node-av 3.1.2 → 4.0.0

package/dist/api/utilities/streaming.d.ts

@@ -1,5 +1,6 @@
-import type { FormatContext } from '../../lib/format-context.js';
-import type { MediaOutput } from '../media-output.js';
+import { FormatContext } from '../../lib/format-context.js';
+import type { AVCodecID } from '../../constants/index.js';
+import type { Muxer } from '../muxer.js';
 /**
  * Streaming protocol utilities.
  *
@@ -8,11 +9,11 @@ import type { MediaOutput } from '../media-output.js';
  *
  * @example
  * ```typescript
- * import { StreamingUtils, MediaOutput } from 'node-av/api';
+ * import { StreamingUtils, Muxer } from 'node-av/api';
  *
  * // Create RTP outputs
- * const videoOutput = await MediaOutput.create('rtp://127.0.0.1:5004');
- * const audioOutput = await MediaOutput.create('rtp://127.0.0.1:5006');
+ * const videoOutput = await Muxer.open('rtp://127.0.0.1:5004');
+ * const audioOutput = await Muxer.open('rtp://127.0.0.1:5006');
  *
  * // Generate SDP for streaming
  * const sdp = StreamingUtils.createSdp([videoOutput, audioOutput]);
@@ -24,23 +25,24 @@ import type { MediaOutput } from '../media-output.js';
  */
 export declare class StreamingUtils {
     /**
-     * Create an SDP (Session Description Protocol) string from media inputs/outputs
+     * Create an SDP (Session Description Protocol) string from demuxer/muxer
      *
      * Generates an SDP description for RTP/RTSP streaming from one or more
-     * configured media inputs/outputs. The inputs/outputs should be configured with RTP
+     * configured demuxer/muxer. The inputs/outputs should be configured with RTP
      * format and have their streams set up before calling this method.
      *
-     * @param inouts - Array of MediaInput or MediaOutput objects configured for RTP
+     * @param inouts - Array of Demuxer or Muxer objects configured for RTP
+     *
      * @returns SDP string if successful, null if failed
      *
      * @example
      * ```typescript
      * // Set up RTP outputs with streams
-     * const output1 = await MediaOutput.create('rtp://239.0.0.1:5004');
-     * await output1.addVideoStream(encoder1);
+     * const output1 = await Muxer.open('rtp://239.0.0.1:5004');
+     * await output1.addStream(encoder1);
      *
-     * const output2 = await MediaOutput.create('rtp://239.0.0.1:5006');
-     * await output2.addAudioStream(encoder2);
+     * const output2 = await Muxer.open('rtp://239.0.0.1:5006');
+     * await output2.addStream(encoder2);
      *
      * // Generate SDP for multicast streaming
      * const sdp = StreamingUtils.createSdp([output1.getFormatContext(), output2.getFormatContext()]);
@@ -51,21 +53,70 @@ export declare class StreamingUtils {
      * ```
      */
     static createSdp(contexts: FormatContext[]): string | null;
+    /**
+     * Generate SDP for RTP/SRTP input using FFmpeg's native SDP generator
+     *
+     * Creates an RFC-compliant SDP description using FFmpeg's internal logic.
+     * This ensures correct codec names, clock rates, and formatting for all codecs.
