SIP SDK 3.6

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I used <a href="www.SipVoipSdk.com">www.SipVoipSdk.com</a>, good one

Ozeki VoIP SIP SDK
Sentinel had written this in response to
http://forums.cabling-design.com/videoconference/SIP-SDK-3-6-774-.htm  :
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.

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

OUTSTANDING PBX COMPATIBILITY
* 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 http://www.voip-sip-sdk.com/ or at
info@voip-sip-sdk.com

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Re: SIP SDK 3.6
Interesting sdk. Seems to be similar with the mizutech java/javascript webphone http://www.mizu-voip.com/Software/WebPhone.aspx but more low level. Will try it.

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