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 Short description Minimize

V.5.4 is available

The Mizu WebPhone is a lightweight standard based VoIP phone software embeddable in any webpage as a browser phone, but it can be also used as an SDK or as a standalone application. Based on the industry standard SIP protocol, it is compatible with all VoIP devices and services. It can call any other SIP phone (softphone or ip phone for free charge) or any landline and mobile number via a VoIP service provider of your choice.

The phone is implemented as a platform independent java applet/application and it is running on any java enabled desktop or browser under all OS (Windows, MAC, Linux, Solaris). The called person can accept the call on a VoIP device (soft phone, IP phone, call-center application) or can be contacted directly on their landline or mobile phone numbers. All usual call features are implemented (call forward, call transfer, conference, etc)

With Mizu Webphone you can quickly add VoIP capabilities to your website or application (homepage, blog, forum, support/sales page, social networking site, callcenter, software integration, etc)

The webphone can be used as:
  • static html: for example as a fully featured pre-customized softphone or click to call button on your website
  • dynamic html: applet parameters generated dynamically from your server side script, for example a click to call button with the called number changing depending on the content
  • controlled from the client side with the JavaScript API
  • a mix of the above
  • as a standalone desktop application (with GUI or from command line; with or without the HTTP API or SDK)
  • as a simple one task module (for example as a click to call button) or a fully featured softphone
The rest is up to your imagination.

    
 Features Minimize
  • Standard SIP client for voice calls (in/out), chat, conference and others
  • SIP and RTP stack compatible with any standard VoIP servers and devices like Cisco, Voipswitch, Asterix, softphones, ATA and others
  • Standard java applet (Runs from browsers under all popular OS. No native installer needed. Java is not needed on your servers)
  • Connects directly to the VoIP server or to peers (no need for any intermediary media server or relay)
  • Transport protocols: UDP, encrypted UDP, TCP, TLS, TCP tunnel, SOCKS proxy traversal, HTTP proxy traversal, HTTP, VPN tunneling, tunnel*
  • NAT/Firewall support: stable SIP and RTP ports ,keep-alive, rport support, fast ICE/fast STUN protocols and auto configuration
  • Peer to peer encrypted media
  • Standard encryption: TLS, DTLS (optional), SRTP
  • Optional signaling and media tunneling and encryption*
  • RFC’s: 2543, 3261, 2976, 3892, 2778, 2779, 3428, 3265, 3515, 3311, 3911, 3581, 3842, 1889, 2327, 3550, 3960, 4028, 3824, 3966, 2663, 3022 and others
  • Supported methods: REGISTER, INVITE, reINVITE, ACK, PRACK, BYE, CANCEL, UPDATE, MESSAGE, INFO, OPTIONS, SUBSCRIBE, NOTIFY, REFER
  • Audio codec: PCMU, PCMA, G.729, GSM, iLBC, SPEEX, OPUS
  • Video codec: H264, H263, H261, MPEG1, MPEG4, MPEG2, VP8,Theora
  • HD Audio: Wideband, ultra-wideband and full-band codecs (speex, opus)
  • Audio enhancements: Stereo output (will convert mono sources to stereo) , PLC (packet loss concealment), AEC (acoustic echo canceller), Noise suppression, Silence suppression, AGC (automatic gain control) and auto QoS
  • Conference calls (built-in RTP mixer)
  • Voice recording (local and/or ftp upload), custom audio streaming
  • DTMF (INFO method in signaling or RFC2833)
  • IM/Chat (RFC 3428), SMS and presence capability
  • Redial, call hold, mute, forward and transfer (attended and unattended)
  • Call park and pickup, barge-in
  • Balance display, call timer, inbound/outbound calls, Caller-ID display
  • Voicemail (MWI)
  • Click to call
  • Additional features: call parking, early media, local ring-back, PRACK and 100rel, replaces
  • Server side integration using PHP, .NET, J2EE , Node.js, etc
  • Integration with any webpage or third party application
  • API: HTTP API, JavaScript API, Native API/VoIP SDK
  • Branding and customization: Use with your own brand. Customizable user interface, skins and languages (with ready to use, modifiable skins)
  • Custom features
  • Unlimited lines
  • Flexibility (all parameters/behavior can be changed/controlled by applet parameters and/or from java script)

FirefoxIEChromeOperaSafari

Windows / Linux / MAC


    
 Highlights Minimize

Easy to use
Let your clients easily initiate new voice calls directly from your website without the need to download any software. The web phone will be hosted by your webserver (one single file)
Calls can be initiated by typing a phone number, by click to call functionally or by your application logic using the JavaScript API.

Easy to deploy
Copy-paste html code in your website. You have to set only your VoIP server address to begin.
All other applet parameters are fine-tuned by default (although there are more than 50 parameters that you can change for a full customization)

Platform independent
The mizu webphone is based on the cross-platform Java SE, which is supported by lots of devices, all major OS (Windows, Linux, MAC, etc) and all major browsers (IE, Firefox, Chrome, Opera, Safari, etc)

Customizable
Full customization is supported by web sip phone applet parameters or using the java script API.
You can even completely change the user interface using your favorite tool (HTML, DHTML, AJAX, FLASH, etc) or control from the server side (PHP, .NET, J2EE, etc)

Based on telecom standards
Connects to any standard based sip server (like Cisco, Asterix, etc) without the need of any third party software.
Integrated sip and rtp stack with industrial codecs. No third party media server is needed, just plug and play. The web sip phone will connect directly to your VoIP server just like any other standard VoIP client does.