+     *
+     * @param configs - Array of stream configurations
+     * @param sessionName - Optional session name for the SDP (default: 'RTP Stream')
+     *
+     * @returns SDP string with proper rtpmap and optional crypto attributes
+     *
+     * @example
+     * ```typescript
+     * import { StreamingUtils } from 'node-av/api';
+     * import { AV_CODEC_ID_OPUS, AV_CODEC_ID_H264 } from 'node-av/constants';
+     *
+     * // Single audio stream with SRTP
+     * const sdp = StreamingUtils.createInputSDP([{
+     *   port: 5004,
+     *   codecId: AV_CODEC_ID_OPUS,
+     *   payloadType: 111,
+     *   clockRate: 48000,
+     *   channels: 2,
+     *   srtp: {
+     *     key: Buffer.alloc(16, 0x12),
+     *     salt: Buffer.alloc(14, 0x34)
+     *   }
+     * }]);
+     *
+     * // Multi-stream (video + audio)
+     * const sdp = StreamingUtils.createInputSDP([
+     *   { port: 5006, codecId: AV_CODEC_ID_H264, payloadType: 96, clockRate: 90000 },
+     *   { port: 5004, codecId: AV_CODEC_ID_OPUS, payloadType: 111, clockRate: 48000, channels: 2 }
+     * ], 'My Stream');
+     * ```
+     */
+    static createInputSDP(configs: {
+        port: number;
+        codecId: AVCodecID;
+        payloadType: number;
+        clockRate: number;
+        channels?: number;
+        fmtp?: string;
+        srtp?: {
+            key: Buffer;
+            salt: Buffer;
+            suite?: 'AES_CM_128_HMAC_SHA1_80' | 'AES_CM_256_HMAC_SHA1_80';
+        };
+    }[], sessionName?: string): string;
     /**
      * Validate if an output is configured for RTP streaming
      *
-     * @param output - MediaOutput to check
+     * @param output - Muxer to check
      * @returns true if configured for RTP
      *
      * @example
      * ```typescript
-     * const output = await MediaOutput.create('rtp://127.0.0.1:5004');
+     * const output = await Muxer.open('rtp://127.0.0.1:5004');
      * if (StreamingUtils.isRtpOutput(output)) {
      *   const sdp = StreamingUtils.createSdpForOutput(output);
      * }
      * ```
      */
-    static isRtpOutput(output: MediaOutput): boolean;
+    static isRtpOutput(output: Muxer): boolean;
     /**
      * Build RTP URL from components
      *
@@ -93,4 +144,12 @@ export declare class StreamingUtils {
         localrtcpport?: number;
         pkt_size?: number;
     }): string;
+    /**
+     * Extract all UDP ports from SDP content
+     *
+     * @param sdp - SDP content string
+     *
+     * @returns Array of port numbers (one per stream)
+     */
+    static extractPortsFromSDP(sdp: string): number[];
 }

package/dist/api/utilities/streaming.js

@@ -1,3 +1,7 @@
+import { AVMEDIA_TYPE_VIDEO } from '../../constants/constants.js';
+import { Codec } from '../../lib/codec.js';
+import { FormatContext } from '../../lib/format-context.js';
+import { Rational } from '../../lib/rational.js';
 import { avSdpCreate } from '../../lib/utilities.js';
 /**
  * Streaming protocol utilities.
@@ -7,11 +11,11 @@ import { avSdpCreate } from '../../lib/utilities.js';
  *
  * @example
  * ```typescript
- * import { StreamingUtils, MediaOutput } from 'node-av/api';
+ * import { StreamingUtils, Muxer } from 'node-av/api';
  *
  * // Create RTP outputs
- * const videoOutput = await MediaOutput.create('rtp://127.0.0.1:5004');
- * const audioOutput = await MediaOutput.create('rtp://127.0.0.1:5006');
+ * const videoOutput = await Muxer.open('rtp://127.0.0.1:5004');
+ * const audioOutput = await Muxer.open('rtp://127.0.0.1:5006');
  *
  * // Generate SDP for streaming
  * const sdp = StreamingUtils.createSdp([videoOutput, audioOutput]);
@@ -23,23 +27,24 @@ import { avSdpCreate } from '../../lib/utilities.js';
  */
 export class StreamingUtils {
     /**
-     * Create an SDP (Session Description Protocol) string from media inputs/outputs
+     * Create an SDP (Session Description Protocol) string from demuxer/muxer
      *
      * Generates an SDP description for RTP/RTSP streaming from one or more
-     * configured media inputs/outputs. The inputs/outputs should be configured with RTP
+     * configured demuxer/muxer. The inputs/outputs should be configured with RTP
      * format and have their streams set up before calling this method.
      *
-     * @param inouts - Array of MediaInput or MediaOutput objects configured for RTP
+     * @param inouts - Array of Demuxer or Muxer objects configured for RTP
+     *
      * @returns SDP string if successful, null if failed
      *
      * @example
      * ```typescript
      * // Set up RTP outputs with streams
-     * const output1 = await MediaOutput.create('rtp://239.0.0.1:5004');
-     * await output1.addVideoStream(encoder1);
+     * const output1 = await Muxer.open('rtp://239.0.0.1:5004');
+     * await output1.addStream(encoder1);
      *
-     * const output2 = await MediaOutput.create('rtp://239.0.0.1:5006');
-     * await output2.addAudioStream(encoder2);
+     * const output2 = await Muxer.open('rtp://239.0.0.1:5006');
+     * await output2.addStream(encoder2);
      *
      * // Generate SDP for multicast streaming
      * const sdp = StreamingUtils.createSdp([output1.getFormatContext(), output2.getFormatContext()]);
@@ -55,15 +60,146 @@ export class StreamingUtils {
         }
         return avSdpCreate(contexts);
     }
+    /**
+     * Generate SDP for RTP/SRTP input using FFmpeg's native SDP generator
+     *
+     * Creates an RFC-compliant SDP description using FFmpeg's internal logic.
