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SIP Web Client -Description

v.4.0. is available     

The Mizu universal WebPhone is a SIP standards based VoIP client software embeddable in any web page as a Browser Softphone, or used as a JavaScript SIP library to build your custom web based VoIP solution, be it a simple click to call button or complex solution integrated with your existing business logic. Based on the industry standard SIP protocol, it is compatible with all VoIP devices and services. The web sip client enables voice calls from/to any computer (PC, MAC, laptop, tablet, mobile), right from a webpage with complete call control such as hold, transfer, conference, record and others. It can call any other SIP phone (softphone or ip phone for free charge) or any landline and mobile number via a VoIP service provider of your choice including your own SIP server/softswitch/PBX.

The webphone is a self-hosted web VoIP client, shipped with life-time license, totally controlled and owned by you. As a cross platform JavaScript SIP library, the webphone is a solution for the "VoIP from browser" problem, using multiple different SIP/media engines targeting different platforms with the optimal solution in order to take out the most from the client side possibilities usable from the browsers, all covered with a simple to use unified API. 

Included web sip engines:

  • WebRTC SIP library: for modern browsers with HTML5/WebRTC support acting as a WebRTC client (SIP signaling in websocket)
  • NS engine: native service/browser plugin (using traditional UDP, TCP or TLS for the SIP signaling)
  • Java VoIP engine: for all java enabled browsers providing native SIP/RTP
  • Flash VoIP: for compatibility with some old browsers
  • NPAPI: native sip plugin for browsers with NP-API support
  • App: for platforms where WebRTC and browser VoIP plugins are otherwise impossible (old iOS/Safari)
  • P2P and Callback: if your VoIP server has an API for these and no better alternatives found
  • Native dial: might be used on mobiles when VoIP is not possible (for example no network access)

The "best" suitable VoIP engine is automatically selected based on browsers/OS capabilities and server support (You can also set or prioritize the engines to be used by the configuration).
The default user interfaces are implemented as HTML/CSS which can be customized after your needs. Developers can use the API to implement any custom functionality or a custom design (any technology with JavaScript binding including HTML, CSS, Flash or generated from server side by PHP, .NET, J2EE, Node.js or others).

The download package includes the followings:

  • the software itself to be copied to your website including all engines
  • documentation
  • JavaScript SIP library: an easy to use SIP JS API to implement your custom VoIP solution
  • a turn-key web softphone implementation: you can easily rebrand and customize or use it as-is as a web phone on your website
  • a click-to-call button implementation: a simple VoIP click to call solution
  • other usage examples and templates

The public version has some limitations and a final package is sent to you on your order (you can use the public demo version for all tests, development and integration and then just replace it with your final build once you are ready)


Download webphone package

How it works?

web sip softphone


What is a webphone?

A webphone is JavaScript SIP client for making telephone calls over the Internet (VoIP/SIP) using a web browser, rather than native applications or a dedicated hardware phone.
See how it works.

