• Open Standards based next generation telephony client
• SIP Compliant VOIP calls
• Transport protocolls: UDP, TCP, TLS, tunneling
• Registrar Support
• Proxy Support
• Outbound Proxy Support
• Call Mute
• Call Hold
• Call Transfer
• Call Forwarding
• Conference Calls (with local mixer and codec conversion when necessary)
• Click to Talk
• Callback
• P2P calls (Phone to Phone)
• VoiceMail (remote and local)
• Send SMS from the softphone (your provider must support it)
• Redial
• Dialpad
• Find-Me
• Speed Dials
• Devices Auto-Configuration (network, audio, video)
• Smooth operation under not voip friendly conditions (low bandwidth, packet loss, NAT, firewall, etc)
• Configurable Sound Events
• SIP re-INVITE and UPDATE support
• Configurable Port Ranges
• Auto-find people near me
• Message Waiting Indications Support
• Multiple accounts and multiple SIP server registrations.
• Multiple incoming/outgoing calls simultaneously
• HD quality video calls (depending on your camera and bandwidth)
• Full screen directx based video
• Remote Webcam viewing
• Full-Screen Video Conferencing
• Instant messaging and presence using the SIMPLE protocol
• Session timers
• Network diagnostics
• Forked requests
• Auto Answer and Do Not Disturb Modes
• File transfer (compatibile with any SIP server)
• File sharing (compatibile with any SIP server)
• Remote Desktop over SIP
• Fax (beta version)
• Call and Chat History
• Audio and video recording
• Audio Codecs: G.711-Alaw, G.711-uLaw, G.723.1, G729, iLBC, L16, Speex
• Video Codecs: MPEG1, MPEG4, Theora, DIV3, MJPG, H263, H264
• WideBand and Ultra WideBand codec (speex)
• Audio tuning wizard
• Dynamic Jitter Buffer
• Packet loss concealment (PLC)
• Automatic Gain Control  (AGC)
• Acoustic Echo Cancellation (AEC)
• Voice activity detection (VAD)
• Noise supression
• Auto QoS
• Dynamic Threshold Algorithm for Silence Detection
• Network handling: UPNP, STUN, ICE, IP Translation, Firewall and NAT detection
• DTMF (Inband DTMF or SIP INFO messages)
• CRM solution: Click to Talk
• Local signaling (Dial tone, busy, ring back, etc.) for user comfort
• Call timer
• Softphone Configuration Wizard
• DNS support
• Balance/credit display
• Personal address book
• Remote profile storage WebDav, XCAP, FTP, HTTP
• Microsoft Outlook synchronization
• Import contactlist from various sources (LDAP,WAB,Outlook,CSV,Active Directory, etc)
• Settings and contactlist backup and restore
• Full encypted communications (protocoll and media too)
• Intelligent P2P based network path detection (will work even if the server is down)
• Not using any .NET and Java Runtime Library
• Customizable interface and language
• Free profile storage
• Free sip proxy/registrar service

and more

Implemented RFC’s and Drafts

• RFC 2543 The old SIP Core Protocol
• RFC 3261 The new SIP Core Protocol
• RFC 3262 Reliability of Provisional Responses in Session Initiation
• RFC 2976 The SIP INFO Method
• RFC 2617 HTTP Authentication
• RFC 3891 Replaces Header
• RFC 3892 The SIP Referred-By Mechanism
• RFC 3325 Private Extensions to the Session Initiation
• RFC 2778 A Model for Presence and Instant Messaging
• RFC 3863 Presence Information Data Format (PIDF)
• RFC 4480 RPID: Rich Presence Extensions to PIDF
• RFC 4482 CIPID: Contact Information in PIDF
• RFC 3856 A Presence Event Package for SIP
• RFC 2387 The MIME Multipart/Related Content-type
• RFC 3856 A Presence Event Package for SIP
• RFC 4479 A Data Model for Presence
• RFC 2779 Instant Messaging / Presence Protocol Requirements
• RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging
• 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 4353 A Framework for Conferencing with SIP
• RFC 4579 SIP Call Control - Conferencing for User Agents
• RFC 4597 Conferencing Scenarios
• RFC 3911 The SIP Join Header
• RFC 3581 Symmetric Response Routing
• RFC 3324 Short Term Requirements for Network Asserted Identity
• RFC 3325 Private Extensions to SIP for Asserted Identity within Trusted Networks
• RFC 3323 A Privacy Mechanism for SIP
• RFC 4189 Requirements for End-to-Middle Security for SIP
• RFC 3842 Message Summary and Message Waiting Indication Event Package
• RFC 1889 RTP: A Transport for Real-Time Applications
• RFC 2190 RTP Payload Format for H.263 Video Streams
• 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 3960 Early Media and Ringing Tone Generation in SIP
• RFC 4028 Session Timers in SIP
• RFC 3824 Using E.164 numbers with SIP
• RFC3903 PUBLISH method
• RFC 3966 The tel URI for Telephone Numbers
• RFC 4145 TCP-Based Media Transport in SIP
• RFC 2663 IP Network Address Translator (NAT) Terminology and Considerations
• RFC 3022 Traditional IP Network Address Translator (Traditional NAT)
• RFC 3489 STUN - Simple Traversal of UDP through NATs
• 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
• draft-ietf-sipping-nat-scenarios
• Custom protocol extensions are possible

Transport

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

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
• 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)

SIP

• 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
• 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

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

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 (Interactive Voice Response supporting applications such as credit card and prepaid services)
• Conference
• Voicemail
• DTMF transcoding on server side
• Video
• T.38 fax relay

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

Accounting

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

Routing

• 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

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.

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
• 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
• VoIP tunnel
• Supervisor

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