General information

  • App Name: VoIP Applet (formerly known as webphone/websipphone)
  • Description: VoIP softphone java applet plugin for web browsers
  • OS: Windows, Linux, MAC
  • Protocol: SIP/SIPS, RTP/SRTP.  Transport protocols: UDP/TCP/TLS/tunnel
  • Compatibility: all SIP servers (PBX/softswitch), ATA's, gateways, SIP clients, softphones, IP phones

The applet is based on the compact Java SIP library.
Since Java applets are deprecated in modern browser we recommend all customers to upgrade to the new universal webphone for web applications (or to Java SIP SDK for desktop applications). However please note that we haven't discontinued the development of our Java VoIP applet and we don't plan to do so, thus you can continue the usage in controlled environments where Java is available in browsers such as Internet Explorer browsers.

We don't plan to frequently upgrade this page anymore, however we will keep improving the applet in the foreseeable future. The change history can be found here. For the latest version please contact us.

This software have been deprecated as VoIP Applets are not supported anymore by browser.

  • If you are looking for a Java SIP client solution, then check our SIP Libray for Java
  • If you are looking for a web/browser/JS SIP client, then check our WebPhone


Short description

The Mizu VoIP Applet is a SIP standard based VoIP phone software embeddable in webpages as a browser phone but it can be also used as an SDK or as a standalone application, compatible with all VoIP server, softswitch, gateway, ip phone or softphone.

The phone is implemented as a platform independent java applet compatible with any java enabled browser under all OS (Windows, MAC, Linux, Solaris).

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

The VoIP Applet 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 or without the HTTP API or SDK)
  • as a command line VoIP application to automate various tasks
  • as a simple one task module (for example as a click to call button) or a fully featured softphone


  • 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
  • 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)


Advantages over Skype buttons:

Unlike Skype, the VoIP Applet 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:

The Mizu VoIP Applet 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 WebRTC solutions:

WebRTC is still not supported by all browsers and when supported, there are still implementation and quality related problems.
WebRTC is still not supported by most VoIP servers so a separate WebRTC to SIP gateway might be required.
WebRTC will require extra server CPU to convert its websocket and DTLS encryption to the usual VoIP SIP/RTP.
With WebRTC you miss the best narrowband codec's such as G.729 or G.723 and will force you to G.711 (PCMU/PCMA) or Opus (you can still transcode but this requires a lot of server side CPU power and will degrade the performance).

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

  • Click to call functionality on any webpage
  • VoIP service providers can deploy the VoIP Applet 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


  • Java SE capable browser
  • Java Script capable browser when the API is used
  • Microphone and speakers (preferably a headset)
  • No install required (runs as a browser plugin)


webphone screenshots

Try it

  • Demo package

The most convenient way to try the functionalities of the VoIP Applet 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 online VoIP app is also based on Mizu VoIP Applet

  • 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 "" 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.

  • VoIP Applet Demo

The 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 "" using username "testuser102" and password "testpwd102")