Merge branch 'develop' v0.8.12

Author: rstrahan

CHANGELOG.md

@@ -7,6 +7,22 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 
 ## [Unreleased]
 
+## [0.8.12] - 2024-05-20
+### Fixed
+- Expanded redaction support to include all languages (and dialects) on the Amazon Transcribe supported languages page.
+- Call Transcriber Lambda
+   - Refactor call end handling, to explicitly process ENDED messages from VoiceConnector, rather than imputing call end from stream data ending - adds robustness when stream can end (temporarily) for other reasons (such as call hold, etc.)
+   - Identified a situation where if a call chained across multiple lambda invocations (lasted longer than 12 minutes), the new timer we introduced that checks if the call ends would leak into the next lambda invocation. This would have no effect for a single long running call other than duplicate checks to DynamoDB—however—if the call ends and a NEW call uses that warm Lambda, the new call would inherit the timers and within 5 seconds mark the call as ended. To fix this, we made modifications by adding proper clean-up of timers when the call ends, and also double checking when the next call begins that the new Lambda did not inherit any calls.
+   - Added the callId to the log output, so that it is easier to trace calls.
+   - Added the request id of the next Lambda invocation in the output of the log, so that it makes it easier to trace sequential lambda invocations (chaining).
+
+- Websocket server and Stream Audio client
+   - Fix state management in Websocket server - add connection specific state variables to a global dictionary to maintain state for active connections
+   - Fix reconnection logic in Stream Audio websocket client - i.e. retrofit the fix from LMA.
+   - Clean up and streamline log formats in Websocket Server and Stream Audio client
+   - Reduce the amount of logs due to health checks. Health check is done every 10 seconds, logging status for each health check which creates huge amounts of log records in cloudwatch. To fix this - Log first health check message from each new source and then log every 2 minutes (interval controlled by a new environment variable which is not exposed in cloudformation, but can be directly updated in the ECS configuration by developers).
+   - Fix authentication code in Websocket client utility (utilites/webosocket-client)
+
 ## [0.8.11] - 2024-02-21
 ### Fixed
 - Added custom resource in the lca-websocket-stack that sets the ECS cluster's desired service count to zero, then deletes the service, before deleting the cluster. This will prevent errors in deleting the stack when disabling websockets or deleting LCA.
@@ -356,7 +372,8 @@ and this project adheres to [Semantic Versioning](https://semver.org/spec/v2.0.0
 
 - Initial release
 
-[Unreleased]: https://github.com/aws-samples/amazon-transcribe-live-call-analytics/compare/v0.8.11...develop
+[Unreleased]: https://github.com/aws-samples/amazon-transcribe-live-meeting-assistant/compare/main...develop
+[0.8.12]: https://github.com/aws-samples/amazon-transcribe-live-call-analytics/compare/v0.8.11...v0.8.12
 [0.8.11]: https://github.com/aws-samples/amazon-transcribe-live-call-analytics/compare/v0.8.10...v0.8.11
 [0.8.10]: https://github.com/aws-samples/amazon-transcribe-live-call-analytics/compare/v0.8.9...v0.8.10
 [0.8.9]: https://github.com/aws-samples/amazon-transcribe-live-call-analytics/compare/v0.8.8...v0.8.9

README.md

@@ -128,7 +128,7 @@ To get LCA up and running in your own AWS account, follow these steps (if you do
        **Amazon Transcribe Configuration**
    22. `Enable Partial Transcripts` - Enable partial transcripts to receive low latency evolving transcriptions for each conversation turn.
    23. `Transcribe API mode` - Set the default API mode for Transcribe. Set to 'analytics' to use the Amazon Transcribe Real-time Call Analytics service, used to support call categories and alerts, call summarization, and PCA integration.
-   24. `Enable Content Redaction for Transcripts` - Enable content redaction from Amazon Transcribe transcription output. **NOTE:** Content redaction is only available when using the English language (en-US). This parameter is ignored when not using the English language
+   24. `Enable Content Redaction for Transcripts` - Enable content redaction from Amazon Transcribe transcription output. **NOTE:** Content redaction is only available when using English (en-US, en-GB, en-AU) or Spanish (es-US). This parameter is ignored when not using other languages.
    25. `Language for Transcription` - Language code to be used for Amazon Transcribe
    26. `Content Redaction Type for Transcription` - Type of content redaction from Amazon Transcribe transcription output
    27. `Transcription PII Redaction Entity Types` - Select the PII entity types you want to identify or redact. Remove the values that you don't want to redact from the default. _DO NOT ADD CUSTOM VALUES HERE_.

