- 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.