Changes done for v2.3.0.

Author: nysenthil

com.ibm.streamsx.sttgateway/CHANGELOG.md

@@ -1,5 +1,13 @@
 # Changes
 
+## v2.3.0
+* Aug/23/2021
+* Made changes in the IBMVoiceGatewaySource operator to parse the SIP invite custom header fields such as Cisco-Guid and include it in the output stream.
+* Made changes in the VgwDataRouter, VgwDataRouterToWatsonSTT and VoiceGatewayToStreamsToWatsonSTT applications to propagate the ciscoGuid value received as part of the speech metadata.
+* Changed the default speech model to the next gen model i.e. from en-US_NarrowbandModel to en-US_Telephony.
+* Changed the STT result schema to include call start date time and utterance result reception time which will help the users to see the time it took for the STT engine to return a given utterance.
+* Added ciscoGuid attribute to the call metadata schema to be useful in call recording and call replay.
+
 ## v2.2.9
 * Feb/11/2021
 * Removed the EndOfCallSignal (EOCS) output stream completely to avoid port locks and out of order processing between the binary speech data (BSD) and the EOCS tuples. Now, a single output stream will deliver both the BSD and EOCS tuples in the correct sequence for downstream processing. 

com.ibm.streamsx.sttgateway/com.ibm.streamsx.sttgateway.watson/IBMVoiceGatewaySource/IBMVoiceGatewaySource.xml

@@ -145,6 +145,10 @@
             <description>Returns an rstring value with the call start date time i.e. system clock time.</description>
             <prototype><![CDATA[rstring getCallStartDateTime()]]></prototype>
           </function>          
+          <function>
+            <description>Returns an rstring value for the SIP invite custom header Cisco-Guid.</description>
+            <prototype><![CDATA[rstring getCiscoGuid()]]></prototype>
+          </function>          
         </customOutputFunction>
       </customOutputFunctions>
 

com.ibm.streamsx.sttgateway/com.ibm.streamsx.sttgateway.watson/IBMVoiceGatewaySource/IBMVoiceGatewaySource_cpp.cgt

@@ -8,7 +8,7 @@
 /*
 ============================================================
 First created on: Sep/20/2019
-Last modified on: Feb/09/2021
+Last modified on: Aug/23/2021
 
 Please refer to the sttgateway-tech-brief.txt file in the 
 top-level directory of this toolkit to read about 
@@ -690,6 +690,7 @@ void MY_OPERATOR::on_open(websocketpp::connection_hdl hdl, bool isTlsConnection)
 	con_metadata.vgwParticipantURI = "";
 	con_metadata.vgwTenantID = "";
 	con_metadata.vgwSIPToURI = "";
+	con_metadata.ciscoGuid = "";
 	con_metadata.vgwIsCaller = false;
 	con_metadata.vgwVoiceChannelNumber = 0;
 	client_connections_map[hdl] = con_metadata;
@@ -812,13 +813,18 @@ void MY_OPERATOR::on_message(EndpointType* s, websocketpp::connection_hdl hdl,
 			// ===========================================================
 			{
 			  "action": "start",
-			  "siprecMetadata": {
-			      "vgwSessionID" : "8443572187904004@192.168.208.5",
-			      "vgwSIPCallID" : "8443572187904004@192.168.208.5",
-			      "vgwParticipantURI" : "sip:alice@10.10.10.7",
-			      "vgwIsCaller" : true
-			  }
-			}
+			  "siprecMetadata": {            
+   	   	   	  	  "vgwTenantID": "siprec-tenant",
+   	   	   	   	  "vgwSIPToURI": "sip:10.145.203.161:5060;transport=udp",
+   	   	   	   	  "vgwIsCaller": true,
+   	   	   	   	  "vgwSessionID": "fccd47f1-a44e-4060-b713-4f19c5baebcf",
+   	   	   	   	  "vgwSIPCallID": "f75e487831ff12e517ee10159b31e0d9020@10.145.203.161",
+   	   	   	   	  "vgwParticipantURI": "sip:9737046347@143.91.19.6",
+    			  "vgwSIPCustomInviteHeaders": {
+      	  	  		 "Cisco-Guid": "3502874379-4113371627-2730858090-1212179876"
+      	  	  	  }
+    		  }
+ 	 	 	}
 			// ===========================================================
 			*/
 			if (action == "start") {
@@ -828,7 +834,7 @@ void MY_OPERATOR::on_message(EndpointType* s, websocketpp::connection_hdl hdl,
 				//
 				// Look for vgwSessionId
 				try {
-					// This is an important metadata that identiies
+					// This is an important metadata that identifies
 					// every unique voice call. It must be present.
 					con_metadata.vgwSessionId = query(root, 
 						"siprecMetadata.vgwSessionID").get_value<std::string>();
@@ -924,7 +930,24 @@ void MY_OPERATOR::on_message(EndpointType* s, websocketpp::connection_hdl hdl,
 							", IBM Voice Gateway sent this message: " << msg->get_payload(), "on_message");
 					}					
 				}
-								
+
+				// Look for Cisco-Guid
+				try {
+					// It is fine even if this field is not present in the JSON message.
+					// Because, it is a SIP invite custom header. Depending on the
+					// underlying voice infrastructure and the configuration done in
+					// the IBM Voice Gateway product, this header may or may not be there.
+					con_metadata.ciscoGuid = query(root, 
+						"siprecMetadata.vgwSIPCustomInviteHeaders.Cisco-Guid").get_value<std::string>();
+				}  catch (std::exception const& e) {
+					if (vgwSessionLoggingNeeded == true) {
+						SPLAPPTRC(L_ERROR, "Operator " << operatorPhysicalName <<
+							"-->Channel " << boost::to_string(udpChannelNumber) <<
+							"-->X1 JSON parsing error when reading the field : " << e.what() <<
+							", IBM Voice Gateway sent this message: " << msg->get_payload(), "on_message");
+					}					
+				}
+
 				// If we reached this far, then we can give an 
 				// affirmative response back to the IBM Voice Gateway for 
 				// its session start message.
@@ -973,7 +996,9 @@ void MY_OPERATOR::on_message(EndpointType* s, websocketpp::connection_hdl hdl,
 						", vgwParticipantURI=" << con_metadata.vgwParticipantURI <<
 						", vgwTenantID=" << con_metadata.vgwTenantID <<
 						", vgwSIPToURI=" << con_metadata.vgwSIPToURI << 
-						".", "on_message");
+						", Cisco-Guid=" << con_metadata.ciscoGuid <<
+						". Full JSON message=" << msg->get_payload(),
+						"on_message");
 				}					
 