WebPhone Features

  • Runs in any browser and all OS using WebRTC, NS (native service plugin), Java applet or Flash engine (Firefox, Chrome, IE 6+, Edge, Safari, Opera and others on Windows/Linux/MAC/iOS/Android and others, including WebViews)
  • Standard JavaScript SIP client for voice calls (in/out), video calls, chat, SMS, conference and others
  • SIP and RTP stack compatible with all SIP servers/softswitch/PBX and devices like Cisco, Voipswitch, Asterisk, softphones, ATA and others
  • Transport protocols: IPv4/IPv6, UDP, TCP, HTTP, RTMP, websocket (uses UDP for media whenever possible with failback to TCP or HTTP when necessary)
  • Encryption: SIPS, TLS, DTLS, SRTP, end to end encryption for webphone to webphone calls
  • Seamless protocol conversions if necessary: RTMP to SIP, WebRTC to SIP, SIP to WebRTC for browser-SIP (protocol conversion avoided whenever possible)
  • NAT/Firewall support: auto detect transport method (UDP/TCP/HTTP), stable SIP and RTP ports ,keep-alive, rport support, UPnP, proxy traversal, auto tunneling when necessary, ICE/STUN/TURN protocols and auto configuration, firewall traversal for corporate networks, VoIP over TCP/HTTP (using UDP transport whenever possible, otherwise it will use encrypted signaling and media over HTTP, TCP tunnel or TURN when firewalls blocks UDP ports) and it has full support for ICE TCP candidates
  • Works over the internet and also on local LAN’s (perfectly fine to be used with your own internal company PBX)
  • Audio Codec: G.711 (PCMU, PCMA), G.729, GSM, iLBC, SPEEX, OPUS (wide-band HD audio)
  • Video Codec: H.263, H.264, VP8, VP9 for WebRTC only for native HTML5 video chat
  • RTC video calls and screen-sharing (optional "as-is")
  • SIP compatible codec auto negotiation and adjustment (for example G.729 - wideband or WebRTC G.711 to G.729 transcoding if needed)
  • Audio enhancements: PLC (packet loss concealment), AEC (acoustic echo canceller), Noise suppression, Silence suppression, AGC (automatic gain control), high-quality low-latency audio and auto QoS
  • Call divert features: rewrite, redial, mute, forward, hold, MOH, transfer (attended and unattended), conference, multiple lines management
  • Call park and pickup, auto-answer, barge-in
  • Voice call recording, audio file streaming
  • IM/Chat, group chat, offline messaging, SMS, file transfer, DTMF, voicemail MWI
  • Multi-line support (multiple simultaneous calls)
  • Multi-account support (multiple SIP accounts)
  • Background call notifications (NS engine can auto launch the browser on incoming calls if configured accrodingly)
  • Contact management: flags, synchronization, favorites, block, presence (DND/online/offline/others) and BLF
  • Extra NS engine features: SIPREC, IMS/3GPP, UUI, 3PCC
  • Balance/rating display, call timer, caller ID display
  • Auto re-connect, re-register and re-dial to maximize reliability on low quality networks or bogus servers
  • No server side dependencies. You can use the webphone to add VoIP to any kind of webpage, be it a simple static page or a java script oriented website. In case if you are using a server side stack, the webphone can be integrated with any technology including PHP, .NET, java servlet, J2EE , Node.js and others. Use any OS and any web server (IIS, Apache, nginx, NodeJS, java, others)
  • Fine-tuned out of the box for general usage, for corner cases, and with a special care for major VoIP server platforms such as Asterisk, FreeSWITCH, VoIPSwitch, Cisco, FreePBX, FusionPBX and others
  • Includes several different technologies (engines) to optimize the VoIP experience across all browsers: Java VoIP applet, WebRTC SIP client, NS (Native VoIP Service or Plugin), Flash VoIP, App (Web Softphone application for mobiles with auto-provisioning), Native dial and server assisted conference rooms, P2P and callback.
  • High level JavaScript SIP API: web developers can build any custom VoIP functionality using the webphone as a JS library
  • Stable API: new releases are always backward compatible so you can upgrade with no changes in your code
  • Integration: the webphone provides an easy way to integrate it with your server (user sign-up, sms, balance, callback, recharge and others website/app-store/VoIP/API usage)
  • Branding and customization: Use with your own brand. Customizable, feature rich responsive user interface with ready to use, modifiable skins
  • Flexibility: all parameters/behavior can be changed/controlled by URL parameters, preconfigured parameters and/or from java script
  • Contact us if your most needed requirement is not on this list

Compatible with all standards based VoIP/SIP device and software:

  • VoIP server/softswitch: Mizu, Voipswitch, FreeSWITCH, Cisco, Siemens, Huawei, others
  • Software PBX: Asterisk, 3CX, Broadsoft, Brekeke, Yate, FreePBX, Elastix, Trixbox, others
  • Service: Twilio, Vonage, others
  • Hardware PBX: CUCM, Avaya, Alcatel, NEC, Mitel, others
  • SIP proxy: SER, OpenSIPS, reSIProcate, others
  • SIP softphones: X-Lite, Bria, Jitsi, Zoiper, Linphone, others
  • SIP SDK: PJSIP, PortSIP, oSIP, others
  • WebRTC clients: webrtc2sip, sipml5, SIP.js, JsSIP, others
  • Devices: gateways, ATA’s, IP Phones, doubango, others
  • VoIP service providers: Vonage, hosted sip providers, others
  • Any SIP capable endpoint: UAC, UAS, proxy, others


Windows / Linux / MAC / iOS / Android
Chrome - Firefox - IE - Edge - Safari - Opera - Others

Usage examples

Flexibility is one of our top priorities. The webphone package can be used in many ways:

