Mizu Softswitch is a rich featured VoIP server system running on Microsoft Windows platform combining ease of use with high stability and throughput making it a perfect choice for enterprise VoIP service providers, carriers, telecom startups and small business companies.

Main modules include:

• SIP and H323 stack (voice + video + presence + messaging)
• callcontrol, AAA and routing
• user management
• media service (RTP/RTCP, NAT handling, conferencing, etc)
• prepaid/postpaid billing and e-payment
• callback
• SMS (in/out)
• IVR, calling cards
• callshop
• reseller module
• encryption and tunneling
• supervisor (watchdog service and alerter client)
• callcenter: CRM and predictive dialer (optional)
• softphone, webphone and mobile client (optional)
• web interface
• remote admin client (MizuManage)

Easy management

     You don’t have to be a highly qualified technician and you don’t have to pay to learn how to use your server. If you are able to use your windows desktop for everyday tasks, then most probably you will be able to manage your VoIP network without any big effort. The VoIP server supports full remote configuration and developers with SQL knowledge can easily modify it's background data structures

• Easy to Install: You can have the Mizu VoIP server up and running within minutes. Just download the software and follow the simple 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 easy analysis provide a smooth user experience.
• Easy maintenance: all data is stored in a central MS-SQL database allowing easy backup and clustering when needed.
• Open interfaces: System integrators can fully utilize their existing MS-SQL knowledge to control the VoIP server behavior. HTTP and database API is exposed to easily implement various VoIP tasks (callback, phone to phone calls, etc.).

Business centric

     The mizu server has been built with your business in mind, offering support for a large set of business needs (end-users, 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 making money.  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 will offer you a feature complete solution out of the box for various type of VoIP businesses:

• Wholesale platform
• Retail business platform
• Broadband VoIP services
• Used as B2BUA
• DID origination consumers
• IPcentrex and virtual PBX solution
• Multiple virtual servers on the same box
• Callcenter platform with built-in CRM and operator client
• SaaS service
• Phone to phone services
• PC to phone services
• PC to PC and unified communication services
• Any 3rd-party sip device, softphone, ATA, Gateway, IPPhone, SIP Proxy, Registrar and SBC supported
• Callshop
• Pre-paid phone cards
• Callback
• SMS
• Softphone and webphone connectivity
• Calling card services
• E-payment, PayPal
• VoIP encryption and tunneling to bypass all VoIP filters (e.g. UAE)
• Can be used as SBC
• and others

Feature-rich

The Mizu 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
• Rich codec support, built-in media transcoding, conferencing
• Any IVR script, callback, phone to phone calls, customer service
• SMS, SMS callback, other SMS triggered operations
• IM, presence
• End-user, reseller and callshop website template
• Class5 features, call transfer/hold/forward, etc
• and many others

High performance, reliability and compatibility

     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 server can handle up to 10000 simultaneous calls with good scalability from low end 1 processor system up to big SMP server with up to 32 CPU.
     The server has been developed and thoroughly tested using numerous configurations with no downtime. 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, hardware phones and mobile clients with no incompatibility issues.

Zero maintenance

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

• Automated backups with the built-in scheduler or by MS-SQL backups, mirroring, log shipping or replication.
• 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)
• 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 a call)
• Runtime change of built-in fileserver passwords
• Online currency conversion when necessary (once a week by default)
• The server is constantly monitored by watchdog service and supervisor which will automatically start the corresponding action when needed
• Email and sms reports (alerts and daily/monthly reports to administrators)

Secure by default

Your server is secured by default:

•   Auto blacklist filtering
•   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
•   HTTP API with automatic ip blacklist
•   Encrypted communications when possible (for example with Mizu softphones)
•   Auto check end-user credentials (minimum password length, etc)
•   Ring timeout, speech length limits and other limits

The Mizu VoIP server can scale well for different hardware configuration from old PIII PC's (for home ussage) up to modern SMP systems with 32 processors (for VoIP carriers)

• Network: ~1 MB for 200 parallel calls without RTP routing or 30 calls with routed RTP (depending on the codec used). Public static IP is preferable with broadband internet connection using Gbit Ethernet card
• CPU: single core PIII for home, testing or small business usage, dual core cpu for less than 500 parallel call, quad core for more than 1000 parallel call
• RAM: minimum 500 MB RAM,1 GB RAM for less than 400 simultaneous calls, 4 GB RAM for more than 1000 simultaneous calls
• Disk: 128 GB HDD without voice recording and callcenter, 512 GB HDD for high load or if you have a long list of callcenter clients or need to use voice recording. Additional disks for big database load (for example one for temp database, another for the mserver database and a third disk for the VoIP application server)
• OS:   Windows Server 2003 or 2008 (Windows XP/Vista/7 for home usage or testing)
• Database: Full or Express versions of MS SQL 2005 or 2008 (included in the install package)
• Dual server configuration: recommended for failowering, load sharing and hot backup
• Carrier connectivity: SIP or H323 protocols with a wide range of codec support

Example configuration for 7000 simultaneous RTP routed calls: 4 core Intel Xeon, 4 GB RAM, 3x128 GB HDD

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 General

•  Standard SQL database backend (MS SQL Express or Full versions)
•  VoIP encryption
•  Virtual servers (multiple separate VoIP server on the same hardware)
•  Multi-homed and multi-domain support
•  Very high throughput
•  Easy maintanance and third party integration