 				return;
@@ -1430,6 +1455,10 @@ void MY_OPERATOR::on_message(EndpointType* s, websocketpp::connection_hdl hdl,
 		  	%> 
 		  		   oTuple.set_<%=$name%>( 
 		  		   	    <%=$operation%>(con_metadata.callStartDateTime));
+		  		  <%} elsif ($operation eq "getCiscoGuid") { 
+		  	%> 
+		  		   oTuple.set_<%=$name%>( 
+		  		  		<%=$operation%>(con_metadata.ciscoGuid));
 		  		  <%}
 			}%>
 
@@ -1934,4 +1963,8 @@ SPL::rstring MY_OPERATOR::getCallStartDateTime(SPL::rstring const & callStartDat
 	return(callStartDateTime);
 }
 
+SPL::rstring MY_OPERATOR::getCiscoGuid(SPL::rstring const & ciscoGuid) {
+	return(ciscoGuid);
+}
+
 <%SPL::CodeGen::implementationEpilogue($model);%>

com.ibm.streamsx.sttgateway/com.ibm.streamsx.sttgateway.watson/IBMVoiceGatewaySource/IBMVoiceGatewaySource_h.cgt

@@ -8,7 +8,7 @@
 /*
 ============================================================
 First created on: Sep/17/2019
-Last modified on: Feb/07/2021
+Last modified on: Aug/21/2021
 ============================================================
 */
 
@@ -112,6 +112,11 @@ public:
 		std::string vgwSIPToURI;
 		// This tells if this connection carries caller's or agent's speech data.
 		bool vgwIsCaller;
+		// This field indicates the SIP invite custom header named Ciso-Guid.
+		// This field will be sent in the SIPREC metadata only if the IBM Voice Gateway
+		// product is configured to parse this field from the SIP headers.
+		std::string ciscoGuid;
+		// 
 		// This indicates the voice channel number i.e. 1 or 2.
 		// Whoever (caller or agent) sends the first round of speech data bytes will
 		// get assigned a voice channel of 1. The next one to follow will get
@@ -207,6 +212,7 @@ private:
 	int32_t getVoiceChannelNumber(int32_t const & voiceChannelNumber);
 	std::string getAgentOrCallerPhoneNumber(std::string const & vgwSessionId, bool const & vgwIsCaller);
 	SPL::rstring getCallStartDateTime(SPL::rstring const & callStartDateTime);
+	SPL::rstring getCiscoGuid(SPL::rstring const & ciscoGuid);
 }; 
 
 <%SPL::CodeGen::headerEpilogue($model);%>

com.ibm.streamsx.sttgateway/info.xml

@@ -14,7 +14,7 @@
 
     **Note:** This toolkit requires c++11 support.
     </description>
-    <version>2.2.9</version>
+    <version>2.3.0</version>
     <requiredProductVersion>4.2.1.6</requiredProductVersion>
   </identity>
   <dependencies>

samples/STTGatewayUtils/com.ibm.streamsx.sttgateway.utils/STTGatewayUtils.spl

@@ -8,7 +8,7 @@
 /*
 ==============================================
 First created on: Nov/24/2020
-Last modified on: Feb/09/2021
+Last modified on: Aug/21/2021
 
 This is a utility composite that will get used in the following applications.
 
