• SIP and RTP stack (compatible with any standard VoIP server or device like Cisco, Asterix, softphones, ATA  and others)
• Standard java applet (no software installation required; runs directly from all browsers)
• VoIP calls with auto QoS
• Transport protocols: UDP, TCP, TLS, HTTP tunnel
• NAT/Firewall support, stable SIP and RTP ports, light STUN protocol and auto configuration
• Media protocols: IM (chat) and presence capability
• RFC’s: 2543, 3261, 2976, 3892, 2778, 2779, 3428, 3265, 3515, 3311, 3911, 3581, 3842, 1889, 2327, 3550, 3960, 4028, 3824, 3966, 2663, 3022
• Supported methods: INVITE , ACK, PRACK, BYE, CANCEL, UPDATE, MESSAGE, INFO, OPTIONS, SUBSCRIBE, NOTIFY, REFER
• Codec: PCMU, PCMA, G.729, GSM, SPEEX
• Wideband and ultra-wideband codec’s and stereo output
• DTMF (INFO method in signaling)
• Redial, call hold, mute and transfer (attended and unattended)
• Unlimited lines
• Balance display, Call timer
• Signaling and media encryption
• Click to call
• Server side integration using PHP, .NET,  J2EE , etc
• JavaScript API
• Customizable GUI and languages
• Custom features

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 webphone is based on java, 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 webphone 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)

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.

• 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)
• Easy integration with your existing infrastructure
• Easy integration with your existing website design

Advantages over Skype buttons:

  -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 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.

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

• Click to call functionality on any webpage
• VoIP service providers can deploy the webphone on their web pages allowing customers to initiate SIP calls without the need of any other equipment directly from their web browsers
• Buy/sell portals
• VoIP gadget
• SaaS services
• 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
• HTML Call me button
• Callback and phone to phone functionality
• VoIP enabled sales when customers can call agents

• 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

• Demo package

The most convenient way to try the functionalities 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 webphone on your website in a few minutes connected to your VoIP server.

If you don't have your own webpage yet or you have no access to a VoIP server, then download MizuPhone (or any other softphone) and register to domain sip.mizu-voip.com:37075 with username testuser102, password testpwd102. Than you can call yourself from the web with the WebPhone. If you would like to see calls going to PSTN (mobile/landline numbers) then please contact us and we will give you an account which has enough credit for some test calls.

• Public Internet Phone

The public internet phone is the best way to try and experiment with our webphone technology
Try it from here. It is free.

• Click-To-Call Demo

When you click on the "Call me" button, a call will start immediately to “testuser102”. Be sure that you are already logged in as testuser102 (with a sip phone) for the call to be able to reach you.

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 webphone.