High performance VoIP SDK for .Net developers

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 VoIP SIP SDK v10.0.9
Release date2013.03.20.
Package size89.2 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 VoIP client software 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 XP, 2003, 2008, Vista, 7
Required .NET framework .NET Framework 3.5 or .NET Framework 4.0
Supported programming languages and environments Microsoft Visual Studio 2003,2005,2008,2010 (C#, VB.NET, J#, ASP.NET,...)
MS Visual Studio 6(VC6, VB6, ...)
Borland C++ 5/6/7
Borland Delphi 6/7
CodeGear Delphi 2007
CodeGear C++ Builder 2007
Web technologies (ActiveX)
Source code Full source code can be purchased. The source code of this VoIP SDK is in C#.Net.
Basic Telephony and telephone functions Hold, Forward, Transfer, Do Not Disturb(DND), Auto answer, Redial, Multiple SIP lines, Call Ignore, Call history, Voice call recording, Voice conferencing, DTMF, Video Call.

PC-PC or PC-phone calls, Caller ID with name, Quick calling and re-dialing, Call history, Connect to PSTN and mobile phones.
Comprehensive configuration support
  • Select media input/output devices (on-the-fly as well during a conversation/conference)
  • Configurable ports (RTP, SIP UDP, SIP TCP, STUN, TURN, ICE) 
  • SIP proxy
Advanced digital voice processing features
  • AGC (auto gain controller)
  • AES (Acoustic echo cancellation or suppression)
  • Noise cancellation or suppression
  • Voice Activation Detection
Fields of application
  • Softphones
  • Webphones
  • Online Chat Communities (e.g.: dating, business meetings)
  • VoIP Providers
  • VoIP Devices
  • Conferencing Applications
Audio file play and record YES (Supports .wav and .mp3 files)
Audio format can be:
  • 8K 16bit mono PCM
  • 8k 8bit mono mulaw/alaw
Supported SIP Methods REGISTER, INVITE, CANCEL, INFO, BYE, ACK, SUBSCRIBE, OPTIONS.
Authentication HTTP authentication (BASIC and Digest).
RTP Package Access Support access incoming and outgoing RTP audio stream directly. And support change RTP audio stream to integrate TTS and ASR engine.
Extensions
  • Subscription to SIP event packages during a specific all SIP preprocessor functionality-inspection
  • Access the incoming audio stream directly
  • Send the PCM stream directly to instead of microphone input
  • Access the incoming SIP message and SIP message header directly
  • Add/modify the SIP message headers
Easy, familiar, event-driven call control
  • Easy to use; quick development
  • Support for all development environments with .Net support
  • Very easy to incorporate
Rich call control feature set
  • Multi-party voice conference support (Conference split/ join, locally mixed conferences)
  • Multi-line support (multiple simultaneous calls)
  • Multiple lines for multiple concurrent calls 
  • Locally mixed conferences
  • Hold/Mute 
  • Call transfer 
  • Call forwarding and rejection
Audio features
  • Adaptive jitter buffer
  • Packet loss concealment for voice
  • Automatic Gain Control (AGC) for voice
  • Voice Activity Detection (VAD)
  • Predictive dialing (answering machine detection)
  • Acoustic Echo Cancellation (AEC)
  • Narrow band and wide-band voice codec choice: G711A, G711U, iLBC, Speex, Speex-wb, GSM, G.729a, L16, G.723, G.726-16, G.726-24, G.726-32, G.726-40, G.728
Video features
  • HD video phoning
  • jitter buffer
  • video codecs: H.263, H.264
  • picture rotate, flip
  • 720p, svga, xvga, vga, cif, QCIF video resolution
Supported PBX systems
  • Cisco Unified CM PBX
  • Cisco Call Manager Express PBX
  • Asterisk PBX
  • 3CX PBX
  • AsteriskNow PBX
  • Kamailio PBX
  • FreeSwitch PBX
  • OpenSIPS PBX
  • SipX ECS PBX
  • Trixbox PBX
  • OpenSER PBX
  • PBXnSIP PBX
  • PBXpress PBX
  • Elastix PBX
  • FreePBX PBX
  • SwyxWare PBX
  • Aastra MX-One PBX
  • Example applications
  • C# WPF softphone
  • Windows Forms Softphone Sample
  • Windows Forms Softphone VB NET
  • Visual C++.Net Softphone
  • Silverlight Video Chat Example
  • SL Mediagateway example
  • MediaGateway SIP Example
  • Flash SIP Client Example
  • MediaGateway SDK Flash chat example
  • WinForms DTMF IVR
  • VB.NET DTMF IVR
  • C# DTMF IVR
  • C# Voice recognition IVR
  • C# Autodialer example
  • C# Speech to text
  • C# Voice conference room
  • C# Command line caller
  • SIP SMS Example
  • C# Callback Form
  • ASP VoIP example
  • Standards
    Features and Specifications
    • Audio call: G.711 aLaw/uLaw, G.729(b), iLBC, GSM, G.722, SPEEX, SPEEX-WB.
    • Call hold, mute speaker, mute microphone
    • Do not disturb(DND), Auto answer(AA)
    • Audio record: record audio as wave file
    • Support accessing incoming audio stream directly
    • Support accessing incoming SIP message directly
    • Support playing wave file to remote side
    • Support adding custom SIP header
    • Support SIP header modification
    • Audio conferencing
    • Message waiting Indicator(MWI)
    • Authentication: HTTP Basic, Digest Authentication
    • DTMF support: Send DTMF tone(RFC2833), detect DTMF tone(RFC2833)
    • Multiple Call
    • Microphone & Speaker Device Selector
    • Microphone & Speaker Volume control
    • Acoustic Echo Cancellation
    • Automatic gain control
    • Comfort Noise Generation
    • Voice Activity Detector
    • STUN/TURN support
    • Outbound proxy server support
    • Jitter buffer
    • Free product version updates: one year free updates.
    • Support develop WPF, Windows Form, Windows Service.
    • application, etc.
    • Call forward
    • Full HD video call