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15 Commits

Author SHA1 Message Date
Vinod Dharashive
961c2589ac freeswitch capture sip error and propagate the same error (#1489)
* fix: propagate SIP 488 error to SBC on endpoint allocation failure

When FreeSWITCH returns a SIP 488 'Not Acceptable Here' error during
endpoint allocation (e.g., codec incompatibility), this error was not
being propagated back to the SBC/client. Instead, the call would wait
indefinitely for websocket commands or return a generic 603 response.

Implementation:
- In _evalEndpointPrecondition(), detect SipError by checking
  err.type === 'SipError' or err.name === 'SipError'
- Extract the SIP status code (e.g., 488), reason, and the Reason
  header from the error response (e.g., Q.850;cause=88;text=INCOMPATIBLE_DESTINATION)
- Send the SIP error response immediately to the SBC with:
  - X-Reason header: endpoint allocation failure details
  - Reason header: original Q.850 cause from FreeSWITCH
- Notify call status change as Failed with proper SIP status
- Release the call immediately instead of waiting for commands

Also added fallback handling in InboundCallSession._onTasksDone() to
propagate the stored error if immediate send was not possible.

* wip

* Simplify SipError check to only use err.name
2026-01-13 08:58:37 -05:00
Hoan Luu Huu
e4ec0025c3 Fix/gladia multi sessions (#1487)
* support gladia transcribe multi channels

* wip
2026-01-07 10:46:33 -05:00
Ed Robbins
ba275ef547 #1485 remove deprecated call to URL.parse (#1488) 2026-01-05 15:53:32 -05:00
Dave Horton
83a8cf6d25 SIGUSR1 should cause fs to commence drying up calls but do not exit when call count reaches zero (#1486) 2026-01-04 12:23:26 -05:00
Sam Machin
09220872ae Update recording (#1483)
* 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
2026-01-02 11:05:38 -05:00
Dave Horton
fdce05fa40 add handler for SIGUSR1 to start drying up calls, useful as a generic mechanism on non-AWS deployments (#1482) 2025-12-30 13:31:42 -05:00
Sam Machin
3bd1dd6323 put removeListner in a try/catch (#1479)
* put removeListner in a try/catch

* typo
2025-12-19 13:31:06 -05:00
Ed Robbins
54dc172ebd Allow defining an ENV for specific webhook error return SIP code (#1476) 2025-12-16 17:14:42 -05:00
Hoan Luu Huu
e007e0e2d3 fixed callsession cannot close tts streaming (#1472) 2025-12-16 07:58:54 -05:00
Hoan Luu Huu
c5cd488fdf fixed gather should ignore transcription if task is killed/resolved. (#1465)
* fixed gather should ignore transcription if task is killed/resolved.

* wip
2025-12-12 09:03:08 -05:00
Sam Machin
57982335e0 add label to STT/TTS alerts (#1468)
* add label to STT/TTS alerts

* update time-series
2025-12-11 11:07:24 -05:00
Hoan Luu Huu
5cea91e18a add support for sending DTMF to ultravox (#1471) 2025-12-11 07:53:59 -05:00
Dave Horton
e396b6aa98 fix #1466: (#1467)
* fix #1466:

* do not send tts streaming events when we are not doing tts streaming
2025-12-09 09:43:53 -05:00
Vinod Dharashive
9104ebb603 Add configurable say chunk size (#1461) 2025-12-08 10:54:27 -05:00
Vinod Dharashive
1ad0261336 Enhance TTS sentence boundary detection for Arabic and Japanese (#1464)
Update sentenceEndRegex to treat the following as sentence boundaries: ASCII .!? followed by whitespace or end-of-text; Arabic question mark (؟) and full stop (۔) with the same rule; Japanese 。, !, ? treated as boundaries regardless of following character; and double newlines (\n\n). This improves streaming chunking for mixed-language content.
2025-12-08 10:44:20 -05:00
21 changed files with 3733 additions and 1787 deletions

