lib/webrtc

Peer-connection side of the WebRTC transport.

The server offers and the client answers. The offer arrives through lib/signaling.js; the answer is munged before it becomes the local description (sps-pps-idr-in-keyframe=1 on the H.264 line, on the Opus line stereo=1 plus a minptime matching the server's a=ptime, which is how audio frames shorter than 10 ms get through, and the surround multiopus taken where it is offered, takeMultiopus) and ICE candidates are exchanged the same way, non-relay ones dropped when forceTurn is set. The server's video and audio arrive as media tracks on the given element; the audio and video m-lines it offers recvonly are reserved as sendonly transceivers for the microphone and the webcam, which are attached later with replaceTrack and no renegotiation.

Everything else rides the two data channels the server creates: input and control upstream through sendDataChannelMessage on the ordered input channel, coalesced pointer motion through sendMotionMessage on the unordered pointer channel, JSON messages downstream, routed by type to the on* callbacks (pipeline, stream_info, stream_stats, cursor, system, ping, latency_measurement, server_settings, display_settings, display_config_update, cc_rate, and clipboard-msg*). Either side may gzip a message once the _gz,1 handshake has been exchanged; the multipart clipboard, server_settings and display_settings kinds keep their arrival order across asynchronous inflation, the rest route as soon as they are readable.

ClassesCopy Anchor Link

WebRTCClientCopy Anchor Link

Defined in: lib/webrtc.js:106

WebRTC client: one peer connection plus its data channel.

Callbacks are assigned as properties: onstatus, ondebug, and onerror receive messages, onconnectionstatechange the peer connection state, ondatachannelopen and ondatachannelclose nothing, onplaystreamrequired fires when autoplay was refused and a user gesture is needed, and onclipboardcontent, oncursorchange, onsystemaction, onstreaminfo, onstreamstats, onlatencymeasurement, onserversettings, ondisplaysettings, ondisplayconfig, and onccrate receive the payload of the data channel message of the same kind, onconnection the server's verdict on this page's connection, true while it is poor, and onpagevisibility the server's ask for this page's visibility (page_visibility).

ConstructorsCopy Anchor Link

ConstructorCopy Anchor Link

new WebRTCClient(
   signaling, 
   element, 
   peer_id
): WebRTCClient;

Defined in: lib/webrtc.js:112

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

WebRTCClient

PropertiesCopy Anchor Link

signalingCopy Anchor Link

signaling: WebRTCSignaling;

Defined in: lib/webrtc.js:114

elementCopy Anchor Link

element: HTMLVideoElement;

Defined in: lib/webrtc.js:117

peer_idCopy Anchor Link

peer_id: number;

Defined in: lib/webrtc.js:120

forceTurnCopy Anchor Link

forceTurn: boolean;

Defined in: lib/webrtc.js:123

Accept only relay ICE candidates and force iceTransportPolicy to relay.

rtcPeerConfigCopy Anchor Link

rtcPeerConfig: any;

Defined in: lib/webrtc.js:126

Configuration handed to RTCPeerConnection.

peerConnectionCopy Anchor Link

peerConnection: RTCPeerConnection;

Defined in: lib/webrtc.js:140

onstatusCopy Anchor Link

onstatus: (arg0) => void;

Defined in: lib/webrtc.js:155

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

ondebugCopy Anchor Link

ondebug: (arg0) => void;

Defined in: lib/webrtc.js:158

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

onerrorCopy Anchor Link

onerror: (arg0) => void;

Defined in: lib/webrtc.js:161

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

onconnectionstatechangeCopy Anchor Link

onconnectionstatechange: (arg0) => void;

Defined in: lib/webrtc.js:164

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

ondatachannelopenCopy Anchor Link

ondatachannelopen: () => void;

Defined in: lib/webrtc.js:167

ReturnsCopy Anchor Link

void

ondatachannelcloseCopy Anchor Link

ondatachannelclose: () => void;

Defined in: lib/webrtc.js:170

ReturnsCopy Anchor Link

void

onstreaminfoCopy Anchor Link

onstreaminfo: (arg0) => void;

Defined in: lib/webrtc.js:173

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

onlatencymeasurementCopy Anchor Link

onlatencymeasurement: (arg0) => void;

Defined in: lib/webrtc.js:176

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

onplaystreamrequiredCopy Anchor Link

onplaystreamrequired: () => void;

Defined in: lib/webrtc.js:179

ReturnsCopy Anchor Link

void

onclipboardcontentCopy Anchor Link

onclipboardcontent: (arg0) => void | Promise<void>;

Defined in: lib/webrtc.js:182

May return a promise, which the ordered receive queue awaits.

