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Protocols |
The Mizu VoIP server is based on the open SIP standards and it has all the common communication protocols built-in to ensure compatibility with a broad range of devices.
Transport protocols: UDP, TCP, HTTP (clear text, XML, JSON, JSONP, SOAP, RDF), websocket (with NAT and proxy handling).
Encryption: HTTPS, TLS, DTLS, SRTP, VPN, custom RSA based and sophisticated obfuscation to bypass all kinds of VoIP blockages in affected countries.
Call protocols: SIP/SIPS, H.323, WebRTC, RTMP with RTP/RTCP for the media.
Codec support: G.729, G.723.1, G.711 (PCMU/PCMA), G.726, G.722, GSM, iLBC, L16, Speex, Opus and bypass all video codec (H.261, H.263, H.264, MPEG 1/2/4, Theora, VP8, VP9)
Codec transcoding and signalling protocol conversion.
A long list of supported RFC's including: 2543, 3261, 2976, 3262, 2617, 3263, 3265, 3420, 3515, 3311, 3581, 3842, 3891, 3325, 2778, 3428, 1889, 2327, 2833, 3264, 3550, 3555 and others.
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Performance |
Using C/C++ language and with high performance in mind, the softswitch has been built by Mizutech from scratch. It boasts with a carefully designed architecture and multithreading to take out the most from your hardware.
The best coding practices and techniques are leveraged at all hot-spot corners of the code, such as IOCP, lock-free data structures, optimized database operations, caching and kernel mode RTP routing.
The service is capable to route up to 8000 simultaneous B2B calls on a standalone server with full stateful proxy and topology hiding. You can easily extend your system by adding more app servers or using SIP load balancer. Optimized for both:
-SMP: it can take advantage of up to 30 CPU core per single instance (or just launch multiple instances if you have more cores)
-low cost hardware: it can easily handle 500 simultaneous call even on hardware which can barely run the windows OS, such as a dual core cpu with 3 GB RAM and 64 GB disk. Scalability can be easily achieved by just adding more app servers.
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Multi-purpose |
The Mizu VoIP server is a multi-functional solution that can be used for numerous purposes, including:
-PBX for small business
-VoIP retail business with Class 5 features
-VoIP wholesale business, VoIP call termination and SIP trunks with Class 4 transit routing
-Carriers and enterprise usage
-Special purposes such as SBC ,SIP proxy, codec transcoding and others
-Special businesses such as residential VoIP, MVNOs, callcenters, calling cards, conference or call-shops
-And many more
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Security |
The VoIP server is secure by default. Some of the techniques leveraged to protect your data and customers are:
-best coding practices and automated tests
-rate limiter (for sip messages, api calls, simultaneous calls, credits)
-built-in dynamic firewall and blacklists
-address/session/user level DOS attack preventions
-flexible robust authentication
-encryption support (TLS/HTTPS, SRTP, custom RSA based VoIP tunneling)
- Windows OS security
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API |
The mizu sever API allows a client program (web/mobile/desktop applications) to connect to a server instance and issue commands defined by the server API.
The API is very flexible about the authentication (multiple levels, username/password or IP based) and the access protocol used (virtually supports all the possible transports: UDP, TCP, TLS, HTTP, HTTPS, websocket, SIP, SMS). The data can be exchanged in plain text, URL parameters, HTML form, ini format, XML, SOAL, RDF and JSON.
A long list of functions are supported, including new user registration, balance/rating request, sendchat, sendsms and many others.
Integration with your existing services is also possible by calling external apps and services or with external authentication/routing and billing. The built-in web control panel is also fully parameterized and it can easily be integrated with your website.
You also have full access to the VoIP database capable to manipulate user records and any other data with a simple SQL.
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Routing |
This is a sophisticated, yet an easy to use module designed to process the incoming and outgoing calls based on your configurations.
Calls between internal users (extensions) are handled automatically, while for outbound calls to mobile/pstn networks you can define your rules to select a destination server (a gateway or a carrier).