+     * This ensures correct codec names, clock rates, and formatting for all codecs.
+     *
+     * @param configs - Array of stream configurations
+     * @param sessionName - Optional session name for the SDP (default: 'RTP Stream')
+     *
+     * @returns SDP string with proper rtpmap and optional crypto attributes
+     *
+     * @example
+     * ```typescript
+     * import { StreamingUtils } from 'node-av/api';
+     * import { AV_CODEC_ID_OPUS, AV_CODEC_ID_H264 } from 'node-av/constants';
+     *
+     * // Single audio stream with SRTP
+     * const sdp = StreamingUtils.createInputSDP([{
+     *   port: 5004,
+     *   codecId: AV_CODEC_ID_OPUS,
+     *   payloadType: 111,
+     *   clockRate: 48000,
+     *   channels: 2,
+     *   srtp: {
+     *     key: Buffer.alloc(16, 0x12),
+     *     salt: Buffer.alloc(14, 0x34)
+     *   }
+     * }]);
+     *
+     * // Multi-stream (video + audio)
+     * const sdp = StreamingUtils.createInputSDP([
+     *   { port: 5006, codecId: AV_CODEC_ID_H264, payloadType: 96, clockRate: 90000 },
+     *   { port: 5004, codecId: AV_CODEC_ID_OPUS, payloadType: 111, clockRate: 48000, channels: 2 }
+     * ], 'My Stream');
+     * ```
+     */
+    static createInputSDP(configs, sessionName = 'RTP Stream') {
+        const contexts = [];
+        try {
+            // Create one FormatContext per stream with individual URL (port)
+            for (const streamConfig of configs) {
+                const codec = Codec.findDecoder(streamConfig.codecId);
+                if (!codec) {
+                    throw new Error(`Codec not found for codec ID: ${streamConfig.codecId}`);
+                }
+                // Create format context with URL containing the port
+                const ctx = new FormatContext();
+                ctx.allocContext();
+                ctx.url = `rtp://127.0.0.1:${streamConfig.port}`;
+                // Create stream with codec parameters
+                const stream = ctx.newStream(null);
+                stream.codecpar.codecId = streamConfig.codecId;
+                stream.codecpar.codecType = codec.type;
+                // Set audio-specific parameters
+                if (codec.type !== AVMEDIA_TYPE_VIDEO) {
+                    if (streamConfig.clockRate) {
+                        stream.codecpar.sampleRate = streamConfig.clockRate;
+                    }
+                    if (streamConfig.channels) {
+                        stream.codecpar.channels = streamConfig.channels;
+                    }
+                }
+                // Set time base (required for SDP generation)
+                stream.timeBase = new Rational(1, streamConfig.clockRate);
+                contexts.push(ctx);
+            }
+            // Generate SDP using FFmpeg's native generator
+            // FFmpeg will automatically use the port from each context's URL and generate RFC-compliant codec names
+            let sdp = avSdpCreate(contexts);
+            if (!sdp) {
+                throw new Error('Failed to generate SDP');
+            }
+            // Post-process: Replace session name if provided
+            if (sessionName !== 'RTP Stream') {
+                sdp = sdp.replace(/^s=.*$/m, `s=${sessionName}`);
+            }
+            // Post-process: Replace payload types, add SRTP crypto lines, and/or custom fmtp
+            // Note: We always need to process because FFmpeg assigns payload types automatically
+            const sdpLines = sdp.split('\n');
+            const newLines = [];
+            let streamIndex = -1;
+            for (let i = 0; i < sdpLines.length; i++) {
+                const line = sdpLines[i];
+                if (line.startsWith('m=')) {
+                    // New media section
+                    streamIndex++;
+                    const streamCfg = configs[streamIndex];
+                    // Replace FFmpeg's auto-assigned payload type with user-specified one
+                    const replaced = line.replace(/RTP\/AVP\s+(\d+)/, `RTP/AVP ${streamCfg.payloadType}`);
+                    newLines.push(replaced);
+                    // Add SRTP crypto line right after m= line if configured
+                    if (streamCfg?.srtp) {
+                        const suite = streamCfg.srtp.suite ?? 'AES_CM_128_HMAC_SHA1_80';
+                        const keyMaterial = Buffer.concat([streamCfg.srtp.key, streamCfg.srtp.salt]).toString('base64');
