Mizu VOIP Server 3.2.14

Mizu VOIP Server 3.2.14 offers a variety of flexible and practical features of an all in one enterprise grade softswitch solution which supports to SIP and H.323. All Class5 features including sophisticated routing, access rules, CDR record generation, billing and accounting and offers full compatibility with your SIP infrastructure.

The software provides:

• Rich featured: H323, all sip related protocols, SIP-H323 conversion, class5 features, IVR, call conference, call forward, call transfer, billing, dtmf and codec transcoding, etc.
• Best ROI: Mizu server has very simple licensing. with the unlimited license plan, you dont have to worry anymore on the number of ports that you plan to use
• High call density: one Mizu server can replace several hardware based expensive solution, making real benefits for your company
• Robustness: the Mizu sofswitch are currently deployed for several VOIP Service Provider and it runs months or years without problems. You have multiple backup solutions to use.
• Intelligent RTP routing: lots of media traffic can flow directly between the endpoints that have to be routed with other sip proxy solutions.
• Embedded firewall and automatic DOS attack prevention
• Easy to manage with all in one management software running on windows desktop
• Feature Reach: IP Centrex style services can be easily handled with Class5 features for your customers
• Sophisticated routing: failowering, rerouting, load balancing, LCR and blacklist filtering.
• Ease of use: with the MizuManage client program you can administrate all components in a highly intuitive manner. Most of the settings will be set automatically for you.

Major Features:

  1. H323

• 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
• 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)

  2. SIP

• Both old and new SIP rfc's are supported
• SIP proxy
• SIP register
• Routed and Direct voice
• 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
• 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
• Signaling and media when needed

  3. Codecs

• 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
• T.38
• DTMF
• Voice:
• Adaptive de-jitter buffer
• Voice Activity Detection/Silence Suppression
• Recording conversations
• QoS
• Packet saver technology

  4. IP

• Ethernet 10/100 Base-T
• Static IP
• PPPoE (DSL or cable modem)
• DialUpISDN
• VPN
• Encrypted communication

  5. Class 5 Features

• Call Forward All/Busy/No Answer
• Caller ID
• RingGrouops
• 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
• Voicemail
• DTMF transcoding on server side
• Interactive Voice Response (IVR) supporting applications such as credit card and prepaid services
• Video
• T.38 fax relay

  6. 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, voice and faxmail, 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

  7. Accounting

• Unlimited accounts
• Automatic pincode generation
• Flexible authentication

  8. Routing

• Multi-Carrier Support
• ACL
• Sophisticated configurations
• Load Balancing on available GSM channels and any other devices
• Rerouting
• Number rewriting (calling and called)
• Fail-overing (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

  9. 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
• 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.

  10. Management

• 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
• Create and edit network elements
• Remote maintenance of Tresto gateways
• 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!

  11. 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

Enhancements: SIP Conferencing and IVR

Requirements:

1. Intel Pentium II 500 MHz
2. 128 MB RAM
3. 30 MB free space

WareSeeker Editor