View File

@@ -139,6 +139,11 @@ const JAMBONES_USE_FREESWITCH_TIMER_FD = process.env.JAMBONES_USE_FREESWITCH_TIM
const JAMBONES_DIAL_SBC_FOR_REGISTERED_USER = process.env.JAMBONES_DIAL_SBC_FOR_REGISTERED_USER || false;
const JAMBONES_MEDIA_TIMEOUT_MS = process.env.JAMBONES_MEDIA_TIMEOUT_MS || 0;
const JAMBONES_MEDIA_HOLD_TIMEOUT_MS = process.env.JAMBONES_MEDIA_HOLD_TIMEOUT_MS || 0;
const JAMBONES_WEBHOOK_ERROR_RETURN = parseInt(process.env.JAMBONES_WEBHOOK_ERROR_RETURN, 10) || 480;
/* say / tts */
const JAMBONES_SAY_CHUNK_SIZE = parseInt(process.env.JAMBONES_SAY_CHUNK_SIZE, 10) || 900;
// jambonz
const JAMBONES_TRANSCRIBE_EP_DESTROY_DELAY_MS =
process.env.JAMBONES_TRANSCRIBE_EP_DESTROY_DELAY_MS;
@@ -231,5 +236,7 @@ module.exports = {
JAMBONES_DIAL_SBC_FOR_REGISTERED_USER,
JAMBONES_MEDIA_TIMEOUT_MS,
JAMBONES_MEDIA_HOLD_TIMEOUT_MS,
JAMBONES_SAY_CHUNK_SIZE,
JAMBONES_TRANSCRIBE_EP_DESTROY_DELAY_MS,
JAMBONES_WEBHOOK_ERROR_RETURN
};

View File

@@ -291,7 +291,7 @@ router.post('/',
}, {
...(account.enable_debug_log && {level: 'debug'})
});
app.requestor.logger = app.notifier.logger = sipLogger;
app.requestor.logger = app.notifier.logger = restDial.logger = sipLogger;
const callInfo = new CallInfo({
direction: CallDirection.Outbound,
req: inviteReq,

View File

@@ -12,7 +12,8 @@ const RootSpan = require('./utils/call-tracer');
const listTaskNames = require('./utils/summarize-tasks');
const {
JAMBONES_MYSQL_REFRESH_TTL,
JAMBONES_DISABLE_DIRECT_P2P_CALL
JAMBONES_DISABLE_DIRECT_P2P_CALL,
JAMBONES_WEBHOOK_ERROR_RETURN
} = require('./config');
const { createJambonzApp } = require('./dynamic-apps');
const { decrypt } = require('./utils/encrypt-decrypt');
@@ -480,7 +481,7 @@ module.exports = function(srf, logger) {
message: `${err?.message}`.trim()
}).catch((err) => this.logger.info({err}, 'Error generating alert for parsing application'));
logger.info({err}, `Error retrieving or parsing application: ${err?.message}`);
res.send(480, {headers: {'X-Reason': err?.message || 'unknown'}});
res.send(JAMBONES_WEBHOOK_ERROR_RETURN, {headers: {'X-Reason': err?.message || 'unknown'}});
app.requestor.close(WS_CLOSE_CODES.GoingAway);
}
}

View File

@@ -12,6 +12,7 @@ class CallInfo {
let srf;
this.direction = opts.direction;
this.traceId = opts.traceId;
this.hasRecording = false;
this.callTerminationBy = undefined;
if (opts.req) {
const u = opts.req.getParsedHeader('from');

