V 1.1 is available
Unlike other solutions, the mizu webphone is a truly cross platform effective solution to the "VoIP from browser" problem, using multiple different SIP/media engines targeting different platforms with the optimal solution to take out the most from the client side possibilities usable from the browsers:
We would appreciate any feedback you can provide regarding the webphone, including issues you have encountered, suggestions and feature requests.
Download Webphone (ZIP)
Pricing and order
A webphone is a software program for making telephone calls over the Internet (VoIP/SIP) using a web browser, rather than native applications or a dedicated hardware phone.
Compatible with all VoIP/SIP device and software:
Windows / Linux / MAC / iOS / Android
Flexibility was one of our top priorities. The webphone package can be used in many ways:
The webphone is recommended for:
Check the online demo
Solves the "VoIP from browser" problem
The Mizu Webphone provides a reliable solution for today's fragmented browser market with sub-optimal VoIP capabilities by implementing VoIP engines with multiple technologies and automatically selecting the "best" engine available on client user OS/browser.
This is especially important on websites/projects where VoIP is a key functionality, such as call-centers and VoIP service providers offering call capabilities for their customers directly from browsers with no or minimal quality compromises, connecting directly from user browser to the SIP server without the use of intermediary gateways or protocol convertors.
The webphone is optimized for native SIP/RTP in most circumstances, regardless of user OS and browser.
To sum up, the webphone is focusing on the following goals:
-maximize quality: use the best possible native SIP/RTP VoIP engine directly from client browsers whenever possible
-maximize coverage: true cross-platform capabilities to cover all OS/browser combination, so the users will be able to make VoIP calls regardless of the circumstances (any OS, any browser)
The webphone fulfils the needs of both developers and non-technical users.
You can use the built-in ready to use softphone or click to call solutions, or leverage your custom solution using the numerous configuration options or the webphone java script API, if you are a web developer with JS knowledge.
The mizu webphone provides an easy and reliable way to offer VoIP to your customers, integrated in your website or application.
Cross platform browser VoIP client
The webphone is platform independent so you can use one single app to cover all platforms.
You can forget about the complexity in the background as the webphone will just magically run on all platforms, offering the "best" suitable engine for the endusers.
A single solution supporting all platforms where VoIP in browser is possible including Windows, Linux, MAC OSX, Android, iOS, Solaris, Chrome OS and others.
Native-like VoIP quality and reliability from browsers
With its built-in multiple different SIP/media engines it is able to take out the most from the browsers’ VoIP capabilities using native SIP/RTP whenever possible with a smooth failover to browser technologies such as WebRTC and Flash when needed.
By using the Mizu webphone you will finally be able to offer browser based VoIP services with the same quality as native desktop (softphone) and hardware (IP phone) based solutions.
Ease of use
Copy-paste html code in your website, no developer knowledge is required, with all settings optimized by default. You only have to set your VoIP server address to begin.
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)
It takes only a few minutes to have a functional VoIP client deployed on your website connecting to your SIP server or your VoIP provider account.
Full customization, skinning and branding is supported by numerous settings or using the java script API if you have web development skills. The numerous configuration options will help to closely integrate with your existing infrastructure and to make the most of the offered features.
You can also completely change/remove/create your own user interface using your favorite tool, be it a static page or dynamically generated HTML/HMTL5, CSS/CSS5, AJAX, FLASH, etc.
Based on telecom standards
Connects to any standard based sip server (like Cisco, Asterisk, etc).
Integrated SIP and RTP stack with industry standards codecs including G.729 and wideband HD audio. The webphone can connect directly to your VoIP server or third party IP phones and softphones just like any other standard VoIP client does. The webphone was designed both for SMB or corporations with large call traffic requirements.
Avoid dependency from subscriber based web phone services. Use this webphone component to be able to make calls via your preferred VoIP service provider or via your own VoIP server. Pay once to get a life-time license with no any recurring payment or hidden charges. The Advanced and Gold versions ships without any limitations so the usage is unlimited.
Advantages over pure WebRTC solutions
WebRTC is supported by a part of modern browsers, however it has clear disadvantages when using with SIP networks such as weak codec support (no G.729), browser differences and incompatibilities with its black-box media stack, unneeded extra layer and error phone complexity to convert from websocket/DTLS to clear SIP/RTP with extra server side processing requirements and it is still not supported by all browsers.
To bypass all these weaknesses the Mizu Webphone has built-in native VoIP engines to be used whenever possible.
This doesn't meant that the webphone ignores WebRTC. Actually, it provides an optimized WebRTC stack with seamless integration with your SIP network, automatically used when more native engines are unavailable/disabled in users browser.
The webphone is also usable when WebRTC is not available (not supported by client browser, not supported by server side or disabled by settings).
In the future we plan to add even more WebRTC related features, with a focus to bring WebRTC as close as possible to native SIP/RTP.
Advantages over browser plugins, flash and other similar solutions
Modern browsers are fragmented enough to make it impossible to cover them with one of these solutions and to be able to provide quality services for endusers at the same time.
Browser plugins have to be build/maintained and deployed separately for all browsers and the endusers need to take extra steps to activate them. They have inferior VoIP capabilities due to recent restrictions introduced by browser vendors which is just getting worse and worse over time.
Flash has its own well known issues (only basic code support, requiring an RTMP SIP server side gateway which adds extra complexity and expensive media conversion) and it has also started to be banned in recent browser releases.
By bringing browser VoIP as close as possible to the traditional SIP standards, service providers will be able to unlock new possibilities leveraging browser based VoIP call capabilities with voice quality similar to desktop softphones and hardware IP phones, consistently across all platforms and all browsers.
The multi-engine / unified API is a new unique solution on the VoIP market, offering great opportunities for established and new VoIP service providers and for anyone who wishes to offer VoIP call capabilities from their website with the maximum call quality and coverage possible, for all mainstream platforms including mobile and desktop browsers.
The Mizu webphone is a flexible future proof solution by allowing to easily add/change/re-prioritize the underlying VoIP engines as browsers will change over time.
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