softphone Features

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

Sound quality comparition

voice recorded with a low-cost sound card (found in most PC's and laptops)

Highlights:

  • MizuPhone was the first business class sip softphone with ultra wideband codec.
  • MizuPhone was the first business class sip softphone with file transfer over basic sip protocol.

Compatibility

This list contain only devices tested by us. Our users are using Mizu with other SIP devices too.

  • ALL7950 02.09.31
  • ALL7950 02.09.33
  • Adore Softphone
  • Alcatel
  • Asterisk PBX
  • Audiocodes-Sip-Gateway-MP-114
  • Audiocodes-Sip-Gateway-MP-118
  • Avaya
  • AVM FRITZ!Box Fon (EU300)
  • AVM FRITZ!Box Fon (fs)
  • AVM FRITZ!Box Fon WLAN 7170
  • BVA8052D (LDTK AR18D ) STUN 0 0 0
  • Broadsoft
  • Cisco ATA 188
  • Cisco IP Phones
  • Cisco-SIPGateway/IOS-12.x
  • CM5K  (610140)
  • CounterPath Bria
  • CounterPath eyeBeam
  • CounterPath X-Lite
  • D-Link/DVG-1402S-1.00.009EU
  • D-Link/DVG-G1402S-1.00.009EU
  • dlink 12-37-5381895-0.8.21.1
  • dlink/dph300s
  • DrayTek UA
  • DrayTek UA-1.2.1 Vigor2200V series
  • DrayTek UA-1.2.3 DrayTek Vigor2910
  • DrayTek V3300V
  • Draytel
  • ETK-MP-114FXS
  • Ekiga
  • Express Talk 2.02
  • Evolutiontel
  • fring
  • FWD
  • Gizmo5
  • Gizmo Project
  • Grandstream BT100
  • Grandstream BT120
  • Grandstream GXP2000
  • Grandstream HT488
  • Broadvoice
  • IP Office 4.0
  • Kapanga
  • KPhone
  • Linksys/PAP2
  • Linksys/PAP2T
  • Linksys/RT31P2
  • Linksys/SPA_All
  • Linksys/WRP400
  • LR SIP Phone
  • M1000/v.4.80A.025.004
  • Minipax
  • Mitel
  • MSC/VR40
  • NEC
  • NCH Swift Sound Express Talk
  • PA168S
  • Pidgin
  • pjsip
  • PortaBilling
  • RTP300-3.1.17
  • sa210
  • SipDroid
  • Sipgate
  • SIPPER for 3CX Phone
  • Sipura/SPA1001
  • Sipura/SPA2100
  • Sipura/SPA3000
  • Sipura/SPA841
  • SJphone
  • StarTel
  • TeloniaSIP/3.0.1
  • TrixBox
  • UTSTARCOM
  • VOIP_Agent
  • VoipBuster
  • Voipdiscount
  • VoipGate
  • Voipswitch
  • WengoPhone
  • WRTP54G
  • VoipBuster
  • Vonage
  • Zoiper
  • X-Lite

and many others