View File

@@ -504,7 +504,12 @@ class CallSession extends Emitter {
}
get isTtsStreamEnabled() {
return this.backgroundTaskManager.isTaskRunning('ttsStream');
// 1st background tts stream
return this.backgroundTaskManager.isTaskRunning('ttsStream') ||
// 2nd current task streaming tts
TaskName.Say === this.currentTask?.name && this.currentTask?.isStreamingTts ||
// 3rd nested verb is streaming tts
TaskName.Gather === this.currentTask?.name && this.currentTask.sayTask?.isStreamingTts;
}
get isListenEnabled() {
@@ -751,69 +756,101 @@ class CallSession extends Emitter {
return this._fillerNoise;
}
async pauseOrResumeBackgroundListenIfRequired(action, silence = false) {
if ((action == 'pauseCallRecording' || action == 'resumeCallRecording') &&
this.backgroundTaskManager.isTaskRunning('record')) {
this.logger.debug({action, silence}, 'CallSession:pauseOrResumeBackgroundListenIfRequired');
const backgroundListenTask = this.backgroundTaskManager.getTask('record');
const status = action === 'pauseCallRecording' ? ListenStatus.Pause : ListenStatus.Resume;
backgroundListenTask.updateListen(
status,
silence
);
}
}
async notifyRecordOptions(opts) {
const {action, silence} = opts;
const {action, silence = false, type = 'siprec'} = opts;
this.logger.debug({opts}, 'CallSession:notifyRecordOptions');
this.pauseOrResumeBackgroundListenIfRequired(action, silence);
/* if we have not answered yet, just save the details for later */
if (!this.dlg) {
if (action === 'startCallRecording') {
this.recordOptions = opts;
return true;
if (type == 'cloud') {
switch (action) {
case 'pauseCallRecording':
if (this.backgroundTaskManager.isTaskRunning('record')) {
this.logger.debug({action, silence, type}, 'CallSession:cloudRecording');
const backgroundListenTask = this.backgroundTaskManager.getTask('record');
backgroundListenTask.updateListen(
ListenStatus.Pause,
silence
);
return true;
} else { return false; }
case 'resumeCallRecording':
if (this.backgroundTaskManager.isTaskRunning('record')) {
this.logger.debug({action, silence, type}, 'CallSession:cloudRecording');
const backgroundListenTask = this.backgroundTaskManager.getTask('record');
backgroundListenTask.updateListen(
ListenStatus.Resume,
silence
);
return true;
} else { return false; }
case 'startCallRecording':
if (!this.backgroundTaskManager.isTaskRunning('record')) {
this.logger.debug({action, silence, type}, 'CallSession:cloudRecording');
this.callInfo.hasRecording = true;
this.updateCallStatus(Object.assign({}, this.callInfo.toJSON()), this.serviceUrl)
.catch((err) => this.logger.error(err, 'redis error'));
if (!this.dlg) {
// Call not yet answered so set flag to record on status change
this.application.record_all_calls = true;
} else {
this.backgroundTaskManager.newTask('record');
}
return true;
} else { return false; }
case 'stopCallRecording':
if (this.backgroundTaskManager.isTaskRunning('record')) {
this.logger.debug({action, silence, type}, 'CallSession:cloudRecording');
this.backgroundTaskManager.stop('record');
return true;
} else { return false; }
}
} else {
// SIPREC
/* if we have not answered yet, just save the details for later */
if (!this.dlg) {
if (action === 'startCallRecording') {
this.recordOptions = opts;
return true;
}
return false;
}
return false;
}
/* check validity of request */
if (action == 'startCallRecording' && this.recordState !== RecordState.RecordingOff) {
this.logger.info({recordState: this.recordState},
'CallSession:notifyRecordOptions: recording is already started, ignoring request');
return false;
}
if (action == 'stopCallRecording' && this.recordState === RecordState.RecordingOff) {
this.logger.info({recordState: this.recordState},
'CallSession:notifyRecordOptions: recording is already stopped, ignoring request');
return false;
}
if (action == 'pauseCallRecording' && this.recordState !== RecordState.RecordingOn) {
this.logger.info({recordState: this.recordState},
'CallSession:notifyRecordOptions: cannot pause recording, ignoring request ');
return false;
}
if (action == 'resumeCallRecording' && this.recordState !== RecordState.RecordingPaused) {
this.logger.info({recordState: this.recordState},
'CallSession:notifyRecordOptions: cannot resume recording, ignoring request ');
return false;
}
/* check validity of request */
if (action == 'startCallRecording' && this.recordState !== RecordState.RecordingOff) {
this.logger.info({recordState: this.recordState},
'CallSession:notifyRecordOptions: recording is already started, ignoring request');
return false;
}
if (action == 'stopCallRecording' && this.recordState === RecordState.RecordingOff) {
this.logger.info({recordState: this.recordState},
'CallSession:notifyRecordOptions: recording is already stopped, ignoring request');
return false;
}