You can define multiple routing patterns and for each pattern you can add multiple outbound servers.
Routing patterns can be defined by caller/called/called prefix/tech prefix/caller group/call type/time and day
If more than one direction is set for a routing pattern, the outbound server can be selected using load balancing, priority, weighting, LCR, quality based or a combination of these.
Servers can failover if their statistics are below the thresholds, thus automatically decreasing their priority.
Unlimited number of complex routes, dial plans and rewrite rules are supported.
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Media server |
The built-in media server is responsible for routing the audio, video, fax and other media packets.
-kernel mode packet routing allows a high amount of RTP proxied calls
-full codec support: all common codec's are supported including G.729, G.723.1, G.711, G.726, G.722, GSM, iLBC, Speex, Opus and video codecs
-HD audio wideband support
- codec transcoding (if enabled)
-NAT handling, auto offloading the RTP based on clients� networks, capabilities and settings
-RTCP, announcements, conference, echo test, SDP features
-Built-in TURN and STUN servers
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Billing |
The softswitch implements both prepaid and postpaid billing with flexible configuration.
Create any billing plan from easy to use user interface defining various conditions for billing rules such as caller/called/prefix/time and many others.
The billing can be performed with various billing steps, minimum amount, free amount and has support for multiple currencies including currency per user.
The server will generate a CDR record after each call with detailed billing details. Complex statistics and invoices can be generated based on these records.
Recharge and E-Payment is also supported. A few providers (such as PayPal) are included by default and the list can be extended with new ones.
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Resellers |
The reseller module is a business feature and it means a multilayered child/parent relationship structure to support your resellers/partners/sales agents.
Resellers are supported with unlimited levels of relationship with both post and prepaid billing.
Resellers can administrate their account from a web interface (add endusers, add sub-resellers, create tariff plans, recharge, view CDR's and statistics).
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Unified communication |
UC provides full interconnection for your users with presence, chat, SMS, voicemail, file transfer and remote desktop sharing.
Boost your company effectiveness by combining a broad range of communication features into a complete UC solution and integrate it with your existing systems.
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PBX and Class 5 features |
Most of the tradition PBX call divert features are supported by the server such as call hold, forward, transfer and conference.
Other features includes caller ID, caller ID block, ring groups with call fork, call waiting, speed dial, voicemail, IVR, DID, shared DID, conference rooms, click to call, call-me and many other call features.
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Text and voice recording |
Keep track of your employee communication with chat and voice call recording and call barging for legal interception and user tracking.
The recorded conversation are saved compressed and encrypted but you can easily export them as simple wave or mp3.
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GeoIP |
Use the GeoIP module to assign detailed location information to your users and CDR's at run time.
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Number portability |
With the number portability module the country specific portability database can be imported into the database and the service will take this into consideration when routing the calls, by routing each number to its real owner carrier. LNP is also supported for the local service by off routing ported DID's.
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Tunneling and encryption |
By using encrypted VoIP transport users will be able to communicate confidentially and your VoIP traffic cannot be sniffed and blocked by third party agencies or corporate firewalls. A complex obfuscation layer is also implemented to bypass VoIP blockage all over the word. It works fine even in countries like Iran. The tunneling module is available also as a separate service for integration with third party VoIP servers.
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Webportal |
The VoIP server comes with a portal which implements a control panel for the users, easy to integrate with your main website.
Endusers, resellers, traffic senders and callshop owners are presented with a different set of options so the users can perform the common tasks related to their accounts, such as new user registration, forgot password, show account details, call history and statistics, change settings, such as call forwarding and make payment via payment gateway (PayPal or other) or recharge code.
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Call Center |
The Mizu Callcenter is based on the following features and modules:
-Automatic Call Distribution: for instance, simple automatic dialing, power dialing, predictive dialing, predictive intelligent dialing
-Lead management
-Call Recording: All calls can be recorded and stored
-PBX Features: IVR, callback, call hold, call wait, call transfer, call forward (conditional and unconditional), call conference, barge-in
-Customizable campaigns and scripts: script tree, with any number of branches, answers, and reason codes
-Statistics generation: customer statistics, operator statistics, call related statistics, work time statistics, campaign statistics
-Supervisor access, quality management
-MAgent client application for the agents with integrated scriptable CRM frontend and built-in VoIP
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Calling Card |
The calling card services supported by the softswitch provides you with an attractive business opportunity.
