Ozeki VoIP SDK - Product Guide
Feature list for Ozeki SIP SDK
Empower your system with the following powerful softphone and webphone related features and just discover how they can make your application and webpages much better!
CAPABLE OF DELIVERING CRYSTAL CLEAR SOUND
When you integrate Ozeki VoIP SIP SDK your application will be capable of delivering crystal clear sound even for both low and high-bandwidth users and SIP compatible devices (hardware and software).
SUPERIOR VOICE QUALITY
Superior voice quality is also guaranteed due to the digital voice processing features.
EXTENDED CODEC SUPPORT
To achieve superior voice quality extended codec support has been included into Ozeki VoIP SIP SDK. Ozeki VoIP SIP SDK for Windows Desktop OS supports for both narrowband and wideband, codecs that's why it works with all type of Internet connections. The following codecs are supported to improve voice quality:
SIP PROXY AUTHENTICATION
Ozeki VoIP SIP SDK allows to register with the SIP proxy server by providing Login ID and Login password.
DIAL/RECEIVE PHONE CALLS
You can dial and receive phone calls through any SIP based server, gateway or Internet Telephony Service Provider (ITSP).
Ozeki VoIP SIP SDK allows to initialize the component with a user-define specific number of lines. You will be free to start the component with 4, 8, 10, 20, 40, 80 or more number of lines. Such feature is used to start conference call, consult call transfer, dial/receive multiple phone calls and for many other purposes
DTMF TONES GENERATION
VoIP SIP allows applications and webpages to generate Dual Tone Multi Frequency (DTMF) tones.
MICROPHONE & SPEAKERS VOLUME
User can control Microphone and Speakers volume directly.
MULTIPLE AND SINGLE CODEC SELECTION SUPPORT
It is possible to select multiple codecs and single codec in Codecs section. In Codecs section you can find the list of available codecs. This function you can also switch between codecs during the conversation.
UDP AND TCP SUPPORT
User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) are supported effectively.
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
- Windows XP
- Microsoft Windows Server 2003
- Microsoft Windows Server 2008
- Microsoft Windows Vista
- Microsoft Windows 7
It connects to a supported VoIP PBX or to a
VoIP service provider
over the Internet. Supports firewall passthrough (STUN/TURN).
REQUIRED .NET FRAMEWORK .NET Framework 3.5 or .NET Framework 4.0 is required for operating Ozeki VoIP SIP SDK effectively.
SUPPORTED PROGRAMMING LANGUAGES
- 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)
Full source code can be purchased. The source code of this VoIP SDK is in C#.Net.
SUPPORTED SIP METHODS
BASIC TELEPHONY AND TELEPHONE FUNCTIONS
- Do Not Disturb(DND)
- Auto answer
- Multiple SIP lines
- Call Ignore
- Call history
- Voice call recording
- Voice conferencing
- PC-PC or PC-phone calls
- Caller ID with name
- Quick calling
- Call history
- Connection to PSTN and mobile phones
Having the above features makes it simple to develop any type of
VoIP-enabled application effectively, like e.g. a SIP softphone. For Ozeki VoIP
SIP SDK clients to interact with each other they must connect to a
SIP gateway or SIP based IP-Telephony service provider. For more information
check the Quick Start Guide.
Do not hesitate to experience the easy and high quality standard Ozeki VoIP SIP SDK that ensures VoIP functionality to your application:
Download Ozeki VoIP SIP SDK - a powerful and highly flexible VoIP SDK to speed up development of VoIP solutions
In case you have any question, please do not hesitate to contact us at firstname.lastname@example.org
Read more about Ozeki VoIP SIP SDK:
Quick Start Guide
What makes Ozeki VoIP SIP SDK better than all the rest
Learn how to VoIP-enable your application