Brasov, Romania, February 10, 2016 - Mizutech released a new Web to SIP universal library.
Mizutech, a leader in VoIP software development, has announced today that they have released an unique web to SIP solution, compatible with all popular OS and browsers.
Mizu webphone is the first customizable web to sip library with multiple VoIP engines including Native plugin, WebRTC, Flash, Java and App. The “best” suitable engine is selected automatically based on the circumstances (browser support, server support, settings, user preferences).
The webphone is a turnkey Web to SIP software (completely owned by you with no "cloud" dependencies) which can be used:
-as a ready to use customizable solution for webmasters/users (including a web softphone)
-or as a JavaScript library for developers to build any custom solutions
Istvan Fenesi, the owner of the company said: "We think that this multi-engine capability finally solves the "VoIP from browser" problem extending pure WebRTC to SIP solutions to work even when web browser or the server doesn’t have WebRTC support, offering a more reliable way for endusers to access your SIP server, fulfilling the quality requirements of VoIP service providers, call-centers and all others seeking for a robust web based VoIP client”.
The most important features in the webphone include: HTML/CSS softphone user interface, click to call support, unlimited lines, strong codec support (g.729, g.711, GSM, speex, opus, wideband HD audio), IM, call hold, call transfer, conference, encryption, java script API and customizable user interface. Its advanced VoIP features combined with an easy to use interface makes it a favorite software for webmasters, JavaScript developers and all VoIP service providers who wish to offer VoIP services directly from their website. The webphone offers integration possibilities with websites where voice communication between people is in focus, like social networking platforms, forums, support/sales services and callcenters.
For more details visit Mizu web to sip.
******************************************
Mizutech Introduces Universal SIP WebPhone
Brasov, Romania, January 20, 2016 -Mizutech SRL, a VoIP software focused company, has announced today the official launch of its new browser webphone with cross-platform multi-engine technology available for download as JavaScript SIP Client.
This new approach provides a solution for the long awaited problem of platform fragmentation, namely the different, sub-optimal VoIP capabilities of today browsers, including the overhead and known issues of the WebRTC based solution, allowing VoIP providers to leverage a stable browser client solution with call quality comparable to native SIP/RTP clients.
The new Webphone uses the "best" available VoIP technology present in the customer browser depending on the Operating System and browser version including Java Applet, optimized WebRTC, Native Service Plugin, Flash/RTMP and others, providing cross-platform capabilities compatible with all mainstream operating systems (Windows, Linux, MAC, Android, iOS) and browsers (Firefox, Chrome, IE, Edge, Opera, Safari and others).
"At the current fragmented browser market we found a big demand for a web softphone which can be run reliably on all platforms, all browsers, taking out the most from the client capabilities, bypassing most of the WebRTC weaknesses by offering native SIP/RTP capabilities directly from user's browser. After one year of development work we have this ready for production. Our new universal webphone is all about unlocking new possibilities for VoIP service providers and anyone who wishes to add stable VoIP call capabilities directly from browsers covered with an unified JavaScript API and customizable user interface" -said Istvan Fenesi, CEO at Mizutech SRL.
The software can be used as a ready to use browser VoIP client solution (softphone or click to call) or as a JavaScript library providing web developers a simple API to add VoIP capabilities into any web project.
The webphone can connect to any legacy SIP network, including any SIP provider, self-hosted VoIP server, PBX, softswitch, SIP proxy, SIP softphones and SIP devices such as gateways, ATA's and IP Phones, offering most of the common VoIP features such as call forward/mute/transfer/conference, contact list, history, favorites and a wide range of audio codec such as G.729 and wideband HD Audio speex and opus.
The solution is designed for VoIP service providers and web developers providing complete branding, customization, skinning and integration capabilities.