Did you know?

This SDK was used to build:

Ozeki Phone System XE - VoIP PBX Software for Developers Which is a high performance PBX system supporting Mobile and Desktop phones.

It was also used to create Ozeki 3D VoIP softphone. A cool SIP client that allows 3D Video calls.

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 version	OZEKI SDK v1.6.0
Release date	2016.11.07.
Package size	165.9 MB
Download url	http://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 2003 Windows server 2008 Windows server 2012 Windows XP Windows Vista Windows 7 Windows 8 Windows 10
Required .NET framework	At least .NET Framework 3.5 or any newer version
Supported programming languages and environments	Microsoft Visual Studio 2008, 2010, 2012, 2013 (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
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	Ozeki Phone System XE Ozeki Phone System XE PBX Cisco Unified CM PBX Cisco Call Manager Express PBX Asterisk PBX 3CX PBX AsteriskNow PBX Kamailio PBX FreeSwitch PBX OpenSIPS PBX Trixbox PBX OpenSER PBX PBXnSIP PBX PBXpress PBX SipX ECS PBX Elastix PBX FreePBX PBX SwyxWare PBX Aastra MX-One PBX
Example applications	Silverlight Video Chat SIP MediaGateway Flash SIP client example Flash chat example WinForms DTMF IVR VB.NET DTMF IVR C# DTMF IVR C# Voice recognition IVR
Standards	RFC 2833:RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals RFC 3261:Session Initiation Protocol RFC 3263:SIP: Locating SIP Servers RFC 3264:An Offer/Answer Model with the (SDP) RFC 3265:SIP Event Notification RFC 3420:Internet Media Type message/sipfrag RFC 3428:SIP Instant Messaging RFC 3489:STUN - Simple Traversal of UDP Through NATs RFC 3515:SIP Refer Method RFC 3550:Real-time Transport Protocol RFC 3551:RTP Audio/Video Conference RFC 3587:IPv6 Global Unicast RFC 3666:SIP, PSTN, Call Flows RFC 3725:Best Practices for Call Control RFC 3842:Message Waiting Indication RFC 3856:Presence Events in SIP RFC 3891:The SIP Replaces Header RFC 3892:SIP Referred-By Mechanism RFC 3920:Extensible Messaging and Presence Protocol (XMPP): Core RFC 4566:Session Description Protocol RFC 5411:A Hitchhiker's Guide to the SIP

Package Contents

Installation

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Ozeki VoIP SDK - Product Guide

Did you know?

Ozeki VoIP SDK data sheet