Adaptive Digital Voice Solutions
We help you design the application you want.
VoIP Engine is portable for use in conjunction with any application or operating system.
VoIP Engine, VoIP Engine X, VoIP Engine XP
VoIP Engine Software: Handles all the voice processing from PCM to Packet and back
By leveraging VoIP Engine, developers can focus on functionality of the end application without dealing with complexities of voice processing at the native layer.
VoIP Engine (VE) is at the core of our Linux, and ARM-based VoIP applications, it provides complete PCM to packet processing. The VoIP Engine software is a software engine package that handles all the voice processing from PCM to Packet and back. Its intended use is in VoIP enabled handsets or desktop phones.
Our product and service offerings draw from extensive industry experience. Because of our extensive knowledge of digital voice processing we are well equipped to provide our customers the resources and support necessary for them to exceed the existing standard in voice quality.
Our engineers work closely with the customer’s engineering staff to transfer technologies, thus ensuring quick integration of licensed components into the customers’ product.
VoIP Engine X
Packet Conference Server
VoIP Engine XP -
VoIP Engine Gateway Solution
VoIP Engine Conference Bridge Solution*
VoIP Engine Terminal Solution*
*Future
Extreme programming (XP) is a software development methodology which is intended to improve software quality and responsiveness to changing customer requirements.
Features List
- G.711 with appendices 1 PLC and 2 (discontinuous transmission)
- G.729A Vocoder
- G.722 (wideband audio) with packet loss concealment
- AMR NB Vocoder
- AMR WB (G.722.2) Vocoder *
- G.726*
- EVRC-B
- MELP
- RTP/Jitter Buffer
- SRTP
- DTMF tone relay transmit (IETF RFC2833)
- HD Acoustic Echo Cancellation
- Noise Reduction
- Tone Generation
- Gain Control
- Automatic Gain Control
- Diagnostics to assist in acoustic tuning
- Equalization
- SIP*
*Optional
The VoIP engine is purely a data processing engine. It has no interface to drivers or peripherals and performs processing solely at the request of the host application. The host application feeds the VoIP engine PCM samples from the audio input and and RTP packets from the network input. The VoIP engine in turn returns, via callbacks to the host application, PCM samples to be sent to the audio output device and RTP packets to be sent to the network interface.
AVAILABLE AT VARIOUS LEVELS OF INTEGRATION
- VE source Code
- VE object Code
- VE Class Library
- VE SDK includes VE Class library and SIP Class Library
- VE Reference Kit includes Class Libraries and Sample Application code
Adaptive Digital’s VoIP Engine/SIP Reference Kits accelerate the development of SIP compliant voice applications.
Reference Kit Includes:
- VoIP Engine Software
- SIP Phone Sample Project with source code
- SDK Quick Start Guide
- Developer Quick start (Read Me)
VoIP Engine Software development kits (SDKs) are supplied with a sample Java application and a sample native application that in turn interfaces with the VoIP Engine software. The sample Java application interfaces with the sample native application via Java Native Interface (JNI) to setup an RTP/IP to RTP/IP VoIP connection. Android developers can incorporate the Java sample code into more complete VoIP-enabled Android applications.
The VoIP Engine API is clean and simple to use.
Add VoIP features to an existing application project or create a fully customized SIP application.
- VoIP Engine Class Library – Provides complete Voice over IP (VoIP) functionality including audio input and output, voice processing, RTP packetization, and network input and output.
- Sample SIP-Phone Program. This program is a fully functioning SIP phone. It is delivered in source code format, complete with all the necessary project files to build, run, test, and modify the program.
- The app can be configured to connect to a standard SIP server. The app can place outgoing phone calls as well as receive inbound phone calls. Furthermore, it supports peer to peer VoIP for applications that do not require SIP.
- SIP Class Library – Provides SIP user agent functionality including network input and output.
VoIP Engine is portable for use in conjunction with any application or operating system.
The VoIP Engine/SIP Reference Kit is a SIP Phone application (complete with audio algorithms, SIP, User Interface, control code, and sample application).
VQE algorithms (AnVoice specific to inherent android issues), HD Acoustic Echo Cancellation, Noise Reduction, , G.711, G.729AB, G.722, AMR WB, RTP, SRTP, SIP plus the sample application, which is provided in source code format along with project/make files.
Because you want to develop your new VoIP_enabled application quickly using robust, field-tested VoIP software. The work has been done for you. Save time and development costs. Flexibility, Wideband algorithms AEC, G.722.1 (Skype is the only other application using WB in cellular network), create new use of vonage, add VoIP capability to new and existing applications.
