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)
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!
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.
Have a look at the skin templates here.
This online VoIP app is also based on Mizu VoIP Applet
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.
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 "sip.mizu-voip.com" using username "testuser102" and password "testpwd102")
Download VoIP Applet
Get support
Download (ZIP)
One-minute guide (PDF)
Documentation (PDF)
Documentation (Online HTML. Old.)
Change log
Screenshots
Pricing
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