What are the functions of a VoIP server What are other names for a VoIP server?

The Mizu VoIP server is a feature rich softswitch fulfilling the needs of both small business and enterprise carriers. The most important capabilities are listed below:

Transport

  • Transport protocols: UDP, TCP, HTTP (clear text, XML, JSON, SOAP, RDF), websocket
  • Encryption: HTTPS, TLS, DTLS, SRTP, VPN, custom RSA based
  • VoIP protocols: SIP/SIPS, H.323, WebRTC, RTMP

SIP

  • Both old and new SIP rfc's are supported
  • SIP proxy
  • SIP registrar
  • Routed and Direct voice (RTP proxy and offload)
  • 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
  • RFC 8599 Push Notification with the SIP
  • RFC 7118 The WebSocket Protocol as a Transport for SIP
  • 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
  • SIP-WebRTC protocol conversion
  • Custom protocol extensions are possible
  • Many other SIP related RFC's

WebRTC

  • Transport protocols: UDP/TCP/HTTP; DTLS/TLS/HTTPS; Websockets (WS, WSS)
  • RFC 7118 support
  • Compatible with all popular webrtc clients such as sipjs and sipml5
  • Built-in ICE, STUN and TURN
  • Auto RTP offload or proxy
  • Auto codec conversion when necessary

H.323

  • 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
  • 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)

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
  • Opus
  • T.38
  • DTMF
  • Voice:
  • Adaptive de-jitter buffer
  • Voice Activity Detection/Silence Suppression
  • Recording conversations
  • QoS
  • Packet saver technology

Class 5 PBX Features

  • Call Forward All/Busy/No Answer
  • Caller ID
  • Ring Groups
  • 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 (Interactive Voice Response supporting applications such as credit card and prepaid services)
  • Conference (3 way SIP, on demand, conference rooms)
  • Voicemail (WMI, auto email forward)
  • DTMF transcoding on server side
  • Video
  • T.38 fax relay
  • Click to call
  • Call me
  • Offline chat
  • And many other built-in PBX features

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, 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
  • Groups

Routing

  • Multi-Carrier Support
  • ACL
  • Sophisticated configurations
  • Load Balancing
  • 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
  • Push notification support for mobile and web clients

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
  • Dashboard
  • Create and edit network elements
  • Gateways remote maintenance
  • 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

Other modules

  • Call-back
  • Calling card
  • Web control panel for users
  • Resellers (unlimited levels)
  • VoIP tunneling and encryption
  • Supervisor
  • Text to speech (TTS)
  • SMS via SMPP, SIP MESSAGE or HTTP(S) GET/POST with any payload (JSON, clear text, XML, etc)

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

Links
What are the functions of a VoIP server What are other names for a VoIP server?

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)
  • 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
  • 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/SPA1001
  • Linksys/SPA2102
  • Linksys/SPA9000
  • Linksys/SPA922
  • Linksys/SPA942
  • 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

What are the functions of a VoIP server?

A VoIP server is used to connect calls to other telephone networks. As long as you have a high-speed internet connection, which includes a router and modem, you are ready to use a VoIP. A typical VoIP configuration involves a desk phone and a SIP server, which is typically a VoIP service provider.

What are other names for VoIP server?

VoIP is the abbreviation of the term: Voice over Internet Protocol. This technology is also known under other names: IP Phone, IP Telephony, Broadband Telephony, Internet Telephony etc.

What does VoIP also known as?

Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet.

What are examples of VoIP?

Here are the most common examples of VoIP apps:.
Nextiva..
Aircall..
Zoiper..
Skype..
WhatsApp..
Google Hangouts..
Viber..
Facebook Messenger..