Struct ClientMetrics

pub struct ClientMetrics {

    pub rtc_frames_dropped: Option<u32>,
    pub rtc_frames_decoded: Option<u64>,
    pub rtc_frames_received: Option<u64>,
    pub rtc_frames_per_second: Option<u8>,
    pub rtc_freeze_count: Option<u32>,
    pub rtc_jitter_sec: Option<f64>,
    pub rtc_keyframes_decoded: Option<u32>,
    pub rtc_total_freezes_duration_sec: Option<f32>,
    pub rtc_frame_height: Option<u32>,
    pub rtc_frame_width: Option<u32>,
    pub rtc_packets_lost: Option<u32>,
    pub rtc_pli_count: Option<u32>,
    pub rtc_pause_count: Option<u32>,
    pub rtc_total_pauses_duration_sec: Option<f32>,
    pub rtc_stun_rtt_sec: Option<f32>,
}

ClientMetrics contains information regarding the state of the peer.

Fields

rtc_frames_dropped: Option<u32>

Counter of the number of WebRTC frames the client has dropped from the inbound video stream.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-framesdropped

rtc_frames_decoded: Option<u64>

Counter of the number of WebRTC frames that the client has decoded from the inbound video stream.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-freezecount

rtc_frames_received: Option<u64>

Counter of the number of WebRTC frames that the client has received from the inbound video stream.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-freezecount

rtc_frames_per_second: Option<u8>

Current number of frames being rendered in the last second. A good target is 60 frames per second, but it can fluctuate depending on network conditions.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-freezecount

rtc_freeze_count: Option<u32>

Number of times the inbound video playback has frozen. This is usually due to network conditions.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-freezecount

rtc_jitter_sec: Option<f64>

Amount of “jitter” in the inbound video stream. Network latency is the time it takes a packet to traverse the network. The amount that the latency varies is the jitter. Video latency is the time it takes to render a frame sent by the server (including network latency). A low jitter means the video latency can be reduced without impacting smooth playback. High jitter means clients will increase video latency to ensure smooth playback.

https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats-jitter

rtc_keyframes_decoded: Option<u32>

Number of “key frames” decoded in the inbound h.264 stream. A key frame is an expensive (bandwidth-wise) “full image” of the video frame. Data after the keyframe become – effectively – “diff” operations on that key frame. The Engine will only send a keyframe if required, which is an indication that some of the “diffs” have been lost, usually an indication of poor network conditions. We like this metric to understand times when the connection has had to recover.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-keyframesdecoded

rtc_total_freezes_duration_sec: Option<f32>

Number of seconds of frozen video the user has been subjected to.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalfreezesduration

rtc_frame_height: Option<u32>

The height of the inbound video stream in pixels.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-frameheight

rtc_frame_width: Option<u32>

The width of the inbound video stream in pixels.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-framewidth

rtc_packets_lost: Option<u32>

Amount of packets lost in the inbound video stream.

https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats-packetslost

rtc_pli_count: Option<u32>

Count the total number of Picture Loss Indication (PLI) packets.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-plicount

rtc_pause_count: Option<u32>

Count of the total number of video pauses experienced by this receiver.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-pausecount

rtc_total_pauses_duration_sec: Option<f32>

Count of the total number of video pauses experienced by this receiver.

https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalpausesduration

rtc_stun_rtt_sec: Option<f32>

Total duration of pauses in seconds.

This is the “ping” between the client and the STUN server. Not to be confused with the E2E RTT documented here

https://www.w3.org/TR/webrtc-stats/#dom-rtcicecandidatepairstats-currentroundtriptime

Trait Implementations

impl Clone for ClientMetrics

fn clone(&self) -> ClientMetrics

Returns a duplicate of the value.

const fn clone_from(&mut self, source: &Self)

Performs copy-assignment from source.

impl Debug for ClientMetrics

fn fmt(&self, f: &mut Formatter<'_>) -> Result

Formats the value using the given formatter.

impl Default for ClientMetrics

fn default() -> ClientMetrics

Returns the “default value” for a type.

impl<'de> Deserialize<'de> for ClientMetrics

fn deserialize<__D>(__deserializer: __D) -> Result<Self, __D::Error>

where __D: Deserializer<'de>,

Deserialize this value from the given Serde deserializer.

impl JsonSchema for ClientMetrics

fn schema_name() -> String

The name of the generated JSON Schema.

fn schema_id() -> Cow<'static, str>

Returns a string that uniquely identifies the schema produced by this type.

fn json_schema(generator: &mut SchemaGenerator) -> Schema

Generates a JSON Schema for this type.

fn is_referenceable() -> bool

Whether JSON Schemas generated for this type should be re-used where possible using the $ref keyword.

impl PartialEq for ClientMetrics

fn eq(&self, other: &ClientMetrics) -> bool

Tests for self and other values to be equal, and is used by ==.

const fn ne(&self, other: &Rhs) -> bool

Tests for !=. The default implementation is almost always sufficient, and should not be overridden without very good reason.

impl Serialize for ClientMetrics

fn serialize<__S>(&self, __serializer: __S) -> Result<__S::Ok, __S::Error>

where __S: Serializer,

Serialize this value into the given Serde serializer.

impl StructuralPartialEq for ClientMetrics