I used

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good one

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vanshu saxena
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Sentinel had written this in response to

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: Ozeki VoIP SIP SDK for .NET is the best tool on the market for software developers working on advanced telephone solutions. With this tool it is very easy to build a high quality software in a short time. Ozeki VoIP SIP SDK is used to create a softphone, build webphone solutions embedded into a webpage, and is a useful tool to make SIP based voice and video applications, and components for contact centers, call centers, CRM systems, IVR systems and IMS solutions. The SDK provides excellent voice and video quality and high performance.

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:

  • G711 Alaw
  • G711 Ulaw
  • G722
  • G729
  • iLBC
  • Speex
  • GSM

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).

MULTI-LINES SUPPORT 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.


  • 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


  • Cisco Unified CM configuration
  • Asterisk configuration
  • 3CX configuration
  • AsteriskNow configuration
  • Kamailio configuration
  • FreeSwitch configuration
  • OpenSIPS configuration
  • SipX ECS configuration
  • Trixbox configuration
  • OpenSER configuration
  • PBXnSIP configuration
  • PBXpress configuration
  • Elastix configuration
  • FreePBX configuration
  • SwyxWare configuration
  • Aastra MX-One configuration

More information at

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or at

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Interesting sdk. Seems to be similar with the mizutech java/javascript webphone

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but more low level. Will try it.

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