WebRTC module

Classes

Name	Description
`SignallingInterface`	Transport-agnostic signalling interface for WebRTC session setup.
`MediaBridge`	Convenience wrapper that creates WebRTC tracks on a PeerConnection and exposes per-track sender/receiver adapters for PacketStream integration.
`PeerSession`	Manages a WebRTC peer connection lifecycle over any signalling transport that implements SignallingInterface.
`WebRtcTrackSender`	PacketProcessor that sends encoded media to a single libdatachannel Track via sendFrame().
`WebRtcTrackReceiver`	PacketStreamAdapter that receives depacketized frames from a single remote libdatachannel Track and emits them as VideoPacket or AudioPacket into a PacketStream.
`CodecNegotiator`	Maps RTP codec names to FFmpeg encoders and queries FFmpeg at runtime to determine what codecs are available.
`TrackHandle`	Result of creating a track: the track itself plus its RTP config. Keep the config around - you need it for WebRtcTrackSender.
`JitterBufferConfig`	Receive-side jitter buffer behaviour for depacketized WebRTC media frames.
`CodecSpec`	Canonical description of a codec supported by icey's WebRTC helpers.
`NegotiatedCodec`	Result of codec negotiation between a remote SDP offer and the local FFmpeg codec inventory.

Macros

Name	Description
`WEBRTC_API`

WEBRTC_API

#define WEBRTC_API ICY_IMPORT

Defined in src/webrtc/include/icy/webrtc/webrtc.h:24

Enumerations

Name	Description
`CodecMediaType`	Whether a codec carries audio or video media.
`CodecId`	Stable codec identifiers used across negotiation and track setup.

CodecMediaType

enum CodecMediaType

Defined in src/webrtc/include/icy/webrtc/codecnegotiator.h:40

Whether a codec carries audio or video media.

Value	Description
`Video`	Video RTP payload.
`Audio`	Audio RTP payload.

CodecId

enum CodecId

Defined in src/webrtc/include/icy/webrtc/codecnegotiator.h:48

Stable codec identifiers used across negotiation and track setup.

Value	Description
`Unknown`	Unrecognized or unsupported codec.
`H264`	H.264 / AVC.
`H265`	H.265 / HEVC.
`VP8`	Google VP8.
`VP9`	Google VP9.
`AV1`	AOMedia AV1.
`Opus`	Opus audio.
`PCMU`	G.711 mu-law audio.
`PCMA`	G.711 A-law audio.
`G722`	G.722 wideband audio.
`AAC`	AAC audio.

Functions

Return	Name	Description
`TrackHandle`	`createVideoTrack`	Create a video send track on a PeerConnection.
`TrackHandle`	`createAudioTrack`	Create an audio send track on a PeerConnection.
`std::shared_ptr< rtc::Track >`	`createVideoReceiveTrack`	Create a pure receive-side video track on a PeerConnection.
`std::shared_ptr< rtc::Track >`	`createAudioReceiveTrack`	Create a pure receive-side audio track on a PeerConnection.
`bool`	`setupReceiveTrack`	Set up the receive-side media handler chain on a remote track.
`uint32_t`	`generateSsrc`	Generate a random SSRC.
`const char *`	`stateToString`	Convert a PeerSession::State to a lowercase C string for logging.

createVideoTrack

TrackHandle createVideoTrack(std::shared_ptr< rtc::PeerConnection > pc, const av::VideoCodec & codec, uint32_t ssrc = 0, const std::string & cname = {}, const std::string & mid = {}, rtc::Description::Direction direction = rtc::Description::Direction::SendRecv, unsigned nackBuffer = 512, std::function< void()> onPli = nullptr, std::function< void(unsigned int)> onRemb = nullptr, int payloadType = -1)

Defined in src/webrtc/include/icy/webrtc/track.h:63

Create a video send track on a PeerConnection.

Sets up the full outgoing media handler chain: Packetizer → SrReporter → NackResponder → PliHandler → RembHandler

The packetizer is selected based on the codec: H264 → H264RtpPacketizer (Annex-B long start sequence) H265 → H265RtpPacketizer (Annex-B long start sequence) VP8/VP9/other → generic RtpPacketizer

Returns

TrackHandle with the track and its RTP config.

