diff --git a/service/src/main/java/org/whispersystems/textsecuregcm/metrics/CallQualitySurveyManager.java b/service/src/main/java/org/whispersystems/textsecuregcm/metrics/CallQualitySurveyManager.java
index 28b22b0df..056fd0391 100644
--- a/service/src/main/java/org/whispersystems/textsecuregcm/metrics/CallQualitySurveyManager.java
+++ b/service/src/main/java/org/whispersystems/textsecuregcm/metrics/CallQualitySurveyManager.java
@@ -86,28 +86,48 @@ public class CallQualitySurveyManager {
       pubSubMessageBuilder.setDebugLogUrl(submitCallQualitySurveyRequest.getDebugLogUrl());
     }
 
-    if (submitCallQualitySurveyRequest.hasRttMedianConnection()) {
-      pubSubMessageBuilder.setRttMedianConnection(submitCallQualitySurveyRequest.getRttMedianConnection());
+    if (submitCallQualitySurveyRequest.hasConnectionRttMedian()) {
+      pubSubMessageBuilder.setConnectionRttMedian(submitCallQualitySurveyRequest.getConnectionRttMedian());
     }
 
-    if (submitCallQualitySurveyRequest.hasRttMedianMedia()) {
-      pubSubMessageBuilder.setRttMedianMedia(submitCallQualitySurveyRequest.getRttMedianMedia());
+    if (submitCallQualitySurveyRequest.hasAudioRttMedian()) {
+      pubSubMessageBuilder.setAudioRttMedian(submitCallQualitySurveyRequest.getAudioRttMedian());
     }
 
-    if (submitCallQualitySurveyRequest.hasJitterMedianRecv()) {
-      pubSubMessageBuilder.setJitterMedianRecv(submitCallQualitySurveyRequest.getJitterMedianRecv());
+    if (submitCallQualitySurveyRequest.hasVideoRttMedian()) {
+      pubSubMessageBuilder.setVideoRttMedian(submitCallQualitySurveyRequest.getVideoRttMedian());
     }
 
-    if (submitCallQualitySurveyRequest.hasJitterMedianSend()) {
-      pubSubMessageBuilder.setJitterMedianSend(submitCallQualitySurveyRequest.getJitterMedianSend());
+    if (submitCallQualitySurveyRequest.hasAudioRecvJitterMedian()) {
+      pubSubMessageBuilder.setAudioRecvJitterMedian(submitCallQualitySurveyRequest.getAudioRecvJitterMedian());
     }
 
-    if (submitCallQualitySurveyRequest.hasPacketLossFractionRecv()) {
-      pubSubMessageBuilder.setPacketLossFractionRecv(submitCallQualitySurveyRequest.getPacketLossFractionRecv());
+    if (submitCallQualitySurveyRequest.hasVideoRecvJitterMedian()) {
+      pubSubMessageBuilder.setVideoRecvJitterMedian(submitCallQualitySurveyRequest.getVideoRecvJitterMedian());
     }
 
-    if (submitCallQualitySurveyRequest.hasPacketLossFractionSend()) {
-      pubSubMessageBuilder.setPacketLossFractionSend(submitCallQualitySurveyRequest.getPacketLossFractionSend());
+    if (submitCallQualitySurveyRequest.hasAudioSendJitterMedian()) {
+      pubSubMessageBuilder.setAudioSendJitterMedian(submitCallQualitySurveyRequest.getAudioSendJitterMedian());
+    }
+
+    if (submitCallQualitySurveyRequest.hasVideoSendJitterMedian()) {
+      pubSubMessageBuilder.setVideoSendJitterMedian(submitCallQualitySurveyRequest.getVideoSendJitterMedian());
+    }
+
+    if (submitCallQualitySurveyRequest.hasAudioRecvPacketLossFraction()) {
+      pubSubMessageBuilder.setAudioRecvPacketLossFraction(submitCallQualitySurveyRequest.getAudioRecvPacketLossFraction());
+    }
+
+    if (submitCallQualitySurveyRequest.hasVideoRecvPacketLossFraction()) {
+      pubSubMessageBuilder.setVideoRecvPacketLossFraction(submitCallQualitySurveyRequest.getVideoRecvPacketLossFraction());
+    }
+
+    if (submitCallQualitySurveyRequest.hasAudioSendPacketLossFraction()) {
+      pubSubMessageBuilder.setAudioSendPacketLossFraction(submitCallQualitySurveyRequest.getAudioSendPacketLossFraction());
+    }
+
+    if (submitCallQualitySurveyRequest.hasVideoSendPacketLossFraction()) {
+      pubSubMessageBuilder.setVideoSendPacketLossFraction(submitCallQualitySurveyRequest.getVideoSendPacketLossFraction());
     }
 
     if (submitCallQualitySurveyRequest.hasCallTelemetry()) {
diff --git a/service/src/main/proto/CallQualitySurveyPubSub.proto b/service/src/main/proto/CallQualitySurveyPubSub.proto
index f6268efcb..86a915ef5 100644
--- a/service/src/main/proto/CallQualitySurveyPubSub.proto
+++ b/service/src/main/proto/CallQualitySurveyPubSub.proto
@@ -73,32 +73,56 @@ message CallQualitySurveyResponsePubSubMessage {
 