VERSION

@@ -1 +1 @@
-0.8.11
+0.8.12

lca-ai-stack/VERSION

@@ -1 +1 @@
-0.8.11
+0.8.12

lca-ai-stack/deployment/lca-ai-stack.yaml

@@ -285,7 +285,7 @@ Parameters:
 
   BootstrapVersion:
     Type: String
-    Default: 0.8.1111
+    Default: 0.8.12
     Description: >
       Artifacts version (semver). Used to point to a specific release in the S3
       bootstrap bucket

lca-ai-stack/samconfig.toml

@@ -8,20 +8,20 @@ use_container = true
 # shared account
 [shared.global.parameters]
 s3_bucket = "lca-artifacts-253873381732-us-east-1"
-s3_prefix = "artifacts/lca/0.8.1110"
+s3_prefix = "artifacts/lca/0.8.12"
 
 [shared.deploy.parameters]
 stack_name = "LiveCallAnalytics"
 s3_bucket = "lca-artifacts-253873381732-us-east-1"
-s3_prefix = "artifacts/lca/0.8.1110"
+s3_prefix = "artifacts/lca/0.8.12"
 region = "us-east-1"
 fail_on_empty_changeset = false
 confirm_changeset = true
 capabilities = "CAPABILITY_IAM CAPABILITY_AUTO_EXPAND"
 parameter_overrides = [
   "BootstrapBucketBaseName=lca-artifacts-253873381732",
   "BootstrapS3Prefix=artifacts/lca",
-  "BootstrapVersion=0.8.1110",
+  "BootstrapVersion=0.8.12",
   "S3BucketName=shared-ai-for-chime-vc-audio",
   "IsContentRedactionEnabled=true",
   "IsSentimentAnalysisEnabled=true",
@@ -32,4 +32,4 @@ parameter_overrides = [
 
 [shared.package.parameters]
 s3_bucket = "lca-artifacts-253873381732-us-east-1"
-s3_prefix = "artifacts/lca/0.8.1110"
+s3_prefix = "artifacts/lca/0.8.12"

lca-ai-stack/source/ui/package-lock.json

@@ -1,6 +1,6 @@
 {
   "name": "lca-ui",
-  "version": "0.8.1111",
+  "version": "0.8.12",
   "private": true,
   "dependencies": {
     "@aws-amplify/ui-components": "^1.9.6",
@@ -77,4 +77,4 @@
       "<roodDir>/node_modules/(?!@awsui)"
     ]
   }
-}
+}

lca-ai-stack/source/ui/package.json

@@ -36,10 +36,12 @@ const StreamAudio = () => {
   const [micInputOption, setMicInputOption] = useState({ label: 'AGENT', value: 'agent' });
 
   const getSocketUrl = useCallback(() => {
-    console.log('Trying to resolve websocket url...');
+    console.log(`DEBUG - [${new Date().toISOString()}]: Trying to resolve websocket url...`);
     return new Promise((resolve) => {
       if (settings.WSEndpoint) {
-        console.log(`Resolved Websocket URL to ${settings.WSEndpoint}`);
+        console.log(`
+          DEBUG - [${new Date().toISOString()}]: Resolved Websocket URL to ${settings.WSEndpoint}
+        `);
         resolve(settings.WSEndpoint);
       }
     });
@@ -50,14 +52,21 @@ const StreamAudio = () => {
       authorization: `Bearer ${JWT_TOKEN}`,
     },
     onOpen: (event) => {
-      console.log(event);
+      console.log(`
+        DEBUG - [${new Date().toISOString()}]: Websocket onOpen Event: ${JSON.stringify(event)}
+      `);
     },
     onClose: (event) => {
-      console.log(event);
+      console.log(`
+        DEBUG - [${new Date().toISOString()}]: Websocket onClose Event: ${JSON.stringify(event)}
+      `);
     },
     onError: (event) => {
-      console.log(event);
+      console.log(`
+        DEBUG - [${new Date().toISOString()}]: Websocket onError Event: ${JSON.stringify(event)}
+      `);
     },
+    shouldReconnect: () => true,
   });
 
   const handleCallIdChange = (e) => {
@@ -110,6 +119,8 @@ const StreamAudio = () => {
   };
 
