Mizu Webphone 1.0.2

Mizu Webphone 1.0.2 provides you with a brilliant way to implement your target. It is a lightweight standard based VOIP phone that can be run even from web-pages. Based on the industry standard SIP protocol, it is compatible with all VOIP devices and services. It can call any other SIP soft phone (for free charge) or any land-line or mobile number via a VOIP service provider of your choice.

The phone is implemented as a java applet/application and it is completely independent platform running on webpages, windows desktop, MAC, Linux, Solaris and mobile devices.
It can be used as a normal softphone running on your website or as Skype-like buttons (Click to Call).
Compared to Skype buttons it has 3 main advantages:

1. Based on standard SIP protocol. This means more control, and you can change your phone service provider whenever you want. There are plenty VOIP providers offering free services too.
2. No client side applications have to be installed. If the browser supports java than the phone will run "from the web"
3. 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)
   

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 land-line or mobile phone numbers. All usual call routing features can be enabled (call forward, call transfer, etc)

With Mizu-WebPhone, you can quickly add VOIP capabilities on your website. For example:

• VOIP enabled support pages when people can call your support people from your website.
• VOIP enabled forums when members can call each other
• VOIP enabled blogs
• VOIP enabled sales when customers can call agents
  

The rest is up to your imagination.

Major Features: 

1. SIP and RTP stack (compatible with standard VOIP servers like Cisco or Asterix)
2. Standard java applet (no installation required. runs directly from browsers)
3. Standard G711 codec’s (PCMU and PCMA) and speex narrow-band
4. DTMF (INFO method in signaling)
5. Basic call features
6. IM (chat) capability based on SIP SIMPLE protocol
7. Other VOIP related features are coming soon (IM/chat, call forward/transfer/mute and more codec’s)
8. Encryption capability

Enhancements: New SIP engine

Requirements:

1. Intel Pentium II 500 MHz
2. 128 MB RAM
3. 30 MB free space