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API for telephony implementation

SDK for telephony implementation

Technology


The Waverlysoft API and Waverlysoft API for .NET are based on asynchronous function calls and events. This way of working offers developers powerful flexibility to arrange the system architecture as they choose. For example, it is possible to arrange for each concurrent call to be processed in its own thread or to handle the processing of calls in a single thread, being driven by API events. Event handling can be done by callback functions, by using a Windows event object or by using Linux signals, whichever is more convenient for the application architecture. When using callback functions, the source code of the application is independent from the operating system.

The Waverlysoft API has a full range of voice, speech, conferencing, unified messaging, fax, and modem functions, and can implement speech enhanced IVR, CTI, UM, or conferencing applications. The supported functions (for example, echo cancellation, automatic gain control, V.90 modem) rely on the type of Media Board used.

Highlights of the Waverlysoft API include:

  • Initiating outgoing calls or processing inbound calls
  • Processing of called and calling (DNIS, CLI), as well as redirected/redirecting numbers
  • Control call ringing, reject, and line busy-out; answering calls based on timeslot/channel or called and calling numbers
  • Call progress — Cause codes, ISDN call states, interpretation of in-band tones, fax status
  • Tone detection — DTMF, busy, ring, SIT tones, modem/fax tones
  • VAD — Onboard voice activity detection, for human speech and silence detection
  • Answering Machine Detection
  • Barge-in (DTMF or VAD trigger)
  • Playing of prompts in A-law, µ-law, PCM, ADPCM, WAV or raw, including rewind, pause etc.
  • Playing of continuous audio streams, for example, for music or TTS
  • Control of volume and speed of playback • Echo cancellation, for example, in ASR and conferencing applications
  • Recording of audio to memory or disk (raw or WAV formats)
  • Call transfer (tromboned, blind, supervised, ECT path replacement)
  • Conferencing, including conference rights (whether members can talk or just listen), active talker detection and AGC mixing
  • Supplementary services — Hold/Retrieve, transfer, CD, MWI, UUI, Facility, etc.
  • User-to-User Signaling
  • Sending and receiving facility messages at any time
  • Full control over all bearer capabilities
  • Efficient streaming of audio for ASR/TTS engines using “audio provider” API
  • Send and receive fax (TIF, G3, color fax - JPEG), single and multiple fax files
  • Concurrent handling of VoIP and traditional telephony calls, for example, E1, T1, Analog
  • Detection and generation of generic tones under developer’s control
  • Monitoring API, allowing passive call recording and logging of numbers
  • Windows and Linux support — Allows source-compatible applications across platforms
  • Microsoft SOA supported by third party “visual SDK”
  • About Us


    Our Vision

    To be one of the foremost Channel Partner Distributor in Asia Pacific for SIP-based telephony fundamental stack.

    Our Mission

    To be the leading regional distribution partner through extensive application support

    Our Product Line handles various application needs in Telecommunications.

    Our customers include all major system integrator companies in Asia Pacific, supported through our network of distributors.

    Supported by a team of tech engineers, we manage customer requirements from initial design to systems deployment.

    Unwavering Application Support are our unique strengths, spearheading our expansion into multiple geographies, adding value to the increasingly competitive global telephony technology distribution markets.

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