VoIP Engine™ Gateway

Multi-channel Linux Based VoIP Engine™ Gateway Solution

Handling all the voice processing from the system’s audio interface to the packet interface, and back again.

Adaptive Digital Voice Solutions

Customizable, Scalable Multi-channel Gateway Solutions

Gateways can be an ingress or egress traffic to a Telecommunication node. A gateway can also act as a protocol converter. Gateways offer a level of Security to the Telecommunications Network.

world-connect gateway

VoIP Engine™(VE) Gateway (VE GW)

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.

Hardware Platforms:  x86_64 Linux, ARM Cortex-Axx (32-bit and 64-bit)


A gateway allows data to flow from one discrete network to another.

Management and configuration of resources during call negotiation and teardown.

Supports multi-host TCP-UDP/IP protocol and failover operation.

Management and configuration of local TDM connections.


  • SIP
  • G.711 μ/a-law appendix I (PCM) and II (VAD/CNG)
  • G.726
  • G.729 A/B including Annex A and Annex B
  • G.722
  • G.722.1 
  • G.722.2 Wide-band 
  • G.723.1
  • GSM-AMR Narrow-band 
  • iLBC
  • iSAC
  • OPUS
  • DTMF, MFR1, MFR2 Detection and Generation
  • Call Progress Detection and Generation
  • DTMF Tone Relay Transmit (IETF RFC2833)
  • Arbitrary Tone Detection
  • High Definition Acoustic Echo Cancellation (AEC)
  • G.168 Echo Cancellation
  • Automatic Gain Control (AGC)
  • Packet Loss Concealment (PLC)
  • Silence Suppression
  • Voice activity detection (VAD)
  • Comfort noise generation (CNG)
  • Comfort noise level control
  • μ-law/a-law to Linear Conversion and Gain Control
  • Play/Record of Audio Files
  • RTP/RTCP Packetization
  • Jitter Buffering and lost packet recovery
  • SRTP (Secure RTP) / AES Encryption
  • 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

VE-GW architecture Variants

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.
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