  • JavaScript VoIP library for developers
  • As a ready to use web softphone running from your website, so the user will not have to install a separate standalone softphone software (just set your VoIP server address to go)
  • Custom web phone (you can customize the web softphone including sip settings, design and branding)
  • VoIP service providers can deploy the mizu webphone on their web pages allowing customers to initiate SIP calls without the need of any other equipment directly from their web browsers
  • Add VoIP capabilities for any software
  • As ready to use VoIP click to call solution (just preconfigure with a sip account and a number to call)
  • Web SIP client for Callcenters (easy to integrate with existing call center software suite or CRM frontpage)
  • Browser VoIP SDK to build your product
  • Hotel guests (international calls, intercom, door control, etc)
  • Self-hosted Twilio alternative
  • Integrate with any JavaScript framework or library
  • Integrate with any CRM or web portal (here is a tutorial to integrate webphone with Salesforce)
  • Integrate with any web server framework, library or language, including PHP, .NET, J2EE, NodeJS, Perl, C++, Python, Ruby on Rails, Express.js, Django and others
  • Integrate SIP client with any custom website, drupal, joomla, wordpress, phpBB, vBulletin and others as a plugin, module, API or just load the softphone skin in an iFrame
  • Use from plain/vanilla JS or from any framework such as React, jQuery, Angular, Vue, Ember, Backbone and others
  • High level WebRTC SIP API which solves all the usual WebRTC related issues (working from corporate networks, proper TURN settings, codec conversion and the other common issues)
  • SIP client browser plugin
  • Push to talk solutions
  • Click to call from email signature or JavaScript web click to call software
  • SaaS services, hosted or cloud sip web client
  • Embedded VoIP client in various devices (PBX box, IP phone and others)
  • Web Phone client for third party systems and CRM's
  • VoIP for PHP, .NET, JSP, Node JS or any popular server script language
  • Web SIP client for social networking websites, facebook softphone
  • Real-time marketing, HR management, advertising, dating and financial services
  • As a portable communication tool between company employees
  • VoIP for sales and support pages where people can call your agent from your website.
  • VoIP for blogs and forums where members can call each other
  • Real time communication and collaboration
  • Integrate with any external app (CRM/Ticketing/Lead management/Sales engagement/Support chat)
  • As a webrtc sip client or WebRTC softphone
  • JS Web phone for Asterisk, web SIP client for FreePBX and other servers
  • As a JavaScript SIP API implementing a SIP client from JavaScript (JavaScript SIP SDK)
  • Interconnect with any sip service provider or use your own IP-PBX
  • Make cheap outbound calls to landline/mobile
  • Make free calls to other webphones, SIP endpoints, softphones or IP phones (including agent to agent calls)
  • Accept incoming VoIP or PSTN calls to your office/sales/support departments
  • VoIP call voice recording: web VoIP call recording from any browser
  • HTML Call me button
  • Remote work / remote office
  • VoIP call from Email signature
  • Help desk VoIP call from browser
  • Unified communication web client
  • OEMs to bundle VoIP into any distributed software package
  • For voip service providers to offer click to call functionality for their customers
  • SIP phone plugin for CRM as a javascript sip library
  • Complete HTML5 softphone implementation
  • Offer cloud based dialer or click to call services for your clients or other companies
  • Office web communication (ERP, CRM, SCM, FFM)
  • Callback and phone to phone calls handled by VoIP server
  • JavaScript SIP client specialized for Operators, ITSPs, Call centers, Integrators, PBX providers
  • The rest is up to your imagination...

The webphone is recommended for:

  • voip service providers: fully customizable/branded solution optimized for your SIP server
  • non-technical people with a need to add VoIP to website: copy-paste html code, just set your SIP provider address in the configuration to begin the usage
  • web developers: build any web VoIP solution using the JavaScript API
  • web designers: easily add VoIP call capabilities to any website, modify existing skins or create your own with simple HTML/CSS
  • call centers
  • companies with local SIP server or PBX: allow SIP calls from web (all browsers are supported on all popular OS)
  • software developers: add standard VoIP to any software such as CRM or embedded devices
  • web-site owners: add VoIP call capability to any website using any VoIP/SIP service provider
  • individuals, small-businesses and enterprise corporations

Check the online demo


Why should I use?

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 javascript web sip client 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 web softphone or click to call solutions, or leverage your custom solution using the numerous configuration options or the sip 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 as a Javascript SIP client library.

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 SIP web client will just magically run on all platforms, offering the "best" suitable engine for the endusers.
All this is covered by a simple JavaScript API allowing web developers to create custom VoIP solutions from any Java Script framework or with pure JS.
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.
Can be integrated into any environment, be it a simple static page, a JavaScript application (pure or any framework), any server side technology (PHP, .NET, JEE, NodeJS, etc) and using any webserver (Apache, IIS, nginx and others).
Let your clients easily initiate voice calls directly from your website without the need to download any software. The web phone will be hosted by your webserver (like regular html/png/etc files, no any runtime is required).
Calls can be initiated from scripts, from user input by typing a phone number, by selecting from contact list, by a click to call button or by your custom application logic using the JavaScript API or generated from server side scripts.
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 browser sip phone was designed both for SMB or corporations with large call traffic requirements.

Life-time license
Avoid dependency from subscriber based web phone services. Use this webphone component to make calls via your preferred VoIP service provider or via your own VoIP server. Pay once to get a perpetual selfhosted license with no any recurring payment or hidden charges. The Advanced and Gold versions ships with unlimited usage, without any user or call limitations.

Advantages over pure WebRTC solutions
WebRTC is supported by most 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, difficult setup to work across corporate firewalls, 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: compare web sip clients.
To bypass all these weaknesses the Mizu Webphone has built-in a highly optimized WebRTC implementation providing seamless integration with your SIP network, automatically used when more native engines are unavailable/disabled in users browser, configured optimally by default with transparent protocol convert from SIP to WebRTC and WebRTC to SIP. Unlike SIPML5, SIP.js or JsSIP, the mizu web sip library is also usable when WebRTC is not available (not supported by client browser, not supported by server side or disabled by settings) and when WebRTC is available, then it provides an optimized WebRTC implementation with robust SIP integration.

Advantages over browser plugins, NPAPI, Flash, Java 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. NPAPI support is already removed from Chrome and planned to be removed also by Firefox, deprecating applications such as Zoiper webphone and Linphone.
The webphone, although it has all these engines, it doesn't depend on any of them. It will always select the optimal SIP backend based on browser capabilities.

New approach
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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