VoIP Server Features

The Mizu VoIP server is a feature rich softswitch fulfilling the needs of both small business and enterprise carriers. The most important capabilities are listed below:


  • Transport protocols: UDP, TCP, HTTP (clear text, XML, JSON, SOAP, RDF), websocket
  • Encryption: HTTPS, TLS, DTLS, SRTP, VPN, custom RSA based
  • VoIP protocols: SIP/SIPS, H.323, WebRTC, RTMP


  • Both old and new SIP rfc's are supported
  • SIP proxy
  • SIP registrar
  • Routed and Direct voice (RTP proxy and offload)
  • Automatic NAT detection
  • Voice Recording and Playback
  • Class 5 features (see details below)
  • RFC 2543 compatibility
  • RFC 3261 compatibility
  • RFC 2976 The SIP INFO Method
  • RFC 3262 Reliability of Provisional Responses in Session Initiation
  • RFC 2617 HTTP Authentication
  • RFC 3263 Locating SIP Servers  
  • RFC 3265 Specific Event Notification 
  • RFC 3420 Internet Media Type message/sipfrag
  • RFC 3515 Refer Method
  • RFC 3311 UPDATE Method
  • RFC 3581 Symmetric Response Routing
  • RFC 3842 Message Summary and Message Waiting Indication Event Package
  • RFC 3891 "Replaces" Header
  • RFC 3325 Private Extensions to the Session Initiation
  • RFC 2778 A Model for Presence and Instant Messaging
  • RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging
  • RFC 1889 RTP: A Transport for Real-Time Applications
  • RFC 2190 RTP Payload Format for H.263 Video Streams  -only routing
  • RFC 2327 SDP: Session Description Protocol
  • RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
  • RFC 3264 An Offer/Answer Model with Session Description Protocol
  • RFC 3550 RTP: A Transport Protocol for Real-Time Applications -replaces RFC 1889
  • RFC 3555 MIME Type Registration of RTP Payload Formats
  • RFC 8599 Push Notification with the SIP
  • RFC 7118 The WebSocket Protocol as a Transport for SIP
  • draft-ietf-mmusic-ice-02 A Methodology for NAT Traversal for Multimedia Session Establishment Protocols
  • draft-ietf-avt-rtp-ilbc-04
  • draft-ietf-sipping-cc-transfer Call Control - Transfer
  • draft-ietf-sip-referredby-05
  • Custom protocol extensions are possible
  • SIP-H.323 protocol conversion
  • SIP-WebRTC protocol conversion
  • Custom protocol extensions are possible
  • Many other SIP related RFC's


  • Transport protocols: UDP/TCP/HTTP; DTLS/TLS/HTTPS; Websockets (WS, WSS)
  • RFC 7118 support
  • Compatible with all popular webrtc clients such as sipjs and sipml5
  • Built-in ICE, STUN and TURN
  • Auto RTP offload or proxy
  • Auto codec conversion when necessary


  • H.323 Standard Features (v.1,2,3,4)
  • Full H.323 proxy
  • H.225.0 Call Signaling
  • Fast Connect/Fast Start
  • H.245
  • H245 tunneling
  • H245 in setup
  • DTMF send/receive
  • Direct endpoint call signaling
  • Gatekeeper routed: call signaling (H.225.0).
  • Gatekeeper routed: call signaling (H.225.0) and control channel (H.245)
  • Gatekeeper routed: call signaling (H.225.0), control channel (H.245) and voice
  • RTP Port Range (for firewalls)
  • Child Gatekeeper capability
  • Backup Gatekeeper capability
  • Gatekeeper clustering support (neighbors, parent/child, alternates)


  • G.723.1
  • G.729
  • G.711 A-law
  • G.711 u-law
  • GSM 06.10
  • MS GSM
  • Speex 2,3,4,5,6
  • G.726 (16,24,32,40 KHz)
  • G.722
  • Opus
  • T.38
  • DTMF
  • Voice:
  • Adaptive de-jitter buffer
  • Voice Activity Detection/Silence Suppression
  • Recording conversations
  • QoS
  • Packet saver technology

Class 5 PBX Features

  • Call Forward All/Busy/No Answer
  • Caller ID
  • Ring Groups
  • Call Return
  • Call Waiting, Call Hold
  • Caller ID Block
  • Selective Caller ID Blocking/Unblocking
  • Speed Dial
  • Three-Way Calling, Conference support
  • Message Waiting Indicator
  • Call transfer  (Attended / Unattended)
  • IVR (Interactive Voice Response supporting applications such as credit card and prepaid services)
  • Conference (3 way SIP, on demand, conference rooms)
  • Voicemail (WMI, auto email forward)
  • DTMF transcoding on server side
  • Video
  • T.38 fax relay
  • Click to call
  • Call me
  • Offline chat
  • And many other built-in PBX features

