A full C Voice & Video SDK source code release
InstaV2IP® (pronounced “Insta-V-2-I-P”) Embedded from Unicoi Systems, Inc. is a video-enhanced V2IP (Voice and Video) software suite designed for standalone use or to be embedded into another application. InstaV2IP Embedded includes all of the trusted VoIP components from the InstaVoIP Embedded software suite like the Call Manager, Voice Engine, and Information Subsystem. But in addition to the InstaVoIP Embedded VoIP components, InstaV2IP Embedded provides a full-featured Video Engine SDK providing developers with a comprehensive V2IP development solution for devices such as Video Phones, Video Callboxes, Softphone App with Video Support, Video Conferencing, and more.
Full Source Code
A full C source code release allows InstaV2IP Embedded to be used on virtually any platform. Simply porting the Fusion Common Layer (FCL) abstraction to a platform’s RTOS/OS, network stack, and file system (optional), and implementing audio/video driver channels for the platform’s audiovisual hardware is all that is necessary to start making voice & video 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.
InstaV2IP Embedded Features
- Video Engine
- Video Engine Codec support for H.264, H.263, H.263+, MPEG-4
- Voice Engine with RTP (includes support for RoIP products)
- 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)
- Configuration Subsystem (file based or can integrate with other platform standard)
- VoIP/V2IP 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.
- 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 Video Engine, offers a complete voice & video solution on any platform, designed to be easily embedded into new or existing applications. The Fusion Video Engine is designed for a wide range of applications such as Video Phones, Video Callboxes, Softphone Apps with Video Support, and more. Whether an app developer or a video hardware engineer, Fusion Video Engine's VoIP/V2IP functionality can save you time and effort.
- Codecs: H.264, H.263, H.263+, MPEG-4
- Algorithms: Camera Support, Configurable Video Frame Rate, Configurable Video Resolution, Dynamic Bitrate Configuration, Audio/Video Sync, Video Snapshot, Video Privacy (transmit image from file), Video Statistics, Audio/Video Recording, Error recovery, Video Jitter Buffer, IPv6 Support
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 VoIP 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)
- 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
- * Not included in Android & iOS
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