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 Softphone Features Minimize

 

  • 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

    
 VoIP Server Features Minimize

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)

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

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

IP

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

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

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

Routing

  • Multi-Carrier Support
  • ACL
  • Sophisticated configurations
  • Load Balancing on available GSM channels and any other devices
  • 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
  • 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!

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

    
 Links Minimize

  
 Sound quality comparition Minimize

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

 

Highlights:

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

  
 Compatibility Minimize

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
  • Asterisk PBX
  • Audiocodes-Sip-Gateway-MP-114 FXS/v.4.80A.025.004
  • Audiocodes-Sip-Gateway-MP-118 FXS/v.4.80A.014.006
  • Audiocodes-Sip-Gateway-MP-118 FXS/v.4.80A.043.005
  • Audiocodes-Sip-Gateway-MP-118 FXS/v.5.20A.039.002
  • AVM FRITZ!Box Fon (EU300) 15.04.01 (Jan 25 2006)
  • AVM FRITZ!Box Fon (fs) 06.04.33 (May 10 2007)
  • AVM FRITZ!Box Fon WLAN 7170 29.04.44 (Nov  1 2007
  • AVM FRITZ!Box Fon WLAN 7170 29.04.47 (Nov 28 2007
  • BVA8052D (LDTK AR18D ) STUN 0 0 0
  • Cisco ATA 188  v3.1.1 atasip (040813T)
  • Cisco IP Phones
  • Cisco-SIPGateway/IOS-12.x
  • CM5K  (610140)
  • CM5K  (706220)
  • 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-1.1.5
  • DrayTek UA-1.2.1
  • DrayTek UA-1.2.1 Vigor2200V series
  • DrayTek UA-1.2.3 DrayTek Vigor2910
  • DrayTek UA-1.2.3 Vigor2200V series v2.6.1_(5)
  • DrayTek V3300V-1.0.0
  • Draytel
  • ETK-MP-114FXS/v.4.80A.014.006
  • Ekiga
  • Express Talk 2.02
  • Evolutiontel
  • fring
  • FWD
  • Gizmo5
  • Gizmo Project
  • Grandstream BT100 1.0.6.7
  • Grandstream BT120 1.0.8.33
  • Grandstream GXP2000 1.0.1.9
  • Grandstream GXP2000 1.1.1.14
  • Grandstream GXP2000 1.1.4.18
  • Grandstream GXP2000 1.1.5.15
  • Grandstream GXP2000 1.1.6.16
  • Grandstream HT488 1.0.2.16
  • Grandstream HT488 1.0.3.18
  • Broadvoice
  • IP Office 4.0
  • Kapanga
  • KPhone
  • Linksys/PAP2-2.0.12(LS)
  • Linksys/PAP2-3.1.12(LS)
  • Linksys/PAP2-3.1.22(LS)
  • Linksys/PAP2-3.1.3(LS)
  • Linksys/PAP2-3.1.9(LSc)
  • Linksys/PAP2T-3.1.15(LS)
  • Linksys/RT31P2-3.1.7(LIa)
  • Linksys/RT31P2-3.1.9(LId)
  • Linksys/SPA1001-3.1.19(SE)
  • Linksys/SPA2102-3.2.7(b)
  • Linksys/SPA2102-3.3.6
  • Linksys/SPA9000-3.3.6
  • Linksys/SPA922-4.1.18
  • Linksys/SPA922-5.1.15(a)
  • Linksys/SPA942-4.1.18
  • Linksys/WRP400-1.00.04.c
  • LR SIP Phone
  • M1000/v.4.80A.025.004
  • Minipax
  • MSC/VR40
  • NCH Swift Sound Express Talk 3.01
  • PA168S V1.49.002 CFG0
  • PA168S V1.53.006 CFG0
  • PA168S V1.55.013 CFG0
  • phone V1.49.002 CFG0
  • Pidgin
  • PortaBilling
  • RTP300-3.1.17
  • sa210
  • Sipgate
  • SIPPER for 3CX Phone
  • Sipura/SPA1001-2.0.13(SEg)
  • Sipura/SPA1001-3.1.8(SEc)
  • Sipura/SPA2100-2.0.5(c)
  • Sipura/SPA2100-3.2.2(g)
  • Sipura/SPA2100-2.0.5(d)
  • Sipura/SPA2100-3.2.5(d)
  • Sipura/SPA3000-2.0.13(GWg)
  • Sipura/SPA841-3.1.4(a)
  • Sipura/SPA941-4.1.8
  • SJphone/1.60.289a (SJ Labs)
  • StarTel
  • TeloniaSIP/3.0.1
  • TrixBox
  • UTSTARCOM F1000/Device ID-0007ba25de51
  • VOIP_Agent_001
  • VoipBuster
  • Voipdiscount
  • VoipGate
  • Voipswitch
  • WengoPhone
  • WRTP54G-3.1.17
  • VoipBuster
  • Vonage
  • Zoiper
  • X-Lite release 1002tx stamp 29712
  • X-Lite release 1011s stamp 41150
  • X-Lite release 1103m
  • X-Lite release 1105x
  • XytonSIP/3.0.4
  • XytonTAgent/3.1.42

and many others