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Feature/minimal media anchoring (#36)

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* initial WIP to remove freeswitch from media path when not recording or transcribing dial calls

* implement release-media and anchor-media operations

* mute/unmute now handled by rtpengine

* Dial: dtmf detection now based on SIP INFO events from sbcs and rtpengine

* add reason to gather action, bugfixes for transcribe and say
This commit is contained in:
Dave Horton
2021-10-21 11:59:45 -04:00
committed by GitHub
parent bedf25c6a2
commit 72345f83c1
10 changed files with 228 additions and 4838 deletions
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2
lib/tasks/transcribe.js
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@@ -91,7 +91,7 @@ class TaskTranscribe extends Task {
ep.addCustomEventListener(GoogleTranscriptionEvents.Transcription, this._onTranscription.bind(this, cs, ep));
ep.addCustomEventListener(GoogleTranscriptionEvents.NoAudioDetected, this._onNoAudio.bind(this, cs, ep));
ep.addCustomEventListener(GoogleTranscriptionEvents.MaxDurationExceeded,
this._onMaxDurationExceeded.bind(this, ep));
this._onMaxDurationExceeded.bind(this, cs, ep));
ep.addCustomEventListener(AwsTranscriptionEvents.Transcription, this._onTranscription.bind(this, cs, ep));
ep.addCustomEventListener(AwsTranscriptionEvents.NoAudioDetected, this._onNoAudio.bind(this, cs, ep));
ep.addCustomEventListener(AwsTranscriptionEvents.MaxDurationExceeded,