SIP

•  Compliant with SIP rfc's
•  UDP, TCP and TLS transports
•  Proxy server
•  Registrar server
•  Location server
•  Redirect server
•  B2B routing
•  PBX features
•  Transcoding B2BUA
•  Conference Server
•  SBC (Session Border Controller)
•  Routed and Direct voice
•  Automatic NAT detection
•  DID Direct Inward Dialing
•  Voice Recording and Playback
•  Absent Subscriber
•  Abbreviated Dialing
•  Multiple Subscriber Aliases
•  Anonymous Call Rejection
•  Access Control Lists
•  Call Baring Incoming/Outgoing
•  Toll Restriction
•  Parallel Hunting
•  Click 2 Call
•  CLIP/CLIR
•  DTMF generation
•  Call-Forward on out-of-service
•  Codec transcoding
•  Advanced statistics support
•  NAT, STUN/ICE Support (Near-End and Far-End NAT traversal)
•  NAT traversal of signaling
•  NAT traversal of media
•  SIP Session timers
•  RTP Timers and media timeout
•  Blind SIP Registration
•  Late Codec Negotiation
•  Multiple SIP registrations per user account
•  Can act as an SBC
•  Max Session Setting
•  Manage Presence
•  Detailed call logs
•  SIP/SIMPLE
•  SIP Reinvites
•  SIP-H.323 protocol conversion
•  Class 4 features
•  Class 5 features

H.323

•  H.323 Standard Features (v.1,2,3,4)
•  SIP-H.323 protocol conversion (Signaling and media when needed)
•  Full H.323 proxy
•  H.225.0 Call Signaling
•  Fast Connect/Fast Start
•  H.245
•  H245 tunneling
•  H245 in setup
•  DTMF send/receive
•  Watchdog
•  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)

RFC compilance

•  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 3911 The SIP "Join" Header
•  RFC 3324 Network Asserted Identity
•  RFC 3326 The Reason Header Field
•  RFC 3581 Symmetric Response Routing
•  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

Codecs

•  G.723.1
•  G.729
•  G.711 A-law
•  G.711 u-law
•  GSM 06.10
•  GSM
•  Speex 2,3,4,5,6 (narrowband, wideband and ultrawideband)
•  G.726 (16,24,32,40 KHz)
•  G.722
•  T.38
•  DTMF
•  Custom 1 kbits codec
•  All other codecs for pass-trough
•  Voice:
•  Adaptive de-jitter buffer
•  Voice Activity Detection/Silence Suppression
•  Recording conversations  (In Stereo caller/callee left/right)
•  QoS
•  Packet saver technology

IP Centrex

•  Call Forward All/Busy/No Answer
•  Caller ID
•  RingGrouops
•  Call Return
•  Call Waiting
•  Call Forking
•  Call Hold/Retrieve
•  Caller ID Block
•  Selective Caller ID Blocking/Unblocking
•  Speed Dial
•  Direct Inward Dialing (DID)
•  Three-Way Calling, Conference support
•  Message Waiting Indicator
•  Call transfer, Attended transfer, Unattended transfer
•  IVR (all applications: call, callback, p2p, forward, etc)
•  VoiceMail 2 Email
•  DTMF transcoding on server side
•  Interactive Voice Response (IVR) supporting applications such as credit card and prepaid services
•  Video
•  Conference calls
•  T.38 fax relay
•  SMS relay
•  SMS commands (callback, P2P)
•  Web interface

Accounting

•  ACD Features
•  Unlimited accounts / Unlimited Extensions
•  Automatic pincode generation
•  Flexibile authentication (digest,IP,port,user,etc)
•  Unlimited resellers

Routing

•  Custom Routing Rules
•  Multi-Carrier Support
•  ACL
•  Sophisticated configurations
•  Load Balancing on available devices
•  Rerouting
•  Number rewriting (calling and called)
•  Failovering (multiple levels)
•  Least Cost Routing
•  BRS -quality based routing
•  Call Control Features (Maximum Talk Time, Max Ring Time)
•  Call routing based on PLMN tariff packages
•  Blacklist/White list filtering
•  Time of Day Routing
•  Direct Inward Dialing (DID)
•  Route capacity
•  Outbound Dial Map
•  Speed Dial Numbers
•  Auto call forwarding
•  ANI Routing
•  IP Blacklists
•  Custom VoIP Providers
•  Fraud detection tools
•  Support for NAT traversal
•  Automatic capacity rebalancing
•  Remote Linked Servers
•  Automatic channel management
•  Number portability support
•  User authentication by  username/password, IP address, techprefix, callernumber
•  GeoIP database

Billing

•  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
•  Unlimited reseller accounts
•  Callshops
•  Directions (traffic sender,prefix,gateway,sim packet) and time based billing.
•  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.
•  Automatic online currency conversion

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

Management

•  Centralized configuration and management for all software and hardware components
•  MManage:

 -easy to use, mdi style
 -almost every data query is parameterized with traffic direction and time
 -all data in one place
 -global system analysis

•  Create and edit network elements
•  Remote maintenance of Mizu gateways
•  Display of system information
•  Realtime call status view
•  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
•  and more