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Softphone Features
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- 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
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VoIP Server Features
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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
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Links
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Sound quality comparition
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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.
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Compatibility
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