Add 5 Jitter-related and 2 RTP-related parameters to call quality
metrics in telephony .proto

Bug: 201960858

Test: builds

Change-Id: I816db62cd628cb9ff0572a1ad2144c646265f2b6
diff --git a/proto/src/telephony.proto b/proto/src/telephony.proto
index 0d9ee80..6d3d711 100644
--- a/proto/src/telephony.proto
+++ b/proto/src/telephony.proto
@@ -2297,6 +2297,28 @@
       // after call is connected
       optional bool tx_silence_detected = 14;
 
+      // the number of Voice frames sent by jitter buffer to audio
+      optional int32 voice_frames = 15;
+
+      // the number of NO_DATA frames sent by jitter buffer to audio
+      optional int32 no_data_frames = 16;
+
+      // the number of RTP voice packets dropped by jitter buffer
+      optional int32 rtp_dropped_packets = 17;
+
+      // the minimum playout delay in the reporting interval, in milliseconds
+      optional int64 min_playout_delay_millis = 18;
+
+      // the maximum playout delay in the reporting interval, in milliseconds
+      optional int64 max_playout_delay_millis = 19;
+
+      // the total number of RTP SID packets received by this device
+      // for an ongoing call
+      optional int32 rx_rtp_sid_packets = 20;
+
+      // the total number of RTP duplicate packets received by this device
+      // for an ongoing call
+      optional int32 rtp_duplicate_packets = 21;
     }
 
     message CallQualitySummary {
diff --git a/src/java/com/android/internal/telephony/metrics/TelephonyMetrics.java b/src/java/com/android/internal/telephony/metrics/TelephonyMetrics.java
index 282cdc8..f3f117a 100644
--- a/src/java/com/android/internal/telephony/metrics/TelephonyMetrics.java
+++ b/src/java/com/android/internal/telephony/metrics/TelephonyMetrics.java
@@ -2198,6 +2198,13 @@
             cq.rtpInactivityDetected = callQuality.isRtpInactivityDetected();
             cq.rxSilenceDetected = callQuality.isIncomingSilenceDetectedAtCallSetup();
             cq.txSilenceDetected = callQuality.isOutgoingSilenceDetectedAtCallSetup();
+            cq.voiceFrames = callQuality.getNumVoiceFrames();
+            cq.noDataFrames = callQuality.getNumNoDataFrames();
+            cq.rtpDroppedPackets = callQuality.getNumDroppedRtpPackets();
+            cq.minPlayoutDelayMillis = callQuality.getMinPlayoutDelayMillis();
+            cq.maxPlayoutDelayMillis = callQuality.getMaxPlayoutDelayMillis();
+            cq.rxRtpSidPackets = callQuality.getNumRtpSidPacketsRx();
+            cq.rtpDuplicatePackets = callQuality.getNumRtpDuplicatePackets();
         }
         return cq;
     }