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void | Promise<void>

onsystemactionCopy Anchor Link

onsystemaction: (arg0) => void;

Defined in: lib/webrtc.js:185

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

oncursorchangeCopy Anchor Link

oncursorchange: (arg0) => void;

Defined in: lib/webrtc.js:188

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

cursor_cacheCopy Anchor Link

cursor_cache: Map<any, any>;

Defined in: lib/webrtc.js:191

onstreamstatsCopy Anchor Link

onstreamstats: (arg0) => void;

Defined in: lib/webrtc.js:194

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

onconnectionCopy Anchor Link

onconnection: (arg0) => void;

Defined in: lib/webrtc.js:197

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

inputCopy Anchor Link

input: Input;

Defined in: lib/webrtc.js:216

clipboardcontentCopy Anchor Link

clipboardcontent: any[];

Defined in: lib/webrtc.js:219

onserversettingsCopy Anchor Link

onserversettings: (arg0) => void;

Defined in: lib/webrtc.js:222

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

ondisplaysettingsCopy Anchor Link

ondisplaysettings: (arg0) => void;

Defined in: lib/webrtc.js:225

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

onpagevisibilityCopy Anchor Link

onpagevisibility: () => void;

Defined in: lib/webrtc.js:228

ReturnsCopy Anchor Link

void

ondisplayconfigCopy Anchor Link

ondisplayconfig: (arg0) => void;

Defined in: lib/webrtc.js:231

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

onprintdocumentCopy Anchor Link

onprintdocument: (arg0) => void;

Defined in: lib/webrtc.js:234

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

onccrateCopy Anchor Link

onccrate: (arg0) => void;

Defined in: lib/webrtc.js:237

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

streamsCopy Anchor Link

streams: any[];

Defined in: lib/webrtc.js:601

AccessorsCopy Anchor Link

webcamTrackCopy Anchor Link

Get SignatureCopy Anchor Link

get webcamTrack(): MediaStreamTrack;

Defined in: lib/webrtc.js:585

The live camera track sent to the server, or null.

ReturnsCopy Anchor Link

MediaStreamTrack

MethodsCopy Anchor Link

setMicrophone()Copy Anchor Link

setMicrophone(enabled, deviceId?): Promise<boolean>;

Defined in: lib/webrtc.js:426

Enables or disables the microphone: attaches a getUserMedia track to the reserved sendonly transceiver (the browser encodes Opus over RTP), or detaches and stops it.

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

Promise<boolean>

False when getUserMedia is unavailable; null when a disable withdrew the enable while its permission request was pending, the stream it got released at once.

ThrowsCopy Anchor Link

When the server withheld the microphone m-line (the microphone is disabled server-side); raised before prompting for permission so the UI never claims an active mic that streams nothing.

setWebcam()Copy Anchor Link

setWebcam(
   enabled, 
   deviceId?, 
   hints?
): Promise<boolean>;

Defined in: lib/webrtc.js:483

Enables or disables the webcam: attaches a getUserMedia video track to the reserved sendonly transceiver (the browser encodes H.264, VP8, VP9, H.265, or AV1 over RTP and the server's virtual camera decodes it), or detaches and stops it. Disabling also deactivates the sender's encodings, because a null or ended track alone does not silence every engine (Firefox keeps the encoder running on it); enabling re-activates them, all without renegotiation.

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

Promise<boolean>

False when getUserMedia is unavailable; null when a disable withdrew the enable while its permission request was pending, the camera it got released at once.

ThrowsCopy Anchor Link

When the server withheld the webcam m-line (the webcam is locked off), raised before prompting for permission; or when the sender refuses the track (its transceiver stopped), the camera released again.

setWebcamCodec()Copy Anchor Link

setWebcamCodec(codec): Promise<boolean>;

Defined in: lib/webrtc.js:541

Sends the camera as one codec: the webcam_encoder name (h264, h265, vp8, vp9, av1) is set on the sender's encoding from the codecs the answer negotiated, so the browser leaves the negotiated order for it; auto, mjpeg, or a codec the answer lacks returns to that order. The codec of an encoding is what the engine offers for this, and an engine without it keeps the negotiated order, logged once.

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

Promise<boolean>

Whether the named codec is what goes out.

dataChannelBufferedAmount()Copy Anchor Link

dataChannelBufferedAmount(): number;

Defined in: lib/webrtc.js:850

Outbound queue depth of the data channel; bulk senders (clipboard, uploads) throttle on this so they cannot starve input and stats on the same channel.

