* feat: support srtpEncryption on dial verb for sip URI targets
Reads the new dial.srtpEncryption option and, for sip: URI targets,
forwards it to sbc-outbound as an X-Jambonz-SRTP header so the outbound
leg negotiates encrypted media (SRTP) — e.g. to a sips:/TLS endpoint
such as LiveKit. No behavior change when the option is unset.
Requires @jambonz/verb-specifications with the srtpEncryption spec
(dependency bump to follow once published) and sbc-outbound honoring
the X-Jambonz-SRTP header.
Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
* update verb specification
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Co-authored-by: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
The optionalDependencies pinned utf-8-validate ^6.0.3, but ws@8 only
accepts ^5.0.2 as its optional peer and ibm-watson's websocket dep
hard-requires ^5.0.2. That 5.x-vs-6.x split forced a dual-version tree
that npm 10 and npm 11 lay out differently, so a lockfile generated with
npm 11 (local) failed 'npm ci' under npm 10 (CI, node 20) with
'Missing: utf-8-validate@5.0.10 from lock file'. The 6.x pin never
accelerated ws anyway. Removing it collapses to a single utf-8-validate
5.0.10 that both npm majors resolve identically.
Co-Authored-By: Claude Opus 4.8 <noreply@anthropic.com>
* refactor recording
removed the test of `(this.cs.accountInfo.account.record_all_calls || this.cs.application.record_all_calls` from backround-task-manager.jsL138 as this check is already done in call-session.js at Line 3007, also allows us to start the record from update or config verbs
* handle start recording for a call that is not yet answered
* return false if not changing recording state
* different check for status
* set hasRecording flag on callInfo when starting
* update redis on recording start
* lint
* update dependency
* compare sdp for transcoding
* refactor sdp check for leading codec
* fix reference to epOther
* minor changes
* minor
* fix#1447
* fix security issue
* use convenience getter appIsUsingWebsockets in CallSession
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Co-authored-by: Dave Horton <daveh@beachdognet.com>
* for deepgram flux include the turn taking events in the transcription payload
* for deepgram flux, including turn_taking_event in the speech payload
* fix prev commit which used wrong field
* update to speech-utils that generates playback id
* modify tts and say task to track current playback id and match against start and stop events
* bump speech utils
* wip
* wip
* fix race condition where say with playbackId gets stop event from previous play from cache file
* logging
* wip
* fix comparison when playing cached files
* logging
Hoan Luu Huu