if (action == 'pauseCallRecording' && this.recordState !== RecordState.RecordingOn) {
this.logger.info({recordState: this.recordState},
'CallSession:notifyRecordOptions: cannot pause recording, ignoring request ');
return false;
}
if (action == 'resumeCallRecording' && this.recordState !== RecordState.RecordingPaused) {
this.logger.info({recordState: this.recordState},
'CallSession:notifyRecordOptions: cannot resume recording, ignoring request ');
return false;
}
this.recordOptions = opts;
this.recordOptions = opts;
switch (action) {
case 'startCallRecording':
return await this.startRecording();
case 'stopCallRecording':
return await this.stopRecording();
case 'pauseCallRecording':
return await this.pauseRecording();
case 'resumeCallRecording':
return await this.resumeRecording();
default:
throw new Error(`invalid record action ${action}`);
switch (action) {
case 'startCallRecording':
return await this.startRecording();
case 'stopCallRecording':
return await this.stopRecording();
case 'pauseCallRecording':
return await this.pauseRecording();
case 'resumeCallRecording':
return await this.resumeRecording();
default:
throw new Error(`invalid record action ${action}`);
}
}
}
@@ -927,7 +964,7 @@ class CallSession extends Emitter {
this.logger.debug('CallSession:enableBackgroundTtsStream - ttsStream enabled');
} else {
this.logger.debug(
'CallSession:enableBackgroundTtsStream - ignoring request as call does not have required conditions');
'CallSession:enableBackgroundTtsStream - ignoring request; conditions not met (probably not using ws api)');
}
} catch (err) {
this.logger.info({err, say}, 'CallSession:enableBackgroundTtsStream - Error creating background tts stream task');
@@ -941,9 +978,11 @@ class CallSession extends Emitter {
}
}
clearTtsStream() {
this.requestor?.request('tts:streaming-event', '/streaming-event', {event_type: 'user_interruption'})
.catch((err) => this.logger.info({err}, 'CallSession:clearTtsStream - Error sending user_interruption'));
this.ttsStreamingBuffer?.clear();
if (this.isTtsStreamEnabled) {
this.requestor?.request('tts:streaming-event', '/streaming-event', {event_type: 'user_interruption'})
.catch((err) => this.logger.info({err}, 'CallSession:clearTtsStream - Error sending user_interruption'));
this.ttsStreamingBuffer?.clear();
}
}
startTtsStream() {
@@ -951,7 +990,7 @@ class CallSession extends Emitter {
}
stopTtsStream() {
if (this.appIsUsingWebsockets) {
if (this.isTtsStreamEnabled) {
this.requestor?.request('tts:streaming-event', '/streaming-event', {event_type: 'stream_closed'})
.catch((err) => this.logger.info({err}, 'CallSession:clearTtsStream - Error sending user_interruption'));
this.ttsStreamingBuffer?.stop();
@@ -1248,9 +1287,10 @@ class CallSession extends Emitter {
}
else {
writeAlerts({
alert_type: AlertType.STT_NOT_PROVISIONED,
alert_type: type === 'tts' ? AlertType.TTS_NOT_PROVISIONED : AlertType.STT_NOT_PROVISIONED,
account_sid: this.accountSid,
vendor,
label,
target_sid: this.callSid
}).catch((err) => this.logger.error({err}, 'Error writing tts alert'));
}
@@ -2460,6 +2500,36 @@ Duration=${duration} `
}
else {
this.logger.error(err, `Error attempting to allocate endpoint for for task ${task.name}`);
// Check for SipError type (e.g., 488 codec incompatibility)
const isSipError = err.name === 'SipError';
if (isSipError && err.status) {
// Extract Reason header from SIP response if available (e.g., Q.850;cause=88;text="INCOMPATIBLE_DESTINATION")
const sipReasonHeader = err.res?.msg?.headers?.reason;
this._endpointAllocationError = {
status: err.status,
reason: err.reason || 'Endpoint Allocation Failed',
sipReasonHeader
};
this.logger.info({endpointAllocationError: this._endpointAllocationError},
'Captured SipError for propagation to SBC');
// Send SIP error response immediately for inbound calls
if (this.res && !this.res.finalResponseSent) {
this.logger.info(`Sending ${err.status} response to SBC due to SipError`);
this.res.send(err.status, {
headers: {
'X-Reason': `endpoint allocation failure: ${err.reason || 'Endpoint Allocation Failed'}`,
...(sipReasonHeader && {'Reason': sipReasonHeader})
}
});
this._notifyCallStatusChange({
callStatus: CallStatus.Failed,
sipStatus: err.status,
sipReason: err.reason || 'Endpoint Allocation Failed'
});
this._callReleased();
}
}
throw new Error(`${BADPRECONDITIONS}: unable to allocate endpoint`);
}
}
@@ -2972,8 +3042,7 @@ Duration=${duration} `
// manage record all call.
if (callStatus === CallStatus.InProgress) {
if (this.accountInfo.account.record_all_calls ||
this.application.record_all_calls) {
if (this.accountInfo.account.record_all_calls || this.application.record_all_calls) {
this.backgroundTaskManager.newTask('record');
}
} else if (callStatus == CallStatus.Completed) {

