VoIP Engine™ Gateway Conference Bridge
Multi-channel Linux Based VoIP Engine™ Gateway Conference Solution
Handling all the voice processing from the system’s audio interface to the packet interface, and back again.
Adaptive Digital Voice Solutions
Conferencing: PCM Signal Processing, PCM to Packet, Packet to Packet
Adaptive Digital’s sophisticated solution enables the developer to create high quality multimedia conferencing systems whether the product be a web application or end equipment.
VoIP Engine™(VE) Gateway (VE GW) Conferencing Bridge
Voice over Internet Protocol (VoIP) media gateways perform the conversion between Time-division multiplexing (TDM) voice to a media streaming protocol, such as the Real-time Transport Protocol, (RTP), as well as a signaling protocol used in the VoIP system. The media gateway’s main function is to connect different types of networks converting between different transmission and coding techniques. Voice quality and data streaming algorithms such as echo cancellation, packet loss concealment (PLC), and tone relay are also located in the media gateway.
Adaptive Digital’s VoIP Engine Gateway product provides the developer an extensive VoIP software library and features that can be configured in such a way to build a wide variety of custom Voice and Video over Packet product solutions.
VoIP solution service providers can offer customers a range of value-added voice services such as conferencing, fax over IP, Caller ID, etc.
VE Gateway-Linux Software Development Kit (SDK) includes LnxVoice™ Adaptive Digital’s Linux based voice engine software framework.
VoIP CONFERENCE BRIDGE FUNCTIONALITY
A gateway allows data to flow from one discrete network to another.
- Simple API
- UDP socket processing
- Configurable RTP/Jitter Buffer
- Packet Loss Concealment
- Multi-conference capable
- Audio Switch/Transcoding mode
- Listen Only mode
- Record/Playback of conference input and output audio
- Number of conference participants is user configurable
- Participants can be added and deleted from conference at any time
- Dominant Speaker Selection
- Priority Member Assignment
VoIP SOFTWARE LIBRARY | FEATURES
Adaptive Digital VoIP Software library included in the VE Gateway -
The VoIP DSP library contains the following modules:
- G.711 μ/a-law appendix I (PCM) and II (VAD/CNG)
- G.726 40/32/24/16 Kbps ADPCM modes– optional
- G.729 A/B including Annex A and Annex B
- G.723.1 – 5.3/6.3 Kbps – optional
- G.722 – optional
- G.722.2 – optional
- AMR – optional
- DTMF, MFR1, MFR2 Detection and Generation
- Call Progress Detection and Generation, AGC
- HD AEC Acoustic Echo Cancellation
- G.168-2000 Echo Cancellation, 128 msec. tail
- Packet Loss Concealment (PLC)
- Silence Suppression
- Voice activity detection (VAD)
- Comfort noise generation (CNG)
- Comfort noise level control
- Tone Generation and Detection
- Jitter Buffering and lost packet recovery
- μ-Law/a-Law to Linear Conversion and Gain Control
- Play/Record of Audio Files
- RTP Packetization
- Conferencing
- Caller ID
- Fax Pass-Through (PCM)
- T.38 real time fax relay
- API support (management, event monitoring/reporting, statistics)
Application Software Packaging
Library Builds are Hardware / Software Specific: Currently available for Linux OS. Near future: Windows/PC, Android, iOS, and MacOS.
VoIP Engine Gateway High Level Block Diagram
VoIP Engine Gateway High Level Block Diagram showing variants
VE Gateway Solution plus PJSIP* SDK
- Gateway SDK Library Build / GW API’s
- Sample Host Application for Gateway
- PJSIP Library Build
- Sample Host Application for PJSIP
- VoIP Engine
*PJSIP is provided as a separate module.
VE Gateway Solution plus ADT*/Other SIP SDK
- Gateway SDK Library Build / GW API’s
- Sample Host Application for Gateway
- ADT SIP Library Build or Other
- LnxVoice VoIP Engine [handles Application Code for SIP (ADT or Another)]
*ADT-SIP is provided as a separate module.
VE Conference Bridge Solution / SDK
- Gateway SDK Library Build / GW API’s
- Sample Host Application for Gateway
- PJSIP Library Build
- Sample Host Application for PJSIP
- VoIP Engine
Core Processing: VoIP Engine – G.711, G.726, G.722 with ADT PLC, Opus
Collaborative Tools: Priority Speaker, Audio Record/Playback, multi-conference room management
Signal Processing: UDP, Transcoding
The VE GW software can be customized to Linux or ARM-Cortex-A platform of choice.
The Gateway APIs provide a method for application programs to establish VoIP calls over RTP utilizing the voice engine library’s signal processing capabilities.
A few simple APIs allow SIP registration, calling, and receiving functionality to be quickly integrated into your project.
Adaptive Digital’s sophisticated solution enables the developer to create high quality multimedia conferencing systems whether the product be a web application or end equipment.
- The Voice Quality competes with and surpasses solutions currently in the market.
- Individual building blocks can be used independently for even more specific designs.
- Supports SIP Protocol stacks: PJSIP, ADT-SIP or none.
- Supports Conferencing: Multiple Conference in multiple locations.
- The API is simple, flexible, and easy to work with.