Call Center

  • Automatic Call Distribution: like simple automatic dialing, power dialing, predictive dialing, predictive intelligent dialing
  • IVR
  • Call Recoding: All calls can be recorded and stored
  • Real time call check out: Supervisors can listen to the ongoing calls real time
  • PBX Features: Call hold, call wait, call transfer, call forward (conditional and unconditional), call conference, CLIP, CLIR
  • Customizable Scripts: script tree, with any number of branches, answers, and reason codes.
  • Customizable IVR: Any number of language, any number of branches, call transfer to the operators
  • Statistic generation: customer statistics, operator statistics, call related statistics, work time statistics, campaign statistics
  • Campaign creation: supervisors can create a campaigns
  • Invitation letter: customization, and automatic printing
  • Report generation: Specific hourly, daily and weekly reports


  • Unlimited accounts
  • Automatic pincode generation
  • Flexible authentication
  • Groups


  • Multi-Carrier Support
  • ACL
  • Sophisticated configurations
  • Load Balancing
  • Rerouting
  • Number rewriting (calling and called)
  • Failovering (multiple levels)
  • Least Cost Routing
  • Call Control Features (Maximum Talk Time, Max Ring Time)
  • Call routing based on PLMN tariff packages
  • Blacklist/White list filtering
  • Fraud detection tools
  • Support for NAT traversal
  • Automatic capacity rebalancing
  • Automatic channel management
  • Number portability support
  • User authentication by  username/password, IP address, techprefix, callernumber
  • Push notification support for mobile and web clients


  • Flexible Rate Definition (peak/offpeak/flat/custom,  enduser/provider/reseller/sales, etc)
  • Automatic and Real Time billing (CDR records already includes the prices)
  • Prepaid and Postpaid platforms
  • Call Credit Limit Control
  • Directions (traffic sender,prefix,gateway,sim packet) and time based billing. Lots of configuration settings.
  • Reporting and price comparisons (LCR)
  • Invoice generation in different formats, PDF generation, email scheduler and invoice printing
  • Complete call rating & accounting services for complex rating schemes
  • Currency and VAT can be set for every packet. Time zone can be changed.


  • Centralized configuration and management for all software and hardware components
  • TManage:
  • -easy to use, mdi style
  • -almost every data query is parameterized with traffic direction and time
  • -all data in one place
  • -lots of data can be obtained from sl,asr,acl forms
  • -global system analysis
  • Dashboard
  • Create and edit network elements
  • Gateways remote maintenance
  • Display of system information
  • Service restart functions
  • Display of the current status of each gateway and channel
  • Real time call supervision (with many grouping options)
  • Real time channel supervision (with many grouping options)
  • Statistics (Text based and graphical ASR,ACD,SL, etc) on any traffic direction and time scale
  • Disconnect Reasons (with many grouping options)
  • CDR monitoring, retrieval, direct CDR access
  • Global system analysis!
  • Routing pattern selection
  • Routing time selection
  • Failovering (in case of channel, gateway, direction etc errors)
  • Best Route Selection
  • Billing module
  • Balance module
  • Real Time Capacity check
  • Ability to insert queries directly into the database
  • Blacklist filtering
  • Self-analysis tools
  • Detailed logging (multiple levels). Detailed call tracing capability
  • Call simulations
  • Reseller/Agent Registration and Management
  • Capacity and system load reports
  • And many more features!

Calling Card

  • Pin Generation Management
  • Pin-less Number Registration
  • Support for multiple account types
  • Management of PINs generation, activation and deactivation
  • Support for unlimited number of PINs
  • Ability to deactivate accounts after certain period or date
  • Import and export of PIN batches
  • Management of call limit per PIN
  • Routing restrictions
  • Max call duration management
  • Automatic User Generation

Other modules

  • Call-back
  • Calling card
  • Web control panel for users
  • Resellers (unlimited levels)
  • VoIP tunneling and encryption
  • Supervisor
  • Text to speech (TTS)
  • SMS via SMPP, SIP MESSAGE or HTTP(S) GET/POST with any payload (JSON, clear text, XML, etc)

There are many other features built into the Mizu VoIP Server. Ask us if you can't find your needs on this list.


This list contain only devices tested by us. Our users are using Mizu with other SIP devices too.