View File

@@ -60,6 +60,19 @@ class InboundCallSession extends CallSession {
}
});
}
else if (this._endpointAllocationError) {
// Propagate SIP error from endpoint allocation failure back to the client
const {status, reason, sipReasonHeader} = this._endpointAllocationError;
this.rootSpan.setAttributes({'call.termination': `endpoint allocation SIP error ${status}`});
this.logger.info({endpointAllocationError: this._endpointAllocationError},
`InboundCallSession:_onTasksDone generating ${status} due to endpoint allocation failure`);
this.res.send(status, {
headers: {
'X-Reason': `endpoint allocation failure: ${reason}`,
...(sipReasonHeader && {'Reason': sipReasonHeader})
}
});
}
else {
this.rootSpan.setAttributes({'call.termination': 'tasks completed without answering call'});
this.logger.info('InboundCallSession:_onTasksDone auto-generating non-success response to invite');

View File

@@ -500,6 +500,10 @@ class TaskGather extends SttTask {
this.addCustomEventListener(ep, GladiaTranscriptionEvents.ConnectFailure,
this._onVendorConnectFailure.bind(this, cs, ep));
this.addCustomEventListener(ep, GladiaTranscriptionEvents.Error, this._onVendorError.bind(this, cs, ep));
// gladia require unique url for each session
const {host, path} = await this.createGladiaLiveSession();
opts.GLADIA_SPEECH_HOST = host;
opts.GLADIA_SPEECH_PATH = path;
break;
case 'soniox':
@@ -881,7 +885,7 @@ class TaskGather extends SttTask {
this._fillerNoiseOn = false; // in a race, if we just started audio it may sneak through here
this.ep.api('uuid_break', this.ep.uuid)
.catch((err) => this.logger.info(err, 'Error killing audio'));
cs.clearTtsStream();
if (cs.isTtsStreamEnabled) cs.clearTtsStream();
}
return;
}
@@ -1318,6 +1322,8 @@ class TaskGather extends SttTask {
}
this.resolved = true;
// gather is resolved, prevent any further transcription events while resolve in progress
this.removeCustomEventListeners();
// If bargin is false and ws application return ack to verb:hook
// the gather should not play any audio
this._killAudio(this.cs);

View File

@@ -146,8 +146,9 @@ class TaskLlmUltravox_S2S extends Task {
return data;
}
_unregisterHandlers() {
_unregisterHandlers(ep) {
this.removeCustomEventListeners();
ep.removeAllListeners('dtmf');
}
_registerHandlers(ep) {
@@ -155,6 +156,7 @@ class TaskLlmUltravox_S2S extends Task {
this.addCustomEventListener(ep, LlmEvents_Ultravox.ConnectFailure, this._onConnectFailure.bind(this, ep));
this.addCustomEventListener(ep, LlmEvents_Ultravox.Disconnect, this._onDisconnect.bind(this, ep));
this.addCustomEventListener(ep, LlmEvents_Ultravox.ServerEvent, this._onServerEvent.bind(this, ep));
ep.on('dtmf', this._onDtmf.bind(this, ep));
}
async _startListening(cs, ep) {
@@ -189,7 +191,7 @@ class TaskLlmUltravox_S2S extends Task {
/* note: the parent llm verb started the span, which is why this is necessary */
await this.parent.performAction(this.results);
this._unregisterHandlers();
this._unregisterHandlers(ep);
}
async kill(cs) {
@@ -346,6 +348,18 @@ class TaskLlmUltravox_S2S extends Task {
excludeEvents: this.excludeEvents
}, 'TaskLlmUltravox_S2S:_populateEvents');
}
_onDtmf(ep, evt) {
this.logger.info({evt}, 'TaskLlmUltravox_S2S:_onDtmf - DTMF received');
const {dtmf} = evt;
const data = {
type: 'user_text_message',
text: `DTMF received: ${dtmf}`,
urgency: 'immediate'
};
this._api(ep, [ep.uuid, ClientEvent, JSON.stringify(data)])
.catch((err) => this.logger.info({err, evt}, 'TaskLlmUltravox_S2S:_onDtmf - Error sending DTMF as text message'));
}
}
module.exports = TaskLlmUltravox_S2S;

