The Mizu SIP/H.323/WebRTC signaling and media stack is a result of our 10+ years of work and experience in VoIP.

The SIP/H.323/WebRTC protocol stack provides all necessary SIP, SDP and RTP functionality, such as encoding, sending, parsing and receiving SIP messages, managing SIP transactions and ensuring reliability. All of them are complete solutions, including everything you need to build a VoIP app or add VoIP capabilities into any existing project.

The server side libraries were written with throughput and stability in mind, while on the client side the compatibility is our first priority. The server sip stack is based on MS-SQL database backend (necessary for advanced features such as VoIP Billing)

SIP/WebRTC client libraries are available for all the major platforms:

• Java VoIP SDK: SIP library for Java SE
• VoIP SDK for Windows: SIP stack for Windows OS
• Android VoIP SDK: SIP library for Android
• iOS SDK: iOS SIP provided "as-is", bundled with other SDK
• WebPhone: JavaScript SIP & WebRTC for browsers
• C++ softphone API for Windows including both server and client libraries: SIP client module
• Cross-platform java/web libraries (java applets and applications for web): VoIP applet
• WebRTC stack
• VoIP Tunnel client for windows/mac/linux (old version)

Most of our libraries has a very similar API to ease the usage.
A demo version is downloadable for most of our SIP SDK/Library, which can be used for development and testing.
Contact us at any time if you are interested in a licensed copy.