   // The median round-trip time, measured in milliseconds, for STUN/ICE packets
   // (i.e. connection maintenance and establishment)
-  optional float rtt_median_connection = 16;
+  optional float connection_rtt_median = 16;
 
   // The median round-trip time, measured in milliseconds, for RTP/RTCP packets
-  // (i.e. call audio/video)
-  optional float rtt_median_media = 17;
+  // for audio streams
+  optional float audio_rtt_median = 17;
 
-  // The median jitter, measured in milliseconds, for the duration of the call
-  // as measured by the client submitting the survey
-  optional float jitter_median_recv = 18;
+  // The median round-trip time, measured in milliseconds, for RTP/RTCP packets
+  // for video streams
+  optional float video_rtt_median = 18;
 
-  // The median jitter, measured in milliseconds, for the duration of the call
-  // as measured by the remote endpoint in the call (either the peer of the
-  // client submitting the survey in a direct call or the SFU in a group call)
-  optional float jitter_median_send = 19;
+  // The median jitter for audio streams, measured in milliseconds, for the
+  // duration of the call as measured by the client submitting the survey
+  optional float audio_recv_jitter_median = 19;
 
-  // The fraction of all packets lost over the duration of the call as measured
-  // by the client submitting the survey
-  optional float packet_loss_fraction_recv = 20;
+  // The median jitter for video streams, measured in milliseconds, for the
+  // duration of the call as measured by the client submitting the survey
+  optional float video_recv_jitter_median = 20;
 
-  // The fraction of all packets lost over the duration of the call as measured
-  // by the remote endpoint in the call (either the peer of the client
+  // The median jitter for audio streams, measured in milliseconds, for the
+  // duration of the call as measured by the remote endpoint in the call (either
+  // the peer of the client submitting the survey in a direct call or the SFU in
+  // a group call)
+  optional float audio_send_jitter_median = 21;
+
+  // The median jitter for video streams, measured in milliseconds, for the
+  // duration of the call as measured by the remote endpoint in the call (either
+  // the peer of the client submitting the survey in a direct call or the SFU in
+  // a group call)
+  optional float video_send_jitter_median = 22;
+
+  // The fraction of audio packets lost over the duration of the call as
+  // measured by the client submitting the survey
+  optional float audio_recv_packet_loss_fraction = 23;
+
+  // The fraction of video packets lost over the duration of the call as
+  // measured by the client submitting the survey
+  optional float video_recv_packet_loss_fraction = 24;
+
+  // The fraction of audio packets lost over the duration of the call as
+  // measured by the remote endpoint in the call (either the peer of the client
   // submitting the survey in a direct call or the SFU in a group call)
-  optional float packet_loss_fraction_send = 21;
+  optional float audio_send_packet_loss_fraction = 25;
+
+  // The fraction of video packets lost over the duration of the call as
+  // measured by the remote endpoint in the call (either the peer of the client
+  // submitting the survey in a direct call or the SFU in a group call)
+  optional float video_send_packet_loss_fraction = 26;
 
   // Technical, machine-generated data about the quality and mechanics of a
   // call; this is a serialized protobuf entity generated (and, critically,
   // explained to the user!) by the calling library
-  optional bytes call_telemetry = 22;
+  optional bytes call_telemetry = 27;
 }
diff --git a/service/src/main/proto/org/signal/chat/call_quality.proto b/service/src/main/proto/org/signal/chat/call_quality.proto
index e67a06a3e..dabcb9fab 100644
--- a/service/src/main/proto/org/signal/chat/call_quality.proto
+++ b/service/src/main/proto/org/signal/chat/call_quality.proto
@@ -59,34 +59,58 @@ message SubmitCallQualitySurveyRequest {
 
   // The median round-trip time, measured in milliseconds, for STUN/ICE packets
   // (i.e. connection maintenance and establishment)
-  optional float rtt_median_connection = 10;
+  optional float connection_rtt_median = 10;
 
   // The median round-trip time, measured in milliseconds, for RTP/RTCP packets
-  // (i.e. call audio/video)
-  optional float rtt_median_media = 11;
+  // for audio streams
+  optional float audio_rtt_median = 11;
 
