Everything sgcWebRTC adds to TsgcRTCPeerConnection, grouped by capability. Signaling and connectivity come from the base component in sgcWebSockets Enterprise; this pack unlocks the media engine on top of it — audio, video, screen capture, data channels, encryption and bandwidth estimation. Every capability works the same in Delphi and C++ Builder, and every license ships full source code.
Enterprise base & SDP
Opus / G.711
VP8 / H.264 / MJPEG
RFC 8831 / 8832
Capture & render
Encrypted media
Congestion control
The base connectivity is already in sgcWebSockets. sgcWebRTC unlocks the SDP, data channel and media surface on the same component.
| Capability | Ships with | Notes |
|---|---|---|
STUN client (TsgcSTUNClient) | sgcWebSockets Standard | Server-reflexive candidate discovery, available below Enterprise. |
TURN client & server, ICE client (TsgcTURNClient, TsgcICEClient) | sgcWebSockets Enterprise | Relayed candidates and full ICE candidate gathering / nomination. |
Base TsgcRTCPeerConnection, legacy WebSocket-relay signaling | sgcWebSockets Enterprise | GatherCandidates, WriteData and the OnRTCWebSocket… / OnRTCCandidatePairNominated / OnRTCConnect events drive ICE connectivity and a DTLS transport through the built-in RTCPeerConnection protocol server, with no SDP, data channels or media. |
| Manual SDP offer/answer signaling | sgcWebRTC | CreateOffer, CreateAnswer, SetLocalDescription, SetRemoteDescription, AddIceCandidate, RestartIce — the JSEP (RFC 8829) state machine a browser expects. |
| Renegotiation & glare resolution | sgcWebRTC | Re-offers after AddTrack / RemoveTrack, OnNegotiationNeeded, rollback, and the W3C perfect-negotiation rule via Polite. |
| Trickle ICE | sgcWebRTC | TrickleICE and RTCOptions.TrickleICEAuto stream candidates as they are found instead of waiting for the full gathering timeout. |
| Connection & signaling state | sgcWebRTC | ConnectionState (W3C RTCPeerConnectionState subset) and SignalingState (RTCSignalingState subset), plus OnConnectionStateChange and OnError. |
| SCTP data channels, RTP audio/video, SRTP encryption | sgcWebRTC | See the sections below — all three require this pack on top of the Enterprise base. |
One AddTrack call attaches an audio track to the peer connection; PCM goes in, RTP goes out, and the same path runs in reverse for the remote track.
| Capability | Member | Notes |
|---|---|---|
| Attach an audio track | AddTrack(rtctkAudio, aCodec) | Returns a TsgcRTCTrack; RemoveTrack stops sending it. |
| Opus codec | cctAudioOpus | 48 kHz / stereo via a dynamically loaded libopus binding. |
| G.711 codec | cctAudioPCMU / cctAudioPCMA | 8 kHz / mono μ-law and A-law, always available, no external library. |
| Send captured audio | TsgcRTCTrack.SendPCM(aPCM, aSamplesPerChannel) | 16-bit signed interleaved PCM at the encoder's sample rate and channel count. |
| Receive decoded audio | OnAudio(aPCM, aSampleRate, aChannels, aSamplesPerChannel) | Fires on the remote track as RTP arrives and is decoded. |
| Encoder / decoder access | AudioEncoder / AudioDecoder | The underlying TsgcAudioEncoderBase / TsgcAudioDecoderBase object, replaceable per track. |
| Track lifecycle | Enabled, Ended, OnEnded | Enabled := False mutes without renegotiating; Ended reflects a track dropped by the remote side or a re-offer. |
| Microphone capture | TsgcMediaCaptureSource descendants | Native capture per platform (waveIn on Windows, plus Linux, Apple and Android backends) feeds SendPCM directly. |
Video works the same shape as audio — frames in, RTP out — with the codec chosen per track and, for H.264, the platform's own hardware encoder underneath.
| Capability | Member | Notes |
|---|---|---|
| Attach a video track | AddTrack(rtctkVideo, aCodec) | Same call as audio, with a video TsgcCodecType. |
| VP8 codec | cctVideoVP8 | libvpx binding, dynamically loaded, available on Windows and Linux. |
| H.264 codec | cctVideoH264 | Native hardware encoder per platform: Media Foundation (Windows), VideoToolbox (iOS/macOS), MediaCodec (Android). Annex-B stream, RFC 6184 payloading, SPS/PPS carried in-band on every keyframe. |
| Motion JPEG codec | cctVideoJPEG | GDI+ based, Windows and Delphi XE or later. |
| Send a raw frame | TsgcRTCTrack.SendVideoFrame(aFrame, aForceKeyFrame) | Encodes and sends a TsgcVideoFrame (I420/NV12/RGB/BGR family). |
| Send a pre-encoded frame | SendEncodedFrame(aData, aIsKeyFrame) | Bypasses the track's own encoder for an externally encoded frame. |
| Receive decoded / raw frames | OnVideoFrame / OnEncodedFrame | Decoded TsgcVideoFrame, or the raw encoded bitstream with its keyframe flag. |
| Request a keyframe | RequestKeyFrame | Sends an RTCP PLI to the remote sender. |
| Encoder / decoder access | VideoEncoder / VideoDecoder | The underlying TsgcVideoEncoderBase / TsgcVideoDecoderBase object. |
| Camera capture | TsgcMediaCaptureSource descendants | Per-platform capture backends (Windows, Linux, Apple, Android) feed SendVideoFrame directly. |
| Rendering | TsgcMediaRenderer descendants | Per-platform renderer backends (Windows, Linux, Apple, Android) for the decoded remote video. |
Ordered or unordered, fully reliable or bounded by a retransmit count or a lifetime, over SCTP-over-DTLS.