  • ALL7950 02.09.31
  • ALL7950 02.09.33
  • Adore Softphone
  • Alcatel
  • Asterisk PBX
  • Audiocodes-Sip-Gateway-MP-114
  • Audiocodes-Sip-Gateway-MP-118
  • Avaya
  • AVM FRITZ!Box Fon (EU300)
  • AVM FRITZ!Box Fon (fs)
  • AVM FRITZ!Box Fon WLAN 7170
  • BVA8052D (LDTK AR18D ) STUN 0 0 0
  • Broadsoft
  • Cisco ATA 188
  • Cisco IP Phones
  • Cisco-SIPGateway/IOS-12.x
  • CM5K  (610140)
  • CM5K  (706220)
  • CounterPath Bria
  • CounterPath eyeBeam
  • CounterPath X-Lite
  • D-Link/DVG-1402S-1.00.009EU
  • D-Link/DVG-G1402S-1.00.009EU
  • dlink 12-37-5381895-
  • dlink/dph300s
  • DrayTek UA
  • DrayTek UA-1.2.1 Vigor2200V series
  • DrayTek UA-1.2.3 DrayTek Vigor2910
  • DrayTek V3300V
  • Draytel
  • ETK-MP-114FXS
  • Ekiga
  • Express Talk 2.02
  • Evolutiontel
  • fring
  • FWD
  • Gizmo5
  • Gizmo Project
  • Grandstream BT100
  • Grandstream BT120
  • Grandstream GXP2000
  • Grandstream HT488
  • Broadvoice
  • IP Office 4.0
  • Kapanga
  • KPhone
  • Linksys/PAP2
  • Linksys/PAP2T
  • Linksys/RT31P2
  • Linksys/SPA1001
  • Linksys/SPA2102
  • Linksys/SPA9000
  • Linksys/SPA922
  • Linksys/SPA942
  • Linksys/WRP400
  • LR SIP Phone
  • M1000/v.4.80A.025.004
  • Minipax
  • Mitel
  • MSC/VR40
  • NEC
  • NCH Swift Sound Express Talk
  • PA168S
  • Pidgin
  • pjsip
  • PortaBilling
  • RTP300-3.1.17
  • sa210
  • SipDroid
  • Sipgate
  • SIPPER for 3CX Phone
  • Sipura/SPA1001
  • Sipura/SPA2100
  • Sipura/SPA3000
  • Sipura/SPA841
  • SJphone
  • StarTel
  • TeloniaSIP/3.0.1
  • TrixBox
  • VOIP_Agent
  • VoipBuster
  • Voipdiscount
  • VoipGate
  • Voipswitch
  • WengoPhone
  • WRTP54G
  • VoipBuster
  • Vonage
  • Zoiper
  • X-Lite

and many others

Features Presentation


The Mizu VoIP server 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.


Using C/C++ language and with high performance in mind, the Mizu VoIP server 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. Optimized for both:
-SMP: it can take advantage of up to 30 CPU core per single instance (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.


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


The Mizu 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: for maximum security, just disable all windows services such as IIS and others and dedicate the box for the VoIP service. We haven�t had any security events in the last 10 years this way.


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. Admin commands are also available. An example balance request might look like this: http://domain.com/mvapireq/?apientry=balance&authid=USERNAME
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.


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.

  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
-codec transcoding (if enabled)
-NAT handling, auto offloading the RTP based on clients� networks, capabilities and settings
-RTCP, announcements, conference, echo test, SDP features


The Mizu VoIP server 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.


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).

  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.

  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 MWI, IVR, DID and many other call divert functionalities.
  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.

Use the GeoIP module to assign detailed location information to your users and CDR's at run time.
  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.
  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 VoIP server comes with a portal which implements a control panel for the users.
Endusers, resellers, traffic senders and callshop owners will be 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.

  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

  Calling Card

The calling card services supported by The Mizu Voip server provides you with an attractive business opportunity.
The following features are available on our server, regarding the calling card service:
-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


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.

  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.


The 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.

  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.

  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.
  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 and push notifications support.


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 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.


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.

  Easy Administration

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 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.

  Load Balancer

The Mizu SIP Load balancer can handle millions of simultaneous calls, saturating a gigabit link with SIP signaling. Add more app servers on demand when you need more capacity and let the load balancer to distribute the traffic intelligently taking care of the individual app servers health and capacity with auto failover and rebalance capabilities.


VoIP Server

All modules preconfigured and optimized

     The Mizu VoIP server has been built with today business requirements in mind, offering support for a large set of business needs (retail or wholesale - call termination business, simple end-users manageent, sophisticated prepaid/postpaid billing, PBX, IPCentrex, calling cards, flexible IVR solutions, callcenters, etc).
     It has never been easier to build up a VoIP business and start selling VoIP services.  Whether we are installing the services for you or you use the automatic installer and configuration wizard, you will have a ready to use solution within 2-3 hours including all the advanced features like LCR or BRS routing, load balancing, bulk calling card generation, ivr scripts, credit card payments, etc. Delivering real business benefits to our customers, our products are designed to meet the needs of small to enterprise sized business. The Mizu platform has already proved its value in various business environments be it integrated with your existing infrastructure or a standalone server handling millions of customers. Our support is assisting customers in all VoIP related issues helping you to have a successful VoIP business.