Parameters

Parameter	Type	Description
`pc`	`std::shared_ptr< rtc::PeerConnection >`	PeerConnection to add the track to.
`codec`	const `av::VideoCodec` &	Video codec. Must name a supported RTP codec or FFmpeg encoder (e.g. "H264" or "libx264").
`ssrc`	`uint32_t`	RTP SSRC. 0 = auto-generate.
`cname`	`const std::string &`	RTCP CNAME. Empty = "icey".
`mid`	`const std::string &`	MID to use for the track when answering an existing offer.
`direction`	`rtc::Description::Direction`	Direction to advertise for the negotiated m-line.
`nackBuffer`	`unsigned`	Max packets stored for NACK retransmission.
`onPli`	`std::function< void()>`	Callback when remote peer requests a keyframe. Connect to your encoder to force IDR.
`onRemb`	`std::function< void(unsigned int)>`	Callback when remote peer reports bandwidth estimate. Bitrate in bits/sec.
`payloadType`	`int`	Explicit RTP payload type to reuse when answering an offer. -1 = use the codec's default/preferred type.

createAudioTrack

TrackHandle createAudioTrack(std::shared_ptr< rtc::PeerConnection > pc, const av::AudioCodec & codec, uint32_t ssrc = 0, const std::string & cname = {}, const std::string & mid = {}, rtc::Description::Direction direction = rtc::Description::Direction::SendRecv, int payloadType = -1)

Defined in src/webrtc/include/icy/webrtc/track.h:94

Create an audio send track on a PeerConnection.

Sets up the outgoing media handler chain: AudioRtpPacketizer → SrReporter

The packetizer clock rate is selected based on codec: opus → 48kHz, PCMU/PCMA → 8kHz, etc.

Returns

TrackHandle with the track and its RTP config.

Parameters

Parameter	Type	Description
`pc`	`std::shared_ptr< rtc::PeerConnection >`	PeerConnection to add the track to.
`codec`	const `av::AudioCodec` &	Audio codec. Must name a supported RTP codec or FFmpeg encoder (e.g. "opus" or "libopus").
`ssrc`	`uint32_t`	RTP SSRC. 0 = auto-generate.
`cname`	`const std::string &`	RTCP CNAME. Empty = "icey".
`mid`	`const std::string &`	MID to use for the track when answering an existing offer.
`direction`	`rtc::Description::Direction`	Direction to advertise for the negotiated m-line.
`payloadType`	`int`	Explicit RTP payload type to reuse when answering an offer. -1 = use the codec's default/preferred type.

createVideoReceiveTrack

std::shared_ptr< rtc::Track > createVideoReceiveTrack(std::shared_ptr< rtc::PeerConnection > pc, const av::VideoCodec & codec, const std::string & mid = {}, rtc::Description::Direction direction = rtc::Description::Direction::RecvOnly, int payloadType = -1)

Defined in src/webrtc/include/icy/webrtc/track.h:108

Create a pure receive-side video track on a PeerConnection.

Unlike createVideoTrack(), this does not add a local SSRC or sender packetizer chain. It is for answers that only receive remote media.

createAudioReceiveTrack

std::shared_ptr< rtc::Track > createAudioReceiveTrack(std::shared_ptr< rtc::PeerConnection > pc, const av::AudioCodec & codec, const std::string & mid = {}, rtc::Description::Direction direction = rtc::Description::Direction::RecvOnly, int payloadType = -1)

Defined in src/webrtc/include/icy/webrtc/track.h:120

Create a pure receive-side audio track on a PeerConnection.

Unlike createAudioTrack(), this does not add a local SSRC or sender packetizer chain. It is for answers that only receive remote media.

setupReceiveTrack

bool setupReceiveTrack(std::shared_ptr< rtc::Track > track)

Defined in src/webrtc/include/icy/webrtc/track.h:141

Set up the receive-side media handler chain on a remote track.

Selects the correct depacketizer based on the track's SDP codec: Video: H264RtpDepacketizer, H265RtpDepacketizer, or generic Audio: OpusRtpDepacketizer, PCMURtpDepacketizer, etc.

Also chains an RtcpReceivingSession for receiver reports.

Call this from PeerConnection::onTrack() before binding a WebRtcTrackReceiver.

Returns

True when a supported depacketizer was installed.

Parameters

Parameter	Type	Description
`track`	`std::shared_ptr< rtc::Track >`	The remote track from onTrack callback.

generateSsrc

uint32_t generateSsrc()

Defined in src/webrtc/include/icy/webrtc/track.h:145

Generate a random SSRC.

stateToString

const char * stateToString(PeerSession::State state)

Defined in src/webrtc/include/icy/webrtc/peersession.h:206

Convert a PeerSession::State to a lowercase C string for logging.

Returns

One of: "idle", "outgoing-init", "incoming-init", "negotiating", "active", "ending", "ended".

Parameters

Parameter	Type	Description
`state`	`PeerSession::State`	State value to convert.