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Adaptive Digital Rolls Out Voice Engine/SIP SDK for VoIP Application Developers

SIP Trunking

Enterprise VoIP Featured Article

April 15, 2013

Adaptive Digital Rolls Out Voice Engine/SIP SDK for VoIP Application Developers

By Jayashree Adkoli
TMCnet Contributor
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Developers seeking to create VoIP applications for Android (News - Alert), Apple and Windows devices using a simple Software Development Kits (SDK) now have a new choice, as Adaptive Digital Technologies (Adaptive Digital) has released the VoIP Engine/SIP SDKs.


Simplifying the mobile handset VoIP application development, Adaptive Digital’s VoIP Engine Development Kit harnesses the power of the company’s VoIP Engine and helps in augmenting both speed-to-market and product quality.

Adaptive Digital’s VoIP Engine is a framework with a collection of many algorithms, such as conferencing, vocoders, noise reduction, echo cancellation and more, all packaged in a library.

As a data processing engine, the VoIP Engine has no interface to drivers or peripherals and performs processing solely at the request of the host application. Consequently, it is portable for use in conjunction with any application or operating system. It takes an audio stream and performs all the processing necessary to hand off an RTP packet to the network stack, and the same in the opposite direction.

As quoted in a statement, Adaptive Digital’s VoIP Engine/SIP Reference Kits, include both a voice over internet protocol (VoIP) engine SDK, and a Session Initiation Protocol (News - Alert) (SIP) SDK that can be used together to accelerate the development of cutting edge VoIP applications, while delivering the best user experience for mobile clients.

Powered by VoIP Engine, the SDK features acoustic echo cancellation, noise reduction, and automatic gain control (AGC) for voice quality enhancement (VQE), as well as speech compression (G.711, G.729AB, and G.722 and other optional codecs), RTP and Jitter Buffering.

Adaptive Digital’s SIP SDK provides a customizable solution to quickly add SIP based dial and receive phone calls features into software applications. It supports: registration, call initiation, call-acceptance, and call-teardown for VoIP telephones.

Currently, VoIP Engine/SIP Reference Kits are available on iPVoice for iOS-based devices and AnVoice for ARM (News - Alert)/Android-based devices. The Reference Design Kits include iPVoice – VoIP/SIP Reference Kit for the iPhone, iPad and iPod touch.

One of the key features included in the Reference Kit is G.722, a wideband compression algorithm, which passes 7 kHz of audio bandwidth rather than the 4 kHz that is carried by wired phones and cell phones. This results in clear voice unlike public switched telephone network.

Some of the test features included in the SIP phone application include tone transmit, tone receive, CSS (News - Alert) transmit, CSS receive and acoustic delay measurement.

The SIP phone application includes SIP client protocol, RTP/jitter buffer, voice conferencing (up to four users), G.711 mu-law and a-law with packet loss concealment, noise reduction, enhanced acoustic echo cancellation and more.




Edited by Rachel Ramsey


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