The Mizu VoIP server application comes loaded with a set of built-in features to cover all your VoIP related needs. 

  • Built-in authentication, routing, billing with unlimited reseller support
  • Compatibility with all servers/devices/clients/hardware phones/softphones on the market
  • HTML5 WebRTC and Flash RTMP for browser client support
  • Rich codec support, built-in media transcoding, conferencing
  • Any IVR script, callback, phone to phone calls, customer service
  • IM, presence, DTMF and SMS support
  • End-user, reseller and callshop website template
  • Class5 features, call transfer/hold/forward, etc
  • and many others

High performance

     The server is built as a multi-threaded native C++ application with high network throughput, having in mind performance to minimize your hardware costs.
     One single instance can handle up to 10000 simultaneous calls with good scalability from low-end 1 processor systems up to big SMP servers with up to 32 CPU.    
     Scalability is easily achieved by adding more nodes to the system.
     The server has been developed and thoroughly tested using numerous configurations. Its multi-codec ability makes it a great server capable of interacting with practically any existing VoIP systems and bandwidths available with zero configurations to peers; currently used with hundreds of different sip softphones, webphones, gateways, hardware phones and mobile clients with no incompatibility issues.

Zero maintenance

The Mizu server will automatically handle all administrative tasks allowing you to concentrate on business and not on technical details.

  • Auto fine-tune based on your usage statistics
  • Passive and proactive monitoring
  • Automated backups
  • Automatic NAT handling for each registered device. The server will know when it needs to bypass or route the media stream, to save up the most of the bandwidth (can be fine-tuned by manual settings)
  • Scaling to your CPU and memory constraints. The server will detect the number of processors and RAM and automatically fine-tune itself accordingly (number of objects, number of threads, timeouts, etc)
  • Scaling based on server usage (number of calls, number of users, simultaneous calls)
  • Media and signaling timeout detection (with sip timeout timers and monitoring the RTP or RTSP when available)
  • Fine-tune your OS for VoIP (quick task scheduler, QoS, firewall settings, ip keep alive settings, etc)
  • Automated daily database maintenance (finetuning stored procedures, clear old unneeded data, etc)
  • Real-time quality measurements for outgoing routes and direction
  • Failowering when necessary (even during call)
  • Automatic online currency conversion when necessary (once a week by default)
    Watchdog service and service supervisor will automatically start the corresponding action,alert or scrip as needed
  • Email and sms reports (alerts and daily/monthly reports to administrators)

Consistent user interface

     The MizuManage remote management client will minimize your time spent for maintenance with an easy to use "all in one" interface usable even by not qualified people.

  • Easy to Install: You can have the Mizu VoIP server up and running within minutes. Just download the software and follow the installation instructions.
  • Easy to Use: The MizuManage client application makes it easy for administrators to manage all aspects of their VoIP server(s) remotely from anywhere. Intuitive interface with wizards, hints, rich statistics and overall analysis provide a smooth and consistent user experience.
  • Easy data maintenance: all data is stored in a central SQL database allowing easy backup, cloning or migration
  • Easy to integrate: With the server open interfaces, system integrators can fully utilize their existing SQL and/or HTTP knowledge to control the VoIP server behavior. HTTP and database API is exposed to easily implement various VoIP tasks or to integrate it with your existing website (user login, user details, webphone, callback, phone to phone calls, etc.).

Secure by default

Your server is secured by default:

  •   Password hardening and automatic account lockout
  •   Rate limiter
  •   Capable to handle large scale Denial of Service (DoS) attacks (even flooding with 1 Gbits rate)
  •   Auto blacklist filtering
  •   Concurrent call limit per account and per group
  •   Consumed credit limit per account and per group
  •   Credit card authentication with auto ban
  •   E-payment pre-authorization, validity checks and 128 bit SSL security
  •   Dynamic firewall with brute-force and dos attack detection
  •   Early message inspection
  •   Realtime billing with credit/profit protection
  •   HTTP API with automatic ip blacklist
  •   Encrypted communications with capable endpoints
  •   Orphan call detection, Ring and speech length timeout, media timeout and other limits

For more details please consult the server Admin guide and the other documentations or have a look at the demo server or the reseller/enduser/callshop web interface.

Contact serversupport@mizu-voip.com for a trial access to a server hosted by us or a test install on your own server.