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99 lines
3.9 KiB
Protocol Buffer
99 lines
3.9 KiB
Protocol Buffer
// Copyright 2025 Signal Messenger, LLC
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// SPDX-License-Identifier: AGPL-3.0-only
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syntax = "proto3";
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option java_multiple_files = true;
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package signalservice;
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message SubmitCallQualitySurveyRequest {
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// Indicates whether the caller was generally satisfied with the quality of
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// the call
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bool user_satisfied = 1;
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// A list of call quality issues selected by the caller
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repeated string call_quality_issues = 2;
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// A free-form description of any additional issues as written by the caller
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optional string additional_issues_description = 3;
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// A URL for a set of debug logs associated with the call if the caller chose
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// to submit debug logs
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optional string debug_log_url = 4;
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// The time at which the call started in milliseconds since the epoch
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int64 start_timestamp = 5;
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// The time at which the call ended in milliseconds since the epoch
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int64 end_timestamp = 6;
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// The type of call; note that direct voice calls can become video calls and
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// vice versa, and this field indicates which mode was selected at call
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// initiation time. At the time of writing, expected call types are
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// "direct_voice", "direct_video", "group", and "call_link".
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string call_type = 7;
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// Indicates whether the call completed without error or if it terminated
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// abnormally
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bool success = 8;
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// A client-defined, but human-readable reason for call termination
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string call_end_reason = 9;
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// The median round-trip time, measured in milliseconds, for STUN/ICE packets
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// (i.e. connection maintenance and establishment)
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optional float connection_rtt_median = 10;
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// The median round-trip time, measured in milliseconds, for RTP/RTCP packets
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// for audio streams
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optional float audio_rtt_median = 11;
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// The median round-trip time, measured in milliseconds, for RTP/RTCP packets
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// for video streams
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optional float video_rtt_median = 12;
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// The median jitter for audio streams, measured in milliseconds, for the
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// duration of the call as measured by the client submitting the survey
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optional float audio_recv_jitter_median = 13;
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// The median jitter for video streams, measured in milliseconds, for the
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// duration of the call as measured by the client submitting the survey
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optional float video_recv_jitter_median = 14;
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// The median jitter for audio streams, measured in milliseconds, for the
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// duration of the call as measured by the remote endpoint in the call (either
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// the peer of the client submitting the survey in a direct call or the SFU in
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// a group call)
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optional float audio_send_jitter_median = 15;
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// The median jitter for video streams, measured in milliseconds, for the
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// duration of the call as measured by the remote endpoint in the call (either
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// the peer of the client submitting the survey in a direct call or the SFU in
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// a group call)
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optional float video_send_jitter_median = 16;
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// The fraction of audio packets lost over the duration of the call as
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// measured by the client submitting the survey
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optional float audio_recv_packet_loss_fraction = 17;
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// The fraction of video packets lost over the duration of the call as
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// measured by the client submitting the survey
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optional float video_recv_packet_loss_fraction = 18;
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// The fraction of audio packets lost over the duration of the call as
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// measured by the remote endpoint in the call (either the peer of the client
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// submitting the survey in a direct call or the SFU in a group call)
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optional float audio_send_packet_loss_fraction = 19;
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// The fraction of video packets lost over the duration of the call as
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// measured by the remote endpoint in the call (either the peer of the client
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// submitting the survey in a direct call or the SFU in a group call)
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optional float video_send_packet_loss_fraction = 20;
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// Machine-generated telemetry from the call; this is a serialized protobuf
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// entity generated (and, critically, explained to the user!) by the calling
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// library
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optional bytes call_telemetry = 21;
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}
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