-  // The median jitter, measured in milliseconds, for the duration of the call
-  // as measured by the client submitting the survey
-  optional float jitter_median_recv = 12;
+  // The median round-trip time, measured in milliseconds, for RTP/RTCP packets
+  // for video streams
+  optional float video_rtt_median = 12;
 
-  // The median jitter, measured in milliseconds, for the duration of the call
-  // as measured by the remote endpoint in the call (either the peer of the
-  // client submitting the survey in a direct call or the SFU in a group call)
-  optional float jitter_median_send = 13;
+  // The median jitter for audio streams, measured in milliseconds, for the
+  // duration of the call as measured by the client submitting the survey
+  optional float audio_recv_jitter_median = 13;
 
-  // The fraction of all packets lost over the duration of the call as measured
-  // by the client submitting the survey
-  optional float packet_loss_fraction_recv = 14;
+  // The median jitter for video streams, measured in milliseconds, for the
+  // duration of the call as measured by the client submitting the survey
+  optional float video_recv_jitter_median = 14;
 
-  // The fraction of all packets lost over the duration of the call as measured
-  // by the remote endpoint in the call (either the peer of the client
+  // The median jitter for audio streams, measured in milliseconds, for the
+  // duration of the call as measured by the remote endpoint in the call (either
+  // the peer of the client submitting the survey in a direct call or the SFU in
+  // a group call)
+  optional float audio_send_jitter_median = 15;
+
+  // The median jitter for video streams, measured in milliseconds, for the
+  // duration of the call as measured by the remote endpoint in the call (either
+  // the peer of the client submitting the survey in a direct call or the SFU in
+  // a group call)
+  optional float video_send_jitter_median = 16;
+
+  // The fraction of audio packets lost over the duration of the call as
+  // measured by the client submitting the survey
+  optional float audio_recv_packet_loss_fraction = 17;
+
+  // The fraction of video packets lost over the duration of the call as
+  // measured by the client submitting the survey
+  optional float video_recv_packet_loss_fraction = 18;
+
+  // The fraction of audio packets lost over the duration of the call as
+  // measured by the remote endpoint in the call (either the peer of the client
   // submitting the survey in a direct call or the SFU in a group call)
-  optional float packet_loss_fraction_send = 15;
+  optional float audio_send_packet_loss_fraction = 19;
+
+  // The fraction of video packets lost over the duration of the call as
+  // measured by the remote endpoint in the call (either the peer of the client
+  // submitting the survey in a direct call or the SFU in a group call)
+  optional float video_send_packet_loss_fraction = 20;
 
   // Machine-generated telemetry from the call; this is a serialized protobuf
   // entity generated (and, critically, explained to the user!) by the calling
   // library
-  optional bytes call_telemetry = 16;
+  optional bytes call_telemetry = 21;
 }
 
 message SubmitCallQualitySurveyResponse {
diff --git a/service/src/test/java/org/whispersystems/textsecuregcm/metrics/CallQualitySurveyManagerTest.java b/service/src/test/java/org/whispersystems/textsecuregcm/metrics/CallQualitySurveyManagerTest.java
index fb6f01e60..67328781c 100644
--- a/service/src/test/java/org/whispersystems/textsecuregcm/metrics/CallQualitySurveyManagerTest.java
+++ b/service/src/test/java/org/whispersystems/textsecuregcm/metrics/CallQualitySurveyManagerTest.java
@@ -62,12 +62,17 @@ class CallQualitySurveyManagerTest {
 
     final byte[] telemetryBytes = TestRandomUtil.nextBytes(32);
 
-    final float rttMedianConnection = ThreadLocalRandom.current().nextFloat();
-    final float rttMedianMedia = ThreadLocalRandom.current().nextFloat();
-    final float jitterMedianRecv = ThreadLocalRandom.current().nextFloat();
-    final float jitterMedianSend = ThreadLocalRandom.current().nextFloat();
-    final float packetLossFractionRecv = ThreadLocalRandom.current().nextFloat();
-    final float packetLossFractionSend = ThreadLocalRandom.current().nextFloat();
+    final float connectionRttMedian = ThreadLocalRandom.current().nextFloat();
+    final float audioRttMedian = ThreadLocalRandom.current().nextFloat();
+    final float videoRttMedian = ThreadLocalRandom.current().nextFloat();
+    final float audioRecvJitterMedian = ThreadLocalRandom.current().nextFloat();
+    final float videoRecvJitterMedian = ThreadLocalRandom.current().nextFloat();
+    final float audioSendJitterMedian = ThreadLocalRandom.current().nextFloat();
+    final float videoSendJitterMedian = ThreadLocalRandom.current().nextFloat();
+    final float audioRecvPacketLossFraction = ThreadLocalRandom.current().nextFloat();
+    final float videoRecvPacketLossFraction = ThreadLocalRandom.current().nextFloat();
+    final float audioSendPacketLossFraction = ThreadLocalRandom.current().nextFloat();
+    final float videoSendPacketLossFraction = ThreadLocalRandom.current().nextFloat();
 
     when(asnInfoProvider.lookup(REMOTE_ADDRESS)).thenReturn(Optional.of(new AsnInfo(asn, asnRegion)));
 
