 3.x on Android and iOS.

PJSip ([url removed, login to view]) is available for Android and iOS.


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 ([url removed, login to view]) plugin will be useable with Cordova 3.x ([url removed, login to view]) 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)[url removed, login to view]"
- 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 [url removed, login to view]() if debug has been enabled

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