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SMS Gateway software
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SMS Gateway technology. Use our SMS Server products on Windows,Linux, or Android

C# SMS API
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Ozeki VOIP SIP SDK

 

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
framework
At least .NET Framework 4.5.2 or any newer version
Supported
programming
languages and
environments
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
  • Supported SIP methods: REGISTER, INVITE, ACK, CANCEL, OPTIONS, BYE, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
  • 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
        Standards

        More information