VaxVoIP SIP activeX SDK V3.0

VaxVoIP SIP activeX SDK 3.0 Description

" VaxVoIP SIP activeX SDK : VaxVoIP SIP SDK provides tools and components to quickly add SIP ( Session Initiation Protocol ) based IP-Telephony make and receive phone calls feature in your web pages and software applications. You can easily develope net-to-phone apps web pages. . "

Edit by 1ware.net: VaxVoIP SIP activeX SDK - VaxVoIP SIP SDK provides tools and components to quickly add SIP ( Session Initiation Protocol ) based IP-Telephony make and receive phone calls feature in your web pages and software applications. It accelerates the development of SIP based soft phone with your own GUI ( graphical user interface ) and brand name.
It delivers superior voice quality by integrating advanced digital voice processing features including acoustic echo cancellation, noise cancellation and adaptive jitter buffering.
In order to eliminate the acoustic feedback an echo canceller is introduced in the VaxVoIP SIP SDK.

-ACOUSTIC ECHO CANCELLATION OR SUPPRESSION
In order to eliminate the acoustic feedback an echo canceller is introduced in the VaxVoIP SIP SDK. Hands-free or Internet telephony imposes several problems. The principal one is due to the coupling between loudspeaker and microphone. The loudspeaker signal is echoed back to the microphone and transmitted back to its origin. As a result the far-end participant perceives this as an echo.

-NOISE CANCELLATION OR SUPPRESSION:
VaxVoIP SIP SDK offers advanced Noise Cancellation technology that allows significant suppression of any background noise and provides high quality of output speech.

-ADAPTIVE JITTER BUFFER
Jitter buffers are used to smooth delay variations in received audio by buffering the packets and adjusting their rendering. The result is a smoother delivery of audio to the user.

-PACKET LOSS CONCEALMENT
Packet Loss Concealment ( PLC ) is a technique used to mask the effects of lost or discarded packets. PLC is generally effective only for small numbers of consecutive lost packets, for example a total of 20-30 milliseconds of speech, and for low packet loss rates.

-NAT AND FIREWALLS FRIENDLY
User can set SIP outbound proxy inorder to make and receive phone calls behind the NAT / firewall.