SMS Gateway software
Ozeki brings you outstanding
SMS Gateway technology. Use our SMS Server products on Windows,Linux, or Android

Developers can use our C# SMS API to send SMS from C#.Net. The C# SMS API comes with full source code

The ozeki PHP SMS gateway software can be used to send SMS from PHP and to receive SMS usig PHP on your website

SMPP SMS Gateway
SMS service providers use our SMPP gateway solution, that offers a high performance SMPP server and SMPP client gateway with amazing routing capabilities


Ozeki VoIP SDK data sheet

Download Data Sheet:OzekiVoIPSDK_DataSheet.pdf
Ozeki VoIP SIP SDK Product information
Product name Ozeki VoIP SIP SDK
Category Software Development Kit
Product website http://www.voip-sip-sdk.com
Latest versionOzeki SDK 1.8.18
Release date2018.03.08.
Package size138.3 MB
Download urlhttp://www.voip-sip-sdk.com/p_21-download.html
Package contents
  • Redistributable .DLL
  • Documentation
  • Example applications
  • Exe demo
  • Full source code (optional)
Main task Makes it possible to build a VoIP client application or a PBX, based on the SIP protocol.
Connectivity It connects to a supported VoIP PBX or to a VoIP service provider over the Internet. Supports firewall passthrough (STUN/TURN).
Supported client OS
  • Windows server 2008
  • Windows server 2012
  • Windows server 2016
  • Windows Vista
  • Windows 7
  • Windows 8
  • Windows 10
  • Windows 11
Required .NET
At least .NET Framework 4.5.2 or any newer version
languages and
Microsoft Visual Studio 2012, 2013, 2015, 2017, 2019 (C#, VB.NET, ASP.NET,...)
Source code Full source code can be purchased. The source code of this VoIP SDK is in C#.Net.
Developer features
  • Easy to use
  • Very easy to incorporate
  • Makes quick development possible
  • Supports all development environments with .NET support
  • Supports the development of WPF, Windows Form, Windows Service application, etc.
  • Free product version updates: one year free updates
Call features
  • Multiple simultaneous call support
  • Make and receive Audio/Video calls
  • Make and receive peer-to-peer calls without a SIP server
  • Call reject
  • Hold/Unhold
  • Call forward
  • Call transfer (Attended and Blind transfer)
  • DTMF support: send/receive DTMF signals via RFC 2833, SIP INFO or inband
  • Do Not Disturb(DND)
  • Auto Answer
  • Redial
  • Secure calls (TLS, SRTP)
  • Codec priority change
  • Caller ID modification for outbound calls
  • Call history
  • Multi-party voice conference support (Conference split/join, locally mixed conferences)
SIP features
  • Multiple SIP account support
  • Secure SIP connection via TLS
  • Message Waiting Indicator (for checking voicemail)
  • Send and receive instant messages
  • Digest authentication
  • Direct access to incoming and outgoing SIP messages (add/modify SIP headers for inspect or repair)
  • Multipart SIP body handling
  • Outbound proxy server support
  • Subscription to SIP event packages
PBX features
  • Fully customized dial plan creation (call and message routing)
  • Custom extension or custom connection creation (eg. voicemail, SIP trunk, echo test)
  • Rich call information and call event notification
  • Third-party call control (forward, hold, transfer, hangup)
  • Direct access to caller and callee RTP stream outside of a call
  • Authentication control
  • Music on hold
Network features
  • Multiple network interface support
  • Supported protocols: UDP, TCP, TLS, SIP, SDP, RTP, SRTP, STUN, TURN, ICE
  • Configurable port range
  • Firewall/NAT passthrough (auto discovery, STUN, static IP setting)
Audio features
  • Microphone & Speaker device selection (on-the-fly as well during a conversation/conference)
  • Device calibration (volume, level, mute, device change, format change)
  • Play wav or mp3 files to remote party
  • Record audio in wav or mp3 format
  • Text-To-Speech support (changing voice, setting speech rate, multiple TTS engines)
  • Speech-To-Text/Speech recognizer support (changing voice, multiple STT engines)
  • Play DTMF tones
  • Play audio from multiple audio sources to remote party
  • HD audio support (HD audio calls)
  • Supports most audio formats (8000-48000 Hz, 16bit, mono/stereo)
  • Automatic audio format conversion
  • Support access incoming and outgoing audio stream directly
  • Adaptive jitter buffer
  • Packet loss concealment
Video features
  • Camera device selection (on-the-fly as well during a conversation/conference)
  • Device calibration (device change, resolution/frame rate change)
  • Play video files to remote party (mp4)
  • Record video in mp4 format
  • 3D video support
  • Real-time video quality change
  • Picture manipulation (rotate, flip)
  • 720p, SVGA, XVGA, VGA, CIF, QCIF video resolutions
  • Support access incoming and outgoing video stream directly
Advanced Digital Signal Processor features
  • Auto Gain Control (AGC)
  • Noise Reduction
  • Voice Activity Detection (VAD)
  • Acoustic Echo Cancellation (AEC)
  • Answer Machine Detection (predictive dialer)
Supported audio codecs
  • PCMA (G.711 aLaw)
  • PCMU (G.711 uLaw)
  • GSM
  • iLBC (20 and 30 ms)
  • Speex Narrowband (8kHz)
  • Speex Wideband (16kHz)
  • Speex Ultrawideband (32kHz)
  • G.722
  • G.723
  • G.726-16
  • G.726-24
  • G.726-32
  • G.726-40
  • G.728
  • G.729
  • L16
  • OPUS
Supported video codecs
  • H.264
  • H.263-1998
  • H.263
Fields of application
  • Softphones
  • Webphones
  • Online chat communities (e.g.: dating, business meetings)
  • PBX servers
  • Automatic call distributors
  • Telephone and call centers
  • VoIP providers
  • Conferencing applications
Supported PBX systems
Example applications

        More information