| Capability | Member | Notes |
|---|---|---|
| Open a channel | CreateDataChannel(aLabel, aOrdered, aMaxRetransmits, aMaxPacketLifeTime, aProtocol) | Returns a TsgcRTCDataChannel; forces RTCOptions.DTLS on. |
| Remote-opened channel | OnDataChannel(aChannel) | Fires when the peer opens a channel on this connection. |
| Send text / binary | Send(aText) / SendBytes(aBytes) | Returns False if the channel is not open yet. |
| Receive | OnMessage(aText) / OnMessageBinary(aBytes) | Fires on the SCTP-driving thread. |
| Lifecycle events | OnOpen, OnClose, OnError(aError) | Matches the W3C RTCDataChannel lifecycle. |
| Channel state | State, HasId, Id | The SCTP stream id is only final once the DTLS role resolves (RFC 8832 parity). |
| Flow control | BufferedAmount, MaxMessageSize | MaxMessageSize tracks the peer's own a=max-message-size and refuses an over-size send locally. |
| Reliability introspection | Reliability, ReliabilityParam, Ordered, MaxRetransmits, MaxPacketLifeTime | Read back exactly what the channel negotiated. |
| Close | Close | RE-CONFIG based close; State goes to closing, then closed with OnClose. |
| Peer connection level | DataChannelCount, DataChannels[i] | Enumerate every open channel on the connection. |
Every RTP, RTCP and SCTP packet on the connection is encrypted from the DTLS handshake onward — there is no plaintext mode.
| Capability | Detail |
|---|---|
| Key exchange | DTLS handshake over the nominated ICE candidate pair (RFC 5764 DTLS-SRTP), deriving the SRTP keying material. |
| DTLS role | In manual signaling the role comes from a=setup (offerer passive, answerer active); IsDTLSClient reports it once resolved. |
| SRTP protection profiles | SRTP_AES128_CM_SHA1_80, SRTP_AES128_CM_SHA1_32, SRTP_AEAD_AES_128_GCM, SRTP_AEAD_AES_256_GCM, negotiated during the DTLS handshake. |
| Per-SSRC crypto state | One cryptographic context (rollover counter, replay window) per SSRC, as RFC 3711 section 3.2.3 requires, even when BUNDLE puts every m-line on one DTLS-SRTP transport. |
| Scope | Covers RTP/RTCP media and the SCTP association carrying the data channels — the whole connection, not just one m-line. |
| Backend | OpenSSL-backed SRTP implementation. |
A delay-based estimator in the style of Google Congestion Control tracks the link and drives the video encoder's target bitrate.
| Capability | Member | Notes |
|---|---|---|
| Delay-based estimator | TsgcBWE_Estimator | Groups packets in 5 ms send bursts, runs a trendline filter over inter-group delay, and classifies the link as normal, overuse or underuse. |
| Rate control | AIMD controller | Overuse decreases to 85% of the acknowledged rate, underuse holds, normal increases (multiplicative while far from the last known ceiling, additive close to it). |
| Loss ceiling | Loss-rate tracking | Above 10% loss the ceiling drops; under 2% it recovers 8% per second up to MaxBitrate. |
| Transport-wide feedback | TWCC (RTCP) | BuildRTCP_TWCC / ParseRTCP_TWCC transport-wide congestion control feedback, with REMB as a fallback ceiling for peers that only support it. |
| Turn it on | RTCOptions.Media.CongestionControl | Only takes effect when the remote description negotiates the matching header extension / feedback too. |
| Loss recovery | RTCOptions.Media.RTX | RFC 4588 retransmission of the video stream on RTCP NACK. |
| Forward error correction | RTCOptions.Media.FEC | RED / ULPFEC protection on the outgoing video stream. |
| Pacing | RTCOptions.Media.Pacing | Paces outgoing video packets at the estimated bitrate instead of bursting them. |
| Keyframe recovery | RTCP PLI | RequestKeyFrame on the receiving track asks the sender for a fresh keyframe after loss. |