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https://github.com/ricardojlrufino/webphone-sip

WebRTC SIP based VoIP client software (+chrome extension)
https://github.com/ricardojlrufino/webphone-sip

asterisk freeswitch voip webrtc

Last synced: about 2 months ago
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WebRTC SIP based VoIP client software (+chrome extension)

Host: GitHub
URL: https://github.com/ricardojlrufino/webphone-sip
Owner: ricardojlrufino
Created: 2017-11-09T01:03:48.000Z (about 7 years ago)
Default Branch: master
Last Pushed: 2020-04-10T23:37:16.000Z (over 4 years ago)
Last Synced: 2024-07-31T19:43:11.814Z (5 months ago)
Topics: asterisk, freeswitch, voip, webrtc
Language: JavaScript
Size: 287 KB
Stars: 95
Watchers: 9
Forks: 56
Open Issues: 5
Metadata Files:
- Readme: README.md

Awesome Lists containing this project

README

        # webphone-sip

WebRTC SIP based VoIP client software (+chrome extension)

It allows you to make calls using your browser in an extremely productive way.

![Preview](/docs/preview-1400x560.png?raw=true "Preview")

## Features

* Make and get calls

* Audio effects using JS Audio API (Ex.: DTMF)

* Phone Controls - HOLD / MUTE / STOP

* Visual Effects in Calls (waveform viewer)

* ONLY JAVA-SCRIPT (using SIP.js)

* Chrome Extension for Click-To-CALL

* Internationalization Support

### TODO 

* Call History

* WebPack build

* Receive Calls "in Backgruound"

## Chrome Extension

Chrome Extension allows you to turn phone numbers and link with the extension to make calls quickly (Click-To-Call).  

This allows integration with any CRM. In the menu you also have an option to make the call. 

https://chrome.google.com/webstore/detail/webphone/mcajodgaocmkmmomogbefkghjepgilnc

## Ready to use

I did a free version (hosted in github) that is used by the chrome extension (as popUP).  

If you improve this code, automatically it's reflected in the extension.  

Please send me feedback if you will use. ;)  

### Requirements

* Chrome / Firefox

* [FreeSwitch](https://freeswitch.org/confluence/display/FREESWITCH/Linux) (+WebRTC)  ([guide](https://sipjs.com/guides/server-configuration/freeswitch/))

* Need Https (see)

PS: I have not done any testing using asterisk so far