+                        newLines.push(`a=crypto:1 ${suite} inline:${keyMaterial}`);
+                    }
+                }
+                else if (line.startsWith('a=rtpmap:')) {
+                    // Replace payload type in rtpmap line
+                    const streamCfg = configs[streamIndex];
+                    const replaced = line.replace(/^a=rtpmap:(\d+)/, `a=rtpmap:${streamCfg.payloadType}`);
+                    newLines.push(replaced);
+                    // If custom fmtp is provided but FFmpeg didn't generate one, add it
+                    if (streamCfg?.fmtp && !sdpLines[i + 1]?.startsWith('a=fmtp:')) {
+                        newLines.push(`a=fmtp:${streamCfg.payloadType} ${streamCfg.fmtp}`);
+                    }
+                }
+                else if (line.startsWith('a=fmtp:')) {
+                    // Replace payload type in fmtp line and optionally replace content
+                    const streamCfg = configs[streamIndex];
+                    if (streamCfg?.fmtp) {
+                        newLines.push(`a=fmtp:${streamCfg.payloadType} ${streamCfg.fmtp}`);
+                    }
+                    else {
+                        const replaced = line.replace(/^a=fmtp:(\d+)/, `a=fmtp:${streamCfg.payloadType}`);
+                        newLines.push(replaced);
+                    }
+                }
+                else {
+                    newLines.push(line);
+                }
+            }
+            return newLines.join('\n');
+        }
+        finally {
+            // Cleanup all contexts
+            for (const ctx of contexts) {
+                ctx.freeContext();
+            }
+        }
+    }
     /**
      * Validate if an output is configured for RTP streaming
      *
-     * @param output - MediaOutput to check
+     * @param output - Muxer to check
      * @returns true if configured for RTP
      *
      * @example
      * ```typescript
-     * const output = await MediaOutput.create('rtp://127.0.0.1:5004');
+     * const output = await Muxer.open('rtp://127.0.0.1:5004');
      * if (StreamingUtils.isRtpOutput(output)) {
      *   const sdp = StreamingUtils.createSdpForOutput(output);
      * }
@@ -116,5 +252,27 @@ export class StreamingUtils {
         }
         return url;
     }
+    /**
+     * Extract all UDP ports from SDP content
+     *
+     * @param sdp - SDP content string
+     *
+     * @returns Array of port numbers (one per stream)
+     */
+    static extractPortsFromSDP(sdp) {
+        const ports = [];
+        const lines = sdp.split('\n');
+        for (const line of lines) {
+            if (line.startsWith('m=')) {
+                // m=audio 5004 RTP/AVP 111
+                // m=video 5006 RTP/AVP 96
+                const match = /m=\w+\s+(\d+)/.exec(line);
+                if (match) {
+                    ports.push(parseInt(match[1], 10));
+                }
+            }
+        }
+        return ports;
+    }
 }
 //# sourceMappingURL=streaming.js.map

package/dist/api/utilities/streaming.js.map

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package/dist/api/utilities/timestamp.d.ts

@@ -1,4 +1,4 @@
-import type { AVRounding } from '../../constants/constants.js';
+import type { AVRounding } from '../../constants/index.js';
 import type { IRational } from '../../lib/types.js';
 /**
  * Timestamp and time base utilities.

package/dist/api/webrtc-stream.d.ts

@@ -0,0 +1,288 @@
+import { RTCRtpCodecParameters } from 'werift';
+import type { AVCodecID } from '../constants/index.js';
+import type { RTPStreamOptions } from './rtp-stream.js';
+/**
+ * Codec information for WebRTC streaming.
+ *
+ * Contains RTP codec parameters and FFmpeg codec IDs for video and audio streams.
+ * Used for codec negotiation in WebRTC peer connections.
+ */
+export interface WebRTCCodecInfo {
+    /**
+     * Video codec configuration.
+     * Combines RTP parameters (mimeType, clockRate, etc.) with FFmpeg codec ID.
+     */
+    video?: Partial<RTCRtpCodecParameters> & {
+        codecId: AVCodecID;
+    };
+    /**
+     * Optional audio codec configuration.
+     * Combines RTP parameters (mimeType, clockRate, channels) with FFmpeg codec ID.
+     */
+    audio?: Partial<RTCRtpCodecParameters> & {
+        codecId: AVCodecID;
+    };
+}
+/**
+ * Options for configuring WebRTC session with werift integration.