@@ -20,7 +20,7 @@ This is a utility composite that will get used in the following applications.
 
 namespace com.ibm.streamsx.sttgateway.utils;
 
-// It is a dummpy Main composite.
+// It is a dummy Main composite.
 // Major goal of this project is to have reusable SPL, C++ and Java functions snd types.
 // Code for the C++ native functions can be found in the impl/include directory of this project.
 //
@@ -44,6 +44,7 @@ namespace com.ibm.streamsx.sttgateway.utils;
 // rstring callStartDateTime --> Call start date time i.e. system clock time.
 // rstring callerPhoneNumber --> Details about the caller's phone number.
 // rstring agentPhoneNumber --> Details about the agent's phone number.
+// rstring ciscoGuid --> Value of the SIP invite custom header Cisco-Guid if present.
 // int32 speechDataFragmentCnt --> Number of fragments (tuples) emitted so far on a given channel (customer or agent) for a given vgwSessionId.
 // int32 totalSpeechDataBytesReceived --> Number of speech bytes received so far on a given channel (customer or agent) for a given vgwSessionId.
 // int32 int32 speechProcessorId --> Id of the speech processor job that is analyzing the speech data.
@@ -53,8 +54,9 @@ type BinarySpeech_t = blob speech, rstring vgwSessionId, boolean isCustomerSpeec
 	int32 vgwVoiceChannelNumber, boolean endOfCallSignal, 
 	rstring id, rstring callStartDateTime, 
 	rstring callerPhoneNumber, rstring agentPhoneNumber, 
-	int32 speechDataFragmentCnt, int32 totalSpeechDataBytesReceived, 
-	int32 speechProcessorId, int32 speechEngineId, int32 speechResultProcessorId;
+	rstring ciscoGuid, int32 speechDataFragmentCnt, 
+	int32 totalSpeechDataBytesReceived, int32 speechProcessorId, 
+	int32 speechEngineId, int32 speechResultProcessorId;
 
 // The following schema will be for the data being sent here by the
 // VgwDataRouter application. It sends us raw binary data which 
@@ -68,7 +70,7 @@ type DataFromVgwRouter_t = int32 msgType, blob payload;
 // voice call.
 type CallMetaData_t = rstring vgwSessionId, boolean isCustomerSpeechData, 
 	int32 vgwVoiceChannelNumber, rstring callerPhoneNumber, 
-	rstring agentPhoneNumber;
+	rstring agentPhoneNumber, rstring ciscoGuid;
 
 // The following schema is for the call recording feature where we will
 // store the call speech data for a specific voice channel of a given

samples/STTGatewayUtils/info.xml

@@ -4,7 +4,7 @@
  <info:identity>
    <info:name>STTGatewayUtils</info:name>
    <info:description>A utility composite used by the VgwDataRouter and the speech processor applications.</info:description>
-   <info:version>1.0.1</info:version>
+   <info:version>1.0.2</info:version>
    <info:requiredProductVersion>4.2.1.6</info:requiredProductVersion>
  </info:identity>
  <info:dependencies/>

samples/VgwDataRouter/com.ibm.streamsx.sttgateway.sample/VgwDataRouter.spl

@@ -8,7 +8,7 @@
 /*
 ==============================================
 First created on: Nov/24/2020
-Last modified on: Feb/10/2021
+Last modified on: Aug/21/2021
 
 A) What does this example application do?
    --------------------------------------
@@ -83,7 +83,7 @@ deploying them. Please refer to the streamsx.sttgateway documentation section ti
 2) Once you are sure that the IBM Voice Gateway can send the speech data to this
 application, then you can give the following command to deploy this application in distributed mode.
 
-There are certain importat submisstion time parameters that are required while
+There are certain important submission time parameters that are required while
 launching this application.
 