View File

@@ -33,7 +33,7 @@ class TaskRedirect extends Task {
}
else {
const baseUrl = this.cs.application.requestor.baseUrl;
const newUrl = URL.parse(this.actionHook);
const newUrl = new URL(this.actionHook);
const newBaseUrl = newUrl.protocol + '//' + newUrl.host;
if (baseUrl != newBaseUrl) {
try {

View File

@@ -1,6 +1,7 @@
const assert = require('assert');
const TtsTask = require('./tts-task');
const {TaskName, TaskPreconditions} = require('../utils/constants');
const {JAMBONES_SAY_CHUNK_SIZE} = require('../config');
const pollySSMLSplit = require('polly-ssml-split');
const { SpeechCredentialError, NonFatalTaskError } = require('../utils/error');
const { sleepFor } = require('../utils/helpers');
@@ -31,7 +32,7 @@ const isMatchingEvent = (logger, filename, playbackId, evt) => {
const breakLengthyTextIfNeeded = (logger, text) => {
// As The text can be used for tts streaming, we need to break lengthy text into smaller chunks
// HIGH_WATER_BUFFER_SIZE defined in tts-streaming-buffer.js
const chunkSize = 900;
const chunkSize = JAMBONES_SAY_CHUNK_SIZE;
const isSSML = text.startsWith('<speak>');
const options = {
softLimit: 100,

View File

@@ -203,26 +203,14 @@ class SttTask extends Task {
if (cs.hasGlobalSttPunctuation && !this.data.recognizer.punctuation) {
this.data.recognizer.punctuation = cs.globalSttPunctuation;
}
if (this.vendor === 'gladia') {
const { api_key, region } = this.sttCredentials;
const {url} = await this.createGladiaLiveSession({
api_key, region,
model: this.data.recognizer.model || 'solaria-1',
options: this.data.recognizer.gladiaOptions || {}
});
const {host, pathname, search} = new URL(url);
this.sttCredentials.host = host;
this.sttCredentials.path = `${pathname}${search}`;
}
}
async createGladiaLiveSession({
api_key,
region = 'us-west',
model = 'solaria-1',
options = {},
}) {
async createGladiaLiveSession() {
const { api_key, region = 'us-west' } = this.sttCredentials;
const model = this.data.recognizer.model || 'solaria-1';
const options = this.data.recognizer.gladiaOptions || {};
const url = `https://api.gladia.io/v2/live?region=${region}`;
const response = await fetch(url, {
method: 'POST',
@@ -252,7 +240,9 @@ class SttTask extends Task {
const data = await response.json();
this.logger.debug({url: data.url}, 'Gladia Call registered');
return data;
const {host, pathname, search} = new URL(data.url);
return {host, path: `${pathname}${search}`};
}
addCustomEventListener(ep, event, handler) {
@@ -286,6 +276,7 @@ class SttTask extends Task {
account_sid: cs.accountSid,
alert_type: AlertType.STT_NOT_PROVISIONED,
vendor,
label,
target_sid: cs.callSid
}).catch((err) => this.logger.info({err}, 'Error generating alert for no stt'));
// the ASR might have fallback configuration, should not done task here.
@@ -486,6 +477,7 @@ class SttTask extends Task {
message: 'STT failure reported by vendor',
detail: evt.error,
vendor: this.vendor,
label: this.label,
target_sid: cs.callSid
}).catch((err) => this.logger.info({err}, `Error generating alert for ${this.vendor} connection failure`));
}
@@ -499,6 +491,7 @@ class SttTask extends Task {
alert_type: AlertType.STT_FAILURE,
message: `Failed connecting to ${this.vendor} speech recognizer: ${reason}`,
vendor: this.vendor,
label: this.label,
target_sid: cs.callSid
}).catch((err) => this.logger.info({err}, `Error generating alert for ${this.vendor} connection failure`));
}