+ */
+export interface WebRTCStreamOptions extends Omit<RTPStreamOptions, 'onVideoPacket' | 'onAudioPacket' | 'supportedVideoCodecs' | 'supportedAudioCodecs'> {
+    /**
+     * ICE servers for NAT traversal and STUN/TURN configuration.
+     *
+     * @default []
+     */
+    iceServers?: {
+        urls: string;
+    }[];
+    /**
+     * Callback invoked when a new ICE candidate is discovered.
+     * Send this candidate to the remote peer via signaling channel.
+     *
+     * @param candidate - ICE candidate string to send to remote peer
+     */
+    onIceCandidate?: (candidate: string) => void;
+}
+/**
+ * Complete WebRTC session management with werift integration.
+ *
+ * Provides end-to-end WebRTC streaming with automatic SDP negotiation,
+ * ICE candidate handling, and peer connection management.
+ * Built on top of {@link RTPStream} for generic RTP streaming with WebRTC-specific
+ * protocol details handled automatically.
+ * Integrates with werift library for RTCPeerConnection and media track handling.
+ * Ideal for building complete WebRTC streaming applications with minimal code.
+ *
+ * @example
+ * ```typescript
+ * import { WebRTCStream } from 'node-av/api';
+ *
+ * // Create session from media source
+ * const session = await WebRTCStream.create('rtsp://camera.local/stream', {
+ *   mtu: 1200,
+ *   hardware: 'auto',
+ *   iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],
+ *   onIceCandidate: (candidate) => {
+ *     sendToClient({ type: 'candidate', value: candidate });
+ *   }
+ * });
+ *
+ * // Process SDP offer from client
+ * const answer = await session.setOffer(clientOffer);
+ * sendToClient({ type: 'answer', value: answer });
+ *
+ * // Start streaming
+ * await session.start();
+ * ```
+ *
+ * @example
+ * ```typescript
+ * // Complete WebSocket signaling server
+ * import { WebSocket } from 'ws';
+ *
+ * ws.on('message', async (data) => {
+ *   const msg = JSON.parse(data);
+ *
+ *   if (msg.type === 'offer') {
+ *     const session = await WebRTCStream.create(msg.url, {
+ *       hardware: 'auto',
+ *       onIceCandidate: (candidate) => {
+ *         ws.send(JSON.stringify({ type: 'candidate', value: candidate }));
+ *       }
+ *     });
+ *
+ *     const answer = await session.setOffer(msg.value);
+ *     ws.send(JSON.stringify({ type: 'answer', value: answer }));
+ *
+ *     await session.start();
+ *   } else if (msg.type === 'candidate') {
+ *     session.addIceCandidate(msg.value);
+ *   }
+ * });
+ * ```
+ *
+ * @see {@link RTPStream} For library-agnostic RTP streaming
+ * @see {@link Demuxer} For input media handling
+ * @see {@link HardwareContext} For GPU acceleration
+ */
+export declare class WebRTCStream {
+    private stream;
+    private pc;
+    private videoTrack;
+    private audioTrack;
+    private options;
+    private pendingIceCandidates;
+    /**
+     * @param options - Session configuration options
+     *
+     * Use {@link create} factory method
+     *
+     * @internal
+     */
+    private constructor();
+    /**
+     * Create a WebRTC session from a media source.
+     *
+     * Opens the input media, creates internal streaming components, and prepares
+     * for WebRTC peer connection negotiation. Does not start streaming yet.
+     * Call {@link setOffer} to negotiate SDP and {@link start} to begin streaming.
+     *
+     * @param inputUrl - Media source URL (RTSP, file path, HTTP, etc.)
+     *
+     * @param options - Session configuration options
+     *
+     * @returns Configured WebRTC session instance
+     *
+     * @example
+     * ```typescript
+     * const session = await WebRTCStream.create('rtsp://camera.local/stream', {
+     *   mtu: 1200,
+     *   hardware: 'auto',
+     *   iceServers: [{ urls: 'stun:stun.l.google.com:19302' }]
+     * });
+     * ```
+     *
+     * @example
+     * ```typescript
+     * // Session from file with hardware acceleration
+     * const session = await WebRTCStream.create('video.mp4', {
+     *   hardware: {
+     *     deviceType: AV_HWDEVICE_TYPE_CUDA
+     *   }
+     * });
+     * ```
+     */
+    static create(inputUrl: string, options?: WebRTCStreamOptions): WebRTCStream;
+    /**
+     * Start streaming media to WebRTC peer connection.