   const stopRecording = async () => {
+    console.log(`DEBUG - [${new Date().toISOString()}]: Stopping recording...`);
+
     if (audioProcessor.current) {
       audioProcessor.current.port.postMessage({
         message: 'UPDATE_RECORDING_STATE',
@@ -130,11 +141,17 @@ const StreamAudio = () => {
         console.log('AudioContext closed.');
       });
     } else {
-      console.log('no media recorder available to stop');
+      console.log(`
+        DEBUG - [${new Date().toISOString()}]: Error trying to stop recording. AudioWorklet Processor node is not active.
+      `);
       setRecording(false);
     }
     if (streamingStarted && !recording) {
       callMetaData.callEvent = 'END';
+      // eslint-disable-next-line prettier/prettier
+      console.log(`
+        DEBUG - [${new Date().toISOString()}]: Send Call END msg: ${JSON.stringify(callMetaData)}
+      `);
       sendMessage(JSON.stringify(callMetaData));
       setStreamingStarted(false);
       setCallMetaData({
@@ -145,6 +162,10 @@ const StreamAudio = () => {
   };
 
   const startRecording = async () => {
+    console.log(`
+      DEBUG - [${new Date().toISOString()}]: Start Recording and Streaming Audio to Websocket server.
+    `);
+
     try {
       audioContext.current = new window.AudioContext();
       displayStream.current = await window.navigator.mediaDevices.getDisplayMedia({
@@ -162,6 +183,10 @@ const StreamAudio = () => {
       callMetaData.samplingRate = SOURCE_SAMPLING_RATE;
 
       callMetaData.callEvent = 'START';
+      // eslint-disable-next-line prettier/prettier
+      console.log(`
+        DEBUG - [${new Date().toISOString()}]: Send Call START msg: ${JSON.stringify(callMetaData)}
+      `);
       sendMessage(JSON.stringify(callMetaData));
       setStreamingStarted(true);
 
@@ -177,15 +202,23 @@ const StreamAudio = () => {
       monoMicSource.connect(channelMerger.current, 0, 0);
       monoDisplaySource.connect(channelMerger.current, 0, 1);
 
+      console.log(`
+        DEBUG - [${new Date().toISOString()}]: Registering and adding AudioWorklet processor to capture audio
+      `);
+
       try {
         await audioContext.current.audioWorklet.addModule('./worklets/recording-processor.js');
       } catch (error) {
-        console.log(`Add module error ${error}`);
+        console.log(`
+          DEBUG - [${new Date().toISOString()}]: Error registering AudioWorklet processor: ${error}
+        `);
       }
 
       audioProcessor.current = new AudioWorkletNode(audioContext.current, 'recording-processor');
       audioProcessor.current.port.onmessageerror = (error) => {
-        console.log(`Error receving message from worklet ${error}`);
+        console.log(`
+          DEBUG - [${new Date().toISOString()}]: Error receving message from worklet ${error}
+        `);
       };
       audioProcessor.current.port.onmessage = (event) => {
         // this is pcm audio

lca-ai-stack/source/ui/src/components/stream-audio/StreamAudio.jsx

@@ -67,7 +67,7 @@ Parameters:
     Default: 'false'
     Description: >-
       Enable content redaction from Amazon Transcribe transcription output. This is only used when
-      the 'en-US' language is selected in the TranscribeLanguageCode parameter.
+      the 'en-US', 'en-AU', 'en-GB' or 'es-US' languages are selected in the TranscribeLanguageCode parameter.
     AllowedValues:
       - 'true'
       - 'false'
@@ -954,7 +954,11 @@ Metadata:
 Conditions:
   ShouldEnableContentRedaction: !And
     - !Equals [!Ref IsContentRedactionEnabled, 'true']
-    - !Equals [!Ref TranscribeLanguageCode, 'en-US']
+    - !Or 
+      - !Equals [!Ref TranscribeLanguageCode, 'en-US']
+      - !Equals [!Ref TranscribeLanguageCode, 'en-AU']
+      - !Equals [!Ref TranscribeLanguageCode, 'en-GB']
+      - !Equals [!Ref TranscribeLanguageCode, 'es-US']
   ShouldEnableLambdaHook: !Not [!Equals [!Ref SiprecLambdaHookFunctionArn, '']]
   ShouldUseCreatedVoiceConnector: !Equals [!Ref CustomVoiceConnectorId, '']