View File

@@ -459,6 +459,14 @@ class TaskTranscribe extends SttTask {
else if (this.data.recognizer?.hints?.length > 0) {
prompt = this.data.recognizer?.hints.join(', ');
}
} else if (this.vendor === 'gladia') {
// gladia require unique url for each session
const {host, path} = await this.createGladiaLiveSession();
await ep.set({
GLADIA_SPEECH_HOST: host,
GLADIA_SPEECH_PATH: path,
})
.catch((err) => this.logger.info(err, 'Error setting channel variables'));
}
await ep.startTranscription({

View File

@@ -274,6 +274,7 @@ class TtsTask extends Task {
account_sid,
alert_type: AlertType.TTS_NOT_PROVISIONED,
vendor,
label,
target_sid: cs.callSid
}).catch((err) => this.logger.info({err}, 'Error generating alert for no tts'));
throw new SpeechCredentialError('no provisioned speech credentials for TTS');
@@ -360,6 +361,7 @@ class TtsTask extends Task {
account_sid: cs.accountSid,
alert_type: AlertType.TTS_FAILURE,
vendor,
label,
detail: err.message,
target_sid: cs.callSid
}).catch((err) => this.logger.info({err}, 'Error generating alert for tts failure'));

View File

@@ -405,19 +405,21 @@ module.exports = (logger) => {
if (ep.amd) {
vendor = ep.amd.vendor;
ep.amd.stopAllTimers();
ep.removeListener(GoogleTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(GoogleTranscriptionEvents.EndOfUtterance, ep.amd.EndOfUtteranceHandler);
ep.removeListener(AwsTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(AzureTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(AzureTranscriptionEvents.NoSpeechDetected, ep.amd.noSpeechHandler);
ep.removeListener(NuanceTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(DeepgramTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(SonioxTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(IbmTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(NvidiaTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(JambonzTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
try {
ep.removeListener(GoogleTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(GoogleTranscriptionEvents.EndOfUtterance, ep.amd.EndOfUtteranceHandler);
ep.removeListener(AwsTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(AzureTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(AzureTranscriptionEvents.NoSpeechDetected, ep.amd.noSpeechHandler);
ep.removeListener(NuanceTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(DeepgramTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(SonioxTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(IbmTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(NvidiaTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
ep.removeListener(JambonzTranscriptionEvents.Transcription, ep.amd.transcriptionHandler);
} catch (error) {
logger.error('Unable to Remove AMD Listener', error);
}
ep.amd = null;
}

View File

@@ -135,26 +135,24 @@ class BackgroundTaskManager extends Emitter {
// Initiate Record
async _initRecord() {
if (this.cs.accountInfo.account.record_all_calls || this.cs.application.record_all_calls) {
if (!JAMBONZ_RECORD_WS_BASE_URL || !this.cs.accountInfo.account.bucket_credential) {
this.logger.error('_initRecord: invalid cfg - missing JAMBONZ_RECORD_WS_BASE_URL or bucket config');
return undefined;
}
const listenOpts = {
url: `${JAMBONZ_RECORD_WS_BASE_URL}/record/${this.cs.accountInfo.account.bucket_credential.vendor}`,
disableBidirectionalAudio: true,
mixType : 'stereo',
passDtmf: true
};
if (JAMBONZ_RECORD_WS_USERNAME && JAMBONZ_RECORD_WS_PASSWORD) {
listenOpts.wsAuth = {
username: JAMBONZ_RECORD_WS_USERNAME,
password: JAMBONZ_RECORD_WS_PASSWORD
};
}
this.logger.debug({listenOpts}, '_initRecord: enabling listen');
return await this._initListen({verb: 'listen', ...listenOpts}, 'jambonz-session-record', true, 'record');
if (!JAMBONZ_RECORD_WS_BASE_URL || !this.cs.accountInfo.account.bucket_credential) {
this.logger.error('_initRecord: invalid cfg - missing JAMBONZ_RECORD_WS_BASE_URL or bucket config');
return undefined;
}
const listenOpts = {
url: `${JAMBONZ_RECORD_WS_BASE_URL}/record/${this.cs.accountInfo.account.bucket_credential.vendor}`,
disableBidirectionalAudio: true,
mixType : 'stereo',
passDtmf: true
};
if (JAMBONZ_RECORD_WS_USERNAME && JAMBONZ_RECORD_WS_PASSWORD) {
listenOpts.wsAuth = {
username: JAMBONZ_RECORD_WS_USERNAME,
password: JAMBONZ_RECORD_WS_PASSWORD
};
}
this.logger.debug({listenOpts}, '_initRecord: enabling listen');
return await this._initListen({verb: 'listen', ...listenOpts}, 'jambonz-session-record', true, 'record');
}
// Initiate Transcribe