 Mandatory submission time parameter(s) for the VgwDataRouter application:
@@ -249,7 +249,7 @@ public composite VgwDataRouter {
 		// Is a password needed for the private key in the certificate file?
 		expression<rstring> $certificatePassword : 
 			getSubmissionTimeValue("certificatePassword", "");
-		// Do you want to specifiy a file name that contains the public certificates of
+		// Do you want to specify a file name that contains the public certificates of
 		// the trusted client(s). If this file name is not empty, then the
 		// WebSocketSink operator will perform a client (mutual) authentication.
 		expression<rstring> $trustedClientCertificateFileName :
@@ -285,7 +285,7 @@ public composite VgwDataRouter {
         // We will allow different speech processor jobs with their own
         // unique ids to connect to this application for getting the
         // speech data to get routed for processing. Such speech processor jobs will
-        // use their respectivce unique speech processor id at the end end of the
+        // use their respective unique speech processor id at the end end of the
         // URL as context path. Hence, we will prepare the WebSocketSink operator
         // invoked in this application to allow a set of context paths.
         // e-g: wss://MyHostName:8444/14  This URL must be used by a 
@@ -331,8 +331,8 @@ public composite VgwDataRouter {
 		// Is IBM Voice Gateway message exchange logging needed for debugging?
 		expression<boolean> $vgwSessionLoggingNeeded : 
 			(boolean)getSubmissionTimeValue("vgwSessionLoggingNeeded", "false");
-		// Under some cicumstances, if the IBMVoiceGatewaySource operator sends 
-		// only one EOCS (End Of Call Sigmal) tuple instead of two as required for
+		// Under some circumstances, if the IBMVoiceGatewaySource operator sends 
+		// only one EOCS (End Of Call Signal) tuple instead of two as required for
 		// the two voice channels, that may eventually cause the application logic
 		// below not be able to release the speech processor jobs properly at the end of
 		// a voice call for a given VGW session id. We have seen it in certain
@@ -450,7 +450,7 @@ public composite VgwDataRouter {
 				// Initial delay before generating the very first tuple.
 				// This is a one time delay when this operator starts up.
 				// This delay should give sufficient time for the
-				// speech processor jpbs to come up and be ready to
+				// speech processor jobs to come up and be ready to
 				// receive the speech data tuples sent by this operator.
 				initDelay: $initDelayBeforeSendingDataToSpeechProcessors;
 				vgwLiveMetricsUpdateNeeded: $vgwLiveMetricsUpdateNeeded;
@@ -467,6 +467,7 @@ public composite VgwDataRouter {
 					vgwVoiceChannelNumber = getVoiceChannelNumber(),
 					callerPhoneNumber = getCallerPhoneNumber(),
 					agentPhoneNumber = getAgentPhoneNumber(),
+					ciscoGuid = getCiscoGuid(), 
 					speechDataFragmentCnt = getTupleCnt(),
 					totalSpeechDataBytesReceived = getTotalSpeechDataBytesReceived();
 
@@ -749,7 +750,7 @@ public composite VgwDataRouter {
 						} else {
 							// Flag an error only when the user configured for two
 							// EOCS tuples to be received for considering a voice call
-							// as complted.
+							// as completed.
 							if ($numberOfEocsNeededForVoiceCallCompletion == 2) {
 								appTrc(Trace.error, 
 									"_YYYYY No speech processor id is available at this time for the " +
@@ -767,10 +768,10 @@ public composite VgwDataRouter {
 
 		// Invoke one or more instances of the WebSocketSink operator.
 		// This operator listens on a TLS port and is configured with
-		// multiple URL context paths to aceept connections from the
+		// multiple URL context paths to accept connections from the
 		// remote speech processor jobs that will each connect to a
-		// particular context path expresded by their speech processor id.
-		// It then routes the voice cslls' speech data and EOCS to the
+		// particular context path expressed by their speech processor id.
+		// It then routes the voice calls' speech data and EOCS to the
 		// corresponding speech processor job.
 		// For now, we will have a parallel width of 1.
 		// If it doesn't perform well, we can think of increasing the 

samples/VgwDataRouter/etc/call-replay-test-data-generator.tar.gz

@@ -4,13 +4,13 @@
  <info:identity>
    <info:name>VgwDataRouter</info:name>
    <info:description>Example that shows how to route VGW speech data to different Speech processor jobs</info:description>
-   <info:version>1.0.2</info:version>
+   <info:version>1.0.3</info:version>
    <info:requiredProductVersion>4.2.1.6</info:requiredProductVersion>
  </info:identity>
  <info:dependencies>
    <info:toolkit>
      <common:name>com.ibm.streamsx.sttgateway</common:name>
-     <common:version>[2.2.9,7.0.0]</common:version>
+     <common:version>[2.3.0,7.0.0]</common:version>
    </info:toolkit>
    <info:toolkit>
      <common:name>com.ibm.streamsx.websocket</common:name>