@@ -82,12 +87,17 @@ class CallQualitySurveyManagerTest {
         .setCallType("direct_video")
         .setSuccess(true)
         .setCallEndReason("caller_hang_up")
-        .setRttMedianConnection(rttMedianConnection)
-        .setRttMedianMedia(rttMedianMedia)
-        .setJitterMedianRecv(jitterMedianRecv)
-        .setJitterMedianSend(jitterMedianSend)
-        .setPacketLossFractionRecv(packetLossFractionRecv)
-        .setPacketLossFractionSend(packetLossFractionSend)
+        .setConnectionRttMedian(connectionRttMedian)
+        .setAudioRttMedian(audioRttMedian)
+        .setVideoRttMedian(videoRttMedian)
+        .setAudioRecvJitterMedian(audioRecvJitterMedian)
+        .setVideoRecvJitterMedian(videoRecvJitterMedian)
+        .setAudioSendJitterMedian(audioSendJitterMedian)
+        .setVideoSendJitterMedian(videoSendJitterMedian)
+        .setAudioRecvPacketLossFraction(audioRecvPacketLossFraction)
+        .setVideoRecvPacketLossFraction(videoRecvPacketLossFraction)
+        .setAudioSendPacketLossFraction(audioSendPacketLossFraction)
+        .setVideoSendPacketLossFraction(videoSendPacketLossFraction)
         .setCallTelemetry(ByteString.copyFrom(telemetryBytes))
         .build();
 
@@ -117,12 +127,17 @@ class CallQualitySurveyManagerTest {
     assertEquals("direct_video", callQualitySurveyResponsePubSubMessage.getCallType());
     assertTrue(callQualitySurveyResponsePubSubMessage.getSuccess());
     assertEquals("caller_hang_up", callQualitySurveyResponsePubSubMessage.getCallEndReason());
-    assertEquals(rttMedianConnection, callQualitySurveyResponsePubSubMessage.getRttMedianConnection());
-    assertEquals(rttMedianMedia, callQualitySurveyResponsePubSubMessage.getRttMedianMedia());
-    assertEquals(jitterMedianRecv, callQualitySurveyResponsePubSubMessage.getJitterMedianRecv());
-    assertEquals(jitterMedianSend, callQualitySurveyResponsePubSubMessage.getJitterMedianSend());
-    assertEquals(packetLossFractionRecv, callQualitySurveyResponsePubSubMessage.getPacketLossFractionRecv());
-    assertEquals(packetLossFractionSend, callQualitySurveyResponsePubSubMessage.getPacketLossFractionSend());
+    assertEquals(connectionRttMedian, callQualitySurveyResponsePubSubMessage.getConnectionRttMedian());
+    assertEquals(audioRttMedian, callQualitySurveyResponsePubSubMessage.getAudioRttMedian());
+    assertEquals(videoRttMedian, callQualitySurveyResponsePubSubMessage.getVideoRttMedian());
+    assertEquals(audioRecvJitterMedian, callQualitySurveyResponsePubSubMessage.getAudioRecvJitterMedian());
+    assertEquals(videoRecvJitterMedian, callQualitySurveyResponsePubSubMessage.getVideoRecvJitterMedian());
+    assertEquals(audioSendJitterMedian, callQualitySurveyResponsePubSubMessage.getAudioSendJitterMedian());
+    assertEquals(videoSendJitterMedian, callQualitySurveyResponsePubSubMessage.getVideoSendJitterMedian());
+    assertEquals(audioRecvPacketLossFraction, callQualitySurveyResponsePubSubMessage.getAudioRecvPacketLossFraction());
+    assertEquals(videoRecvPacketLossFraction, callQualitySurveyResponsePubSubMessage.getVideoRecvPacketLossFraction());
+    assertEquals(audioSendPacketLossFraction, callQualitySurveyResponsePubSubMessage.getAudioSendPacketLossFraction());
+    assertEquals(videoSendPacketLossFraction, callQualitySurveyResponsePubSubMessage.getVideoSendPacketLossFraction());
     assertArrayEquals(telemetryBytes, callQualitySurveyResponsePubSubMessage.getCallTelemetry().toByteArray());
   }
 }