ReturnsCopy Anchor Link

number

dataChannelOpen()Copy Anchor Link

dataChannelOpen(): boolean;

Defined in: lib/webrtc.js:861

Whether the data channel can carry a send right now. Bulk senders check this to report "not connected" instead of dropping into sendDataChannelMessage's quiet no-op.

ReturnsCopy Anchor Link

boolean

waitForDataChannelDrain()Copy Anchor Link

waitForDataChannelDrain(threshold?): Promise<void>;

Defined in: lib/webrtc.js:876

Resolves once queued sends (including the asynchronous gzip queue) have reached the channel and its buffered amount is below threshold. Bulk senders call this between chunks; without it a burst overflows the SCTP send buffer and Chromium closes the channel with OperationError, killing the session. It resumes on the bufferedamountlow event rather than a poll: polling lets the buffer drain to empty between chunks, which collapses throughput, while keeping about threshold bytes queued keeps the pipe full and still yields the channel to input and stats.

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

Promise<void>

motionBufferedAmount()Copy Anchor Link

motionBufferedAmount(): number;

Defined in: lib/webrtc.js:895

Outbound queue depth of the channel pointer motion goes out on (sendMotionMessage), which the motion sender slows down on.

ReturnsCopy Anchor Link

number

sendMotionMessage()Copy Anchor Link

sendMotionMessage(message): void;

Defined in: lib/webrtc.js:906

Sends coalesced pointer motion on the unordered pointer channel while it is open, so a lost sample holds nothing behind it, else on the input channel.

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

sendDataChannelMessage()Copy Anchor Link

sendDataChannelMessage(message): void;

Defined in: lib/webrtc.js:925

Sends a message on the data channel. Nothing is sent while the channel is not open: periodic senders fire before it opens while connecting, and reporting that would mask real failures. Without negotiated gzip the send is synchronous, so the input hot path pays no latency; with it, strings of 512 bytes or more gzip asynchronously and every send passes through an order-preserving queue so a later small message cannot overtake a large one still compressing.

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

onGamepadDisconnect()Copy Anchor Link

onGamepadDisconnect(gp_num): void;

Defined in: lib/webrtc.js:955

Reports a gamepad disconnect as a status message.

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

void

getConnectionStats()Copy Anchor Link

getConnectionStats(): Promise<any>;

Defined in: lib/webrtc.js:976

Collects connection statistics from getStats().

general comes from the transport and candidate-pair reports (its connectionType through the selected pair's remote candidate), video and audio from the inbound-rtp reports (their codecName through the linked codec report; video.decoder reads unknown outside a capturing media context, the only state that exposes decoderImplementation), and data from the data-channel report; the raw reports are attached as reports (with the local candidates and the outbound-rtp reports of the microphone and webcam, by kind, for lib/stream-stats.js) and allReports. The audio section's NetEQ concealment counters are the RED acceptance metric; Chrome reports opus+red under the codec name opus, so RED presence is confirmed from the SDP or the packet size, never from codecName.

ReturnsCopy Anchor Link

Promise<any>

playStream()Copy Anchor Link

playStream(): void;

Defined in: lib/webrtc.js:1165

Starts playback of the element. Engines refuse autoplay before a user gesture; that refusal is reported through onplaystreamrequired.

ReturnsCopy Anchor Link

void

connect()Copy Anchor Link

connect(): void;

Defined in: lib/webrtc.js:1186

Creates the peer connection (relay-only when forceTurn is set) and connects the signaling client, which starts the offer/answer exchange.

ReturnsCopy Anchor Link

void

reset()Copy Anchor Link

reset(): void;

Defined in: lib/webrtc.js:1213

Resets the connection: forgets the cursor cache, closes the data channel and the peer connection, and reconnects, after a three-second pause when signaling was not stable.

ReturnsCopy Anchor Link

void

FunctionsCopy Anchor Link

takeMultiopus()Copy Anchor Link

function takeMultiopus(offer, answer): string;

Defined in: lib/webrtc.js:68

Answers the server's surround stream with the codec it is offered as.

With more than two audio channels the server offers Chromium's multiopus ahead of stereo RED and Opus. No engine lists it in an answer of its own, so its payload type is put first on each audio line the server sends on, with the offer's rtpmap, fmtp, and rtcp-fb for it. An engine that decodes it lists it on its receivers and gets the surround stream; for one that does not, _onSDP sends the answer the engine created itself, and the server streams that page stereo.

ParametersCopy Anchor Link

ReturnsCopy Anchor Link

string

The answer, taking multiopus wherever the offer carries it.