View File

@@ -100,6 +100,30 @@ module.exports = (logger) => {
else if (K8S) {
lifecycleEmitter.scaleIn = () => process.exit(0);
}
else {
process.on('SIGUSR1', () => {
logger.info('received SIGUSR1: begin drying up calls for scale-in');
dryUpCalls = true;
const {srf} = require('../..');
const {writeSystemAlerts} = srf.locals;
if (writeSystemAlerts) {
const {SystemState, FEATURE_SERVER} = require('./constants');
writeSystemAlerts({
system_component: FEATURE_SERVER,
state : SystemState.GracefulShutdownInProgress,
fields : {
detail: `feature-server with process_id ${process.pid} shutdown in progress`,
host: srf.locals?.ipv4
}
});
}
pingProxies(srf);
// Note: in response to SIGUSR1 we start drying up but do not exit when calls reach zero.
// This is to allow external scripts that sent the signal to manage the lifecycle.
});
}
async function pingProxies(srf) {

View File

@@ -127,7 +127,6 @@ class SttLatencyCalculator extends Emitter {
calculateLatency() {
if (!this.isRunning) {
this.logger.debug('Latency calculator is not running, cannot calculate latency, returning default values');
return null;
}

View File

@@ -1085,13 +1085,6 @@ module.exports = (logger) => {
...(keyterms && keyterms.length > 0 && {DEEPGRAMFLUX_SPEECH_KEYTERMS: keyterms.join(',')}),
};
}
else if ('gladia' === vendor) {
const {host, path} = sttCredentials;
opts = {
GLADIA_SPEECH_HOST: host,
GLADIA_SPEECH_PATH: path,
};
}
else if ('soniox' === vendor) {
const {sonioxOptions = {}} = rOpts;
const {storage = {}} = sonioxOptions;

View File

@@ -163,7 +163,6 @@ class TtsStreamingBuffer extends Emitter {
}
clear() {
this.logger.debug('TtsStreamingBuffer:clear');
if (this._connectionStatus !== TtsStreamingConnectionStatus.Connected) return;
clearTimeout(this.timer);
this._api(this.ep, [this.ep.uuid, 'clear']).catch((err) =>
@@ -437,7 +436,15 @@ class TtsStreamingBuffer extends Emitter {
const findSentenceBoundary = (text, limit) => {
// Look for punctuation or double newline that signals sentence end.
const sentenceEndRegex = /[.!?](?=\s|$)|\n\n/g;
// Includes:
// - ASCII: . ! ?
// - Arabic: ؟ (question mark), ۔ (full stop)
// - Japanese: 。 (full stop), , (full-width exclamation/question)
//
// For languages that use spaces between sentences, we still require
// whitespace or end-of-string after the mark. For Japanese (no spaces),
// we treat the punctuation itself as a boundary regardless of following char.
const sentenceEndRegex = /[.!?؟۔](?=\s|$)|[。!?]|\n\n/g;
let lastSentenceBoundary = -1;
let match;
while ((match = sentenceEndRegex.exec(text)) && match.index < limit) {

5114
package-lock.json generated

File diff suppressed because it is too large Load Diff

View File

@@ -33,8 +33,8 @@
"@jambonz/realtimedb-helpers": "^0.8.15",
"@jambonz/speech-utils": "^0.2.26",
"@jambonz/stats-collector": "^0.1.10",
"@jambonz/time-series": "^0.2.14",
"@jambonz/verb-specifications": "^0.0.122",
"@jambonz/time-series": "^0.2.15",
"@jambonz/verb-specifications": "^0.0.123",
"@modelcontextprotocol/sdk": "^1.9.0",
"@opentelemetry/api": "^1.8.0",
"@opentelemetry/exporter-jaeger": "^1.23.0",