• 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, GSM, iLBC, L16, Speex, OPUS
• Video Codecs: MPEG1, MPEG4, Theora, DIV3, MJPG, H263, H264, VP8
• WideBand and Ultra WideBand codec (speex, opus)
• Audio tuning wizard
• Dynamic Jitter Buffer
• Packet loss concealment (PLC)
• Automatic Gain Control  (AGC)
• Acoustic Echo Cancellation (AEC)
• Voice activity detection (VAD)
• Noise supression
• Push notifications
• 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

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