+     *
+     * Begins the media processing pipeline, reading packets from input,
+     * transcoding if necessary, and sending RTP packets to media tracks.
+     * Note: The stream is automatically started by {@link setOffer}, so calling
+     * this method explicitly is optional. This method is provided for cases where
+     * you need explicit control over when streaming begins.
+     *
+     * @returns Promise that resolves when streaming completes
+     *
+     * @throws {FFmpegError} If transcoding or muxing fails
+     *
+     * @example
+     * ```typescript
+     * const session = await WebRTCStream.create('input.mp4');
+     * session.onIceCandidate = (c) => sendToRemote(c);
+     *
+     * const answer = await session.setOffer(remoteOffer);
+     * sendToRemote(answer);
+     * // Stream is already started by setOffer
+     * ```
+     */
+    start(): Promise<void>;
+    /**
+     * Stop streaming gracefully and clean up all resources.
+     *
+     * Stops the stream, closes peer connection, and releases all resources.
+     * Safe to call multiple times. After stopping, you can call start() again
+     * to restart the session.
+     *
+     * @example
+     * ```typescript
+     * const session = await WebRTCStream.create('input.mp4');
+     * await session.start();
+     *
+     * // Stop after 10 seconds
+     * setTimeout(async () => await session.stop(), 10000);
+     * ```
+     */
+    stop(): Promise<void>;
+    /**
+     * Get detected codec information.
+     *
+     * Returns RTP codec parameters and FFmpeg codec IDs for video and audio.
+     * Useful for inspecting what codecs will be used in the WebRTC session.
+     * The input is automatically opened during {@link create}, so codecs can be
+     * detected immediately after session creation.
+     *
+     * @returns Codec configuration for video and audio streams
+     *
+     * @example
+     * ```typescript
+     * const session = await WebRTCStream.create('input.mp4');
+     * const codecs = session.getCodecs();
+     *
+     * console.log('Video:', codecs.video.mimeType);
+     * console.log('Audio:', codecs.audio?.mimeType);
+     * ```
+     */
+    getCodecs(): WebRTCCodecInfo;
+    /**
+     * Process SDP offer from remote peer and generate SDP answer.
+     *
+     * Creates RTCPeerConnection with detected codecs, sets up media tracks,
+     * processes the remote SDP offer, and generates a local SDP answer.
+     * Also configures ICE candidate handling via {@link onIceCandidate} callback.
+     * Must be called before {@link start}.
+     *
+     * @param offerSdp - SDP offer string from remote WebRTC peer
+     *
+     * @returns SDP answer string to send back to remote peer
+     *
+     * @example
+     * ```typescript
+     * const session = await WebRTCStream.create('input.mp4');
+     *
+     * // Setup ICE candidate handler first
+     * session.onIceCandidate = (candidate) => {
+     *   sendToRemote({ type: 'candidate', value: candidate });
+     * };
+     *
+     * // Process offer and send answer
+     * const answer = await session.setOffer(remoteOffer);
+     * sendToRemote({ type: 'answer', value: answer });
+     *
+     * // Start streaming
+     * await session.start();
+     * ```
+     */
+    setOffer(offerSdp: string): Promise<string>;
+    /**
+     * Add ICE candidate from remote peer.
+     *
+     * Processes ICE candidates received from the remote peer via signaling channel.
+     * Should be called whenever a new candidate message arrives from remote peer.
+     * Can be called multiple times as candidates are discovered.
+     *
+     * Supports ICE trickling: If called before {@link setOffer}, candidates are buffered
+     * and applied automatically once the remote description is set.
+     *
+     * @param candidate - ICE candidate string from remote peer
+     *
+     * @example
+     * ```typescript
+     * // In signaling message handler
+     * if (msg.type === 'candidate') {
+     *   session.addIceCandidate(msg.value);
+     * }
+     * ```
+     */
+    addIceCandidate(candidate: string): void;
+    /**
+     * Get video codec configuration from input stream.
+     *
+     * @param codecId - FFmpeg codec ID
+     *
+     * @returns RTP codec parameters with codec ID or undefined if unsupported
+     *
+     * @internal
+     */
+    private getVideoCodecConfig;
+    /**
+     * Get audio codec configuration from input stream.
+     *
+     * @param codecId - FFmpeg codec ID
+     *
+     * @returns RTP codec parameters with codec ID or undefined if unsupported
+     *
+     * @internal
+     */
+    private getAudioCodecConfig;
+}