AnVoice: Android 2.1+
LnxVoice/ARM: ARM Cortex-A, Linux x86
iVoIPEngine: iOS, iPhone, & iPad
You will need to contact sales in order to download the reference kit software.
By phone: +1 610-825-0182 x120, or email sales@adaptivedigital.com
Detailed Users’ Guide, and sample code – API functionality explained.
Superior AEC right out of the box.
Based upon Adaptive Digital’s long history of echo cancellation products, the Enhanced AEC is customized to be able to deal with the challenges that exist primarily in the handset environment:
- Longer and non-deterministic audio buffering delays.
- Excessive speaker to microphone gain due to proximity of speaker and microphone.
- User controlled volume changes that take effect in the audio path between the AEC and the speaker.
VoIP Engine SDK – The VoIPEngine class encapsulates Adaptive Digital’s VoIP Engine software along with platform specific audio and network I/O to provide all voice processing functionality needed for a VoIP enabled application.
SIP SDK – The SIP class encapsulates Adaptive Digital’s SIP stack software along with platform specific network I/O to provide SIP functionality needed for a VoIP enabled application.
The VoIP Engine/SIP Reference Kit is subject to change as new features are added.
Yes, you will need to be familiar with application programming.
Yes, the project is own by Adaptive Digital. At the time of production/distribution, a license needs to be obtained.
G.711, G.729AB, G.722, AMR, and AMR-WB (G.722.2)
The reference kits are free for developmental purpose only. Product must be licensed for production/distribution.
Develop VoIP applications such as IP Intercom, social networking, add VoIP to multi-user gaming, etc.
Yes, the evaluation software allows for up to a 5 min phone call. The free version expires (90 days).
Download new, or contact us is same version is required for development.
See included User’s Guide for features and usability. Sample application source code is provided. For customization by Adaptive Digital’s team of experts, contact sales.
Contact sales +1 610-825-0182 x120 to discuss your requirements.
Contact sales to discuss licensing per your requirements.
Yes, see website for software availability, or contact sales +1 610-825-0182 x120 to discuss your requirements.
Reference Kits
AnVoice
The AnVoice – Android VoIP/SIP Reference Kit is a sample Android SIP Phone application named AnVoice that utilizes Adaptive Digital Technologies’ VoIPEngine and Sip class libraries. The application is provided as source code in an Eclipse project. The VoIPEngine and Sip class libraries are provided as jar files.
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.
- SIP Phone Sample Project with source code (Android based)
- AnVoice VoIP Engine SDK(includes an evaluation version of the VoIP class library, header file and docs)
- SIP SDK for Android (includes an evaluation version of the SIP class library, header file and docs)
- SDK Quick Start Guide
- Developer Quick start (Read Me)
The kit is designed to get you running quickly. Through the use of a few basic APIs you can configure the system to your liking and easily make and receive calls.
LnxVoice SDK is a complete VoIP stack solution that handles all voice processing from the system’s audio interface to the packet interface, and back. LnxVoice SDK features acoustic echo cancellation, noise reduction, and automatic gain control for voice quality enhancement , as well as speech compression (G.711, G.729AB, and G.722 and other optional codecs), RTP and Jitter Buffering.
The LnxVoice SDK directly interfaces with your system’s ALSA driver and network sockets, taking the difficult part out of VoIP implementations.
LnxVoice
- SIP Phone Sample Project with source code (Linux based)
- LnxVoice VoIP Engine SDK(includes an evaluation version of the VoIP library, header file and docs). Demo runs on TI Sitara platform.
- SIP SDK for Linux/x86 (includes an evaluation version of the SIP class library, header file and docs)
- SDK Quick Start Guide
- Developer Quick start (Read Me)
iVoIP Engine
Voice enable your iOS application
iPVoice sample program demonstrates the use of Adaptive Digital’s VoIP Engine software in a mobile handset environment.
In particular, the VoIP program, iVoIP Engine turns an iPhone (or iPod or iPad) into a limited capability IP phone. The limitation is that the program communicates via RTP using user specified IP address/port number.
- SIP Phone Sample Project with source code (iOS based)
- iVoIPEngine SDK(includes an evaluation version of the VoIP class library, header file and docs)
- SIP SDK for iOS (includes an evaluation version of the SIP class library, header file and docs)
- SDK Quick Start Guide
- Developer Quick start (Read Me)