The following features are available on our server, regarding the calling card service:
- IVR
-Pin Generation Management
-Pin-less Number Registration
-Management of PINs generation, activation and deactivation
-Import and export of PIN batches
-Management of call limit and maximum call duration per PIN
-Automatic User Generation
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Call-shop |
Callshop is a module that allows customers to provide low cost international call services in any country.
All parts of the business is covered:
-Server administration, includes callshop creation, routing, billing and administration
-Call-shop owners can login on a web interface to administer their callshop -Call-shop endusers (cabins): custom softphones or ip phones, with balance and rating display.
The call shop owner can monitor their cabins in real time and once a customer finishes the call he pays according to the total cost displayed for its cabin usage.
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Callback and P2P |
With these features users are able to make a call via the server even without internet access.
Callback and P2P (phone to phone call) allows interconnection of 2 endpoints on demand, initiated by making a traditional call to an access number (unconnected) or from the web interface.
The feature is available also via API so it is very easy to add this to your website. The user just visit the page, type its phone number (if not already known) and types the target phone number. Then the server will call both parties and interconnect them once both calls are connected.
These are convenient features when calls from server to client country is cheaper than inverse or the user don't have internet access.
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Registrar |
The built-in fast registrar server allows the clients to connect (register) to the server and will keep a list of their location.
The Mizu VoIP server has its own database to keep the location information and it can handle millions of connected devices.
AAA can be integrated with Radius, LDAP or external database.
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Conference Calls |
The Mizu VoIP server has support for both standard SIP conference and built-in conference mixer, adding third parties on the fly via DTMF commands or API. Conference rooms are also supported via the web/http API or via the 5100-5299 conference room access numbers which are enabled by default if you select the PBX extra module. Built-in support for HD audio / wide-band conference.
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Browser clients |
Browser based softphones and webphones are fully supported by the server via HTML5 WebRTC and Flash RTMP protocols.
By relying on these technologies, service providers can enable users to access their VoIP service while on the go without specialized applications.
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Desktop and Mobile clients |
We provide fully customized softphones with preconfigured account settings, branding, color theme, integration and other customizations for Web, Windows, Android, iOS, Symbian and BlackBerry.
All clients come with a long list of built-in features including G.729 and HD call quality wideband codecs with push notification support.
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WebPhone |
One single webphone that magically works on all OS and all browsers.
The Mizu webphone is a special browser softphone with multiple built-in VoIP engines to allow compatibility will all OS/browsers including WebRTC, Java applet, Flash, Native and App engines.
Beside a turnkey softphone like user interface, other solutions are also included, such as a web click to call button and a JavaScript library that might be used in your projects.
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Supervisor |
The server is supervised on multiple levels:
Built-in supervisor will detect all unusual usage or malfunctions.
Local watchdog service will also monitor the VoIP service and it is capable to act on malfunctions such as sending alerts or restart the service.
External supervisor, to monitor the whole system and notify administrators about malfunctions on custom set events such as bypassing CPU usage limit.
The supervisors can perform various actions when a condition is met, such as: desktop alerts (popup, beep), emails, SMS to admins, execute app or SQL, API call, service/OS restart.
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Easy Administration |
The Mizu VoIP server comes with a comfortable native admin client named " MizuManage" providing easy to use centralized management and monitoring.
All administration can be performed using this single application with a GUI for most of the important settings such as configuration wizard, user management, routing, billing, statistics, CDR's and others.
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And many more |
Check out the Feature list or consult the documentation for more details.
More then 2000 (!) global configuration options (such as sip session timers, ring timeout, max speech length, etc) and 300 configuration option per user/device/extension.
Contact us if your most needed requirement is not on the list.
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