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Call quality survey integration.

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This commit is contained in:
Alex Hart
2025-11-26 13:51:03 -04:00
committed by jeffrey-signal
parent 804f479cb0
commit 54fb7ff23f
24 changed files with 772 additions and 159 deletions
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21
libsignal-service/src/main/java/org/whispersystems/signalservice/api/calling/CallingApi.kt
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@@ -7,6 +7,7 @@ package org.whispersystems.signalservice.api.calling
import org.signal.libsignal.zkgroup.calllinks.CreateCallLinkCredentialRequest
import org.signal.libsignal.zkgroup.calllinks.CreateCallLinkCredentialResponse
import org.signal.storageservice.protos.calls.quality.SubmitCallQualitySurveyRequest
import org.whispersystems.signalservice.api.NetworkResult
import org.whispersystems.signalservice.api.messages.calls.CallingResponse
import org.whispersystems.signalservice.api.messages.calls.TurnServerInfo
@@ -17,6 +18,7 @@ import org.whispersystems.signalservice.internal.push.CreateCallLinkAuthRequest
import org.whispersystems.signalservice.internal.push.CreateCallLinkAuthResponse
import org.whispersystems.signalservice.internal.push.GetCallingRelaysResponse
import org.whispersystems.signalservice.internal.push.PushServiceSocket
import org.whispersystems.signalservice.internal.putCustom
import org.whispersystems.signalservice.internal.websocket.WebSocketRequestMessage
/**
@@ -24,9 +26,28 @@ import org.whispersystems.signalservice.internal.websocket.WebSocketRequestMessa
*/
class CallingApi(
private val auth: SignalWebSocket.AuthenticatedWebSocket,
private val unAuth: SignalWebSocket.UnauthenticatedWebSocket,
private val pushServiceSocket: PushServiceSocket
) {
/**
* Submit call quality information (with the user's permission) to the server on an unauthenticated channel.
*
* PUT /v1/call_quality_survey
* - 204: The survey response was submitted successfully
* - 422: The survey response could not be parsed
* - 429: Too many attempts, try after Retry-After seconds.
*/
fun submitCallQualitySurvey(request: SubmitCallQualitySurveyRequest): NetworkResult<Unit> {
val webSocketRequestMessage = WebSocketRequestMessage.putCustom(
path = "/v1/call_quality_survey",
body = request.encode(),
headers = mapOf("Content-Type" to "application/octet-stream")
)
return NetworkResult.fromWebSocketRequest(unAuth, webSocketRequestMessage)
}
/**
* Get 1:1 relay addresses in IpV4, Ipv6, and URL formats.
*

111
libsignal-service/src/main/protowire/CallQualitySurvey.proto
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@@ -1,25 +1,100 @@
/*
* Copyright 2025 Signal Messenger, LLC
* SPDX-License-Identifier: AGPL-3.0-only
*/
// Copyright 2025 Signal Messenger, LLC
// SPDX-License-Identifier: AGPL-3.0-only
syntax = "proto3";
option java_package = "org.signal.storageservice.protos.calls.quality";
option java_multiple_files = true;
package org.signal.storageservice.protos.calls.quality;
message SubmitCallQualitySurveyRequest {
optional bool user_satisfied = 1;
repeated string call_quality_issues = 2;
// Indicates whether the caller was generally satisfied with the quality of
// the call
bool user_satisfied = 1;
// A list of call quality issues selected by the caller
repeated string call_quality_issues = 2;
// A free-form description of any additional issues as written by the caller
optional string additional_issues_description = 3;
optional string debug_log_url = 4;
optional int64 start_timestamp = 5;
optional int64 end_timestamp = 6;
optional string call_type = 7;
optional bool success = 8;
optional string call_end_reason = 9;
optional float rtt_median = 10;
optional float jitter_median = 11;
optional float packet_loss_fraction = 12;
optional bytes call_telemetry = 13;
}
// A URL for a set of debug logs associated with the call if the caller chose
// to submit debug logs
optional string debug_log_url = 4;
// The time at which the call started in microseconds since the epoch (see
// https://cloud.google.com/bigquery/docs/reference/standard-sql/data-types#timestamp_type)
int64 start_timestamp = 5;
// The time at which the call ended in microseconds since the epoch (see
// https://cloud.google.com/bigquery/docs/reference/standard-sql/data-types#timestamp_type)
int64 end_timestamp = 6;
// The type of call; note that direct voice calls can become video calls and
// vice versa, and this field indicates which mode was selected at call
// initiation time. At the time of writing, expected call types are
// "direct_voice", "direct_video", "group", and "call_link".
string call_type = 7;
// Indicates whether the call completed without error or if it terminated
// abnormally
bool success = 8;
// A client-defined, but human-readable reason for call termination
string call_end_reason = 9;
// The median round-trip time, measured in milliseconds, for STUN/ICE packets
// (i.e. connection maintenance and establishment)
optional float connection_rtt_median = 10;
// The median round-trip time, measured in milliseconds, for RTP/RTCP packets
// for audio streams
optional float audio_rtt_median = 11;
// The median round-trip time, measured in milliseconds, for RTP/RTCP packets
// for video streams
optional float video_rtt_median = 12;
// 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 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 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 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 = 21;
}