A full ANSI C VoIP SDK source code release
InstaVoIP™ Embedded from Unicoi is a platform-independent VoIP framework designed for standalone use or to be embedded into another application. The InstaVoIP Embedded software suite provides a full-featured Call Manager, Voice Engine, and Information Subsystem, providing developers with a comprehensive VoIP development solution for devices such as IP Phones, VoIP ATAs, RoIP Gateways, security call-boxes, and many others.
Full Source Code
A full ANSI C source code release allows InstaVoIP Embedded to be used on virtually any platform. Simply porting the Fusion Common Layer (FCL) abstraction to a platform’s RTOS/OS, TCP/IP stack, file system (optional), and implementing an audio driver channel for the platform’s audio hardware is all that is necessary to start making VoIP calls. Additionally, having full source code allows customer changes to be made to the code (such as implementing newer or less popular RFCs) and aids in debugging low-level problems.
InstaVoIP Embedded Features
- Voice Engine with RTP (includes support for RoIP products)
- Configuration Subsystem (file based or can integrate with other platform standard)
- Voice Engine Codecs: G.711 (µ-law, A-law), G.722, G.726 (16/24/33/40 kbps), G.729, DVI4 (narrow/HD/Ultra HD), iLBC, Speex (narrow/HD/Ultra HD), SILK, L16 PCM (narrow/HD/Ultra HD/CD)
- VoIP Call Manager (includes support for incoming calls, outgoing calls, hold, conferencing, transfers, mic/speaker volume control and mute, etc.)
- Optional Security (SIPS, SRTP)
Call Manager is the logical bridge among the VoIP layer, Voice Engine (VE) and the User Interface (UI). Fusion Call Manager effectively handles all incoming calls, outgoing calls, call transfer, call hold, call conference, etc. It processes the requests from the VoIP layer and User Interface, while controling the state of the Voice Engine as required.
- Supported Workflows: SoftPhone, Desktop Phone, POTS FXS, POTS FXO
- Actions: place calls, answer calls, disconnect calls, on/off hold, transfer, conference, generate DTMF, etc.
- Events: incoming call, peer on/off hold, peer disconnect, being transferred, DTMF received, registered/unregistered, etc.
- Call manager control via HTTP/JSON-based web service for remote control
The Fusion Voice Engine is a comprehensive solution consisting of an extensive set of algorithms and codecs designed for Digital Voice applications. The Fusion Voice Engine is designed for a wide range of applications, like PC VoIP, Smart Phones, RoIP devices, and Embedded Voice applications.
- Codecs: G.711 (µ-law, A-law), G.722, G.726 (16/24/33/40 kbps), G.729, DVI4 (narrow/HD/Ultra HD), iLBC, Speex (narrow/HD/Ultra HD), SILK, L16 PCM (narrow/HD/Ultra HD/CD)
- Algorithms: Gain Control, Automatic Gain Control (AGC), DC Blocker, High-Pass Filter, Voice Activity Detector (VAD), Audio Resampler (8K, 16K, 32K, 48K), DTMF (Generator/Detector), Call Progress Tone Generator, Custom Ring Tone Generator, Comfort Noise Generator (CNG), Conference Bridge (Mixer), Packet Loss Compensation, Acoustic Echo Canceller (AEC), Noise Reduction, Frequency Equalizer
The Information Subsystem is a built-in database providing intuitive access to configuration and status data. It manages information on behalf of the core InstaVoIP components as well as any data required by the end application.
The InstaVoIP Information Subsystem consists of the following items:
- Configuration Information Management
- File-based by default
- Can integrate with platform’s configuration style
- Runtime Information Management (e.g. call status)
- User-friendly, hierarchical organization of data
- Access via HTTP/JSON-based web service for remote configuration and status monitoring
Web-Based Configuration UI
An Easy-to-Use Web UI is provided with the following features:
- Optional: HTTPS for secure access
- Expandable to include user-application configuration
- Allows real-time control of key system functionality such as dialing and answering calls
Phone: 678-208-2250 E-mail: firstname.lastname@example.org