    
 Benefits Minimize
  • Compatible with all browsers (IE, Firefox, Safari, Opera, Chrome, etc) and all OS (Windows, Linux, MAC, etc) with Java SE support
  • Full compatibility with your VoIP servers including Class 5 features
  • Users don’t have to download anything to be able to initiate true VoIP calls
  • No need for third party media server. Full SIP functionality is embedded so it can connect directly to your server like any other hardware IP phone or softphone. Not ActiveX based.
  • Easy to use and easy to deploy (copy paste HTML code)
  • Highly configurable by (not only) web developers
  • Easy integration with your existing infrastructure. It takes you only a few minuted to get started
  • Easy integration with your existing website design. Use/modify the default skins or create your own with simple HTML/CSS
  • The easiest way to offer VoIP for your customers integrated in your web or application

    
 Advantages Minimize

Advantages over Skype buttons:

Unlike Skype, the Mizu webphone is based on standard SIP protocol. This means more control, and you can change your phone service provider whenever you want. There are many VoIP providers offering free services too. No client side applications have to be installed. If the browser supports java then the phone will run "from the web"
It is compatible with any VoIP service provider or for more control, you can use your own VoIP server (there are several free open source servers like Asterisk, or you can buy a cheap VOIP server with support)

Advantages over proper web based communication software’s:

Mizu WebPhone is based on SIP and can be integrated with any other standard VOIP server. In this way beside to make calls between your users you have the possibility to make real calls to any landline or mobile phone.
There are several free VOIP server that you can use for this purpose, for example Asterix or Mizu Softswitch.

Advantages over ActiveX solutions:

You can find many ActiveX web phone solution, but the time is over for ActiveX. These are simple executables (running only on windows) and because this they have a bad security reputations. Now it is a deprecated technology, and by default they are disabled in all browsers. In short: they are useless.

Advantages over NPAPI solutions:

NPAPI doesn't work on IE which is the browser with the biggest market share. In addition NPAPI solutions are much more inflexible than java.

Advantages over HTML5 solutions:

WebRTC in HTML5 is still not reliable for VoIP and it is supported only by 2 browsers (Firefox and Chrome) in their beta versions only (users have to turn on manually in the settings)

Advantages over Flash based solutions:

VoIP calls can be made with flash, but it is a very inefficient and complicated solution. You have to install a separate flash media server for this (or rent) and do the media and signaling conversion there, because flash doesn’t know VoIP protocols and doesn’t have standard codec’s. This is a very CPU intensive process with high failure rate.  Also it is a known fact that flash clients has lower quality and high voice delay.

That is why we created our java based unique solution. True VoIP calls from any webpage from now is an easy task that works!


    
 Usage examples Minimize
  • Click to call functionality on any webpage
  • VoIP service providers can deploy the mizu webphone on their web pages allowing customers to initiate SIP calls without the need of any other equipment directly from their web browsers
  • For web based callcenters
  • Add VoIP capabilities for any software
  • Buy/sell portals
  • SIP browser plugin
  • jQuery phone plugin
  • VoIP gadget
  • SaaS services
  • Browser VoIP SDK to build your product
  • Embedded VoIP device
  • VoIP CRM integration
  • VoIP plugin for PHP, .NET, JSP or any popular server script language
  • Social networking websites
  • As a portable communication tool between company employees
  • VoIP enabled support pages where people can call your support people from your website.
  • VoIP enabled blogs and forums where members can call each other
  • As a facebook  phone
  • Wordpress voip plugin
  • HTML Call me button
  • VoIP call from Email signature
  • Help desk VoIP call from browser
  • Browser phone plugin
  • For voip service providers to offer click to call functionality for their customers
  • Callback and phone to phone functionality
  • SIP phone plugin for all popular CRM and blog web engine
  • VoIP enabled sales when customers can call agents

    
 Requirements Minimize
  • Java SE capable browser (all commonly used browsers)
  • Java Script capable browser when the API is used (all commonly used browsers)
  • Microphone and speakers (preferably a headset)
  • Minimum 400 MHz P3 or similar processor for the advanced codec’s (e.g. g.729, speex wideband)
  • No software or plugin installation required

    
 Webphone screenshots Minimize

webphone screenshots


    
 Try it Minimize
  • Demo package

The most convenient way to try the functionalities of the mizu webphone is to download the demo package which includes a trial edition, documentation and html/java script examples. If you have a minimal HTML and VoIP knowledge you should be able to "install" the websipphone on your website in a few minutes connected to your VoIP server.

  • Skin demo

Have a look at the skin templates here.

  • Public Internet Phone

This free service is also based on mizu webphone.
Try from here.

  • Click-To-Call Demo

When you click on the "Call me" button, a call will start immediately to “testuser102”. To be able to receive the call you must register first with any softphone to "sip.mizu-voip.com" using username "testuser102" and password "testpwd102".

         

On your webpage the design of the buttons is up to you. The availability of the users (with different buttons) can be loaded from a database.

  • Web Softphone Demo

This is a demo application that is able to call any VoIP, mobile or PSTN number. Some features are disabled in this demo. Click the button below to launch the web sip phone. (To be able to receive a test call you must login with any softphone to "sip.mizu-voip.com" using username "testuser102" and password "testpwd102")

        Web Phone


    
 FAQ Minimize

What is a webphone?

A webphone is a software program for making telephone calls over the Internet (VoIP/SIP) using a web browser, rather than native applications or a dedicated hardware phone.

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