3.0.8 release

CHANGELOG.md

@@ -1,5 +1,11 @@
 # Next Release
 
+# 3.0.8 - 15th January 2027
+Add caching Digest Authentication code from Leone25 Enrico
+Fix bug causing an error with Panasonic Cameras returning RTP/AVP/UDP in SETUP
+Add -t to the example demo.js to pick the RTP transport (-t udp or -t tcp)
+Handle multiple WWW-Authenticate responses (which is valid) and pick Digest over Basic
+
 # 3.0.7 - 13th January 2027
 Add H266 (VVC) support with output to a .266 file.
 Tested with https://github.com/jimm98y/SharpRealTimeStreaming which uses the SharpRTSP Library

README.md

@@ -36,7 +36,7 @@ An example of most API features can be found at [examples/demo.js](examples/demo
 connect to a RTSP Stream and dump H264, H265, AV1 and AAC contents to a file.
 For example
 ```node examples\demo.js rtsp://myhostname/stream1```
-```node examples\demo.js -u username -p password rtsp://myhostname/stream1```
+```node examples\demo.js -u username -p password -t tcp rtsp://myhostname/stream1```
 
 To testthe output file with a video player you can use FFMPEG's ffplay command
 ```ffplay outfile.264```

dist/RTSPClient copy.d.ts

@@ -0,0 +1,107 @@
+/// <reference types="node" />
+/// <reference types="node" />
+/// <reference types="node" />
+/// <reference types="node" />
+import * as net from "net";
+import * as tls from "tls";
+type SocketUnion = net.Socket | tls.TLSSocket;
+import { EventEmitter } from "events";
+import * as util from "./util";
+import { Transport } from "./util";
+import * as transform from "sdp-transform";
+declare enum ReadStates {
+    SEARCHING = 0,
+    READING_RTSP_HEADER = 1,
+    READING_RTSP_PAYLOAD = 2,
+    READING_RAW_PACKET_SIZE = 3,
+    READING_RAW_PACKET = 4
+}
+type Connection = "udp" | "tcp";
+type AuthOptions = {
+    type: "Digest" | "Basic";
+    realm?: string;
+    nonce?: string;
+    qop?: string;
+    algorithm?: "MD5" | "SHA-256";
+};
+type Headers = {
+    [key: string]: string | number | undefined;
+    Session?: string;
+    Location?: string;
+    CSeq?: number;
+    "WWW-Authenticate"?: string;
+    Transport?: string;
+    Unsupported?: string;
+};
+type Detail = {
+    codec: string;
+    mediaSource: ({
+        type: string;
+        port: number;
+        protocol: string;
+        payloads?: string | undefined;
+    } & transform.MediaDescription);
+    transport: Transport['parameters'];
+    isH264: boolean;
+    rtpChannel: number;
+    rtcpChannel: number;
+    sr_ntpMSW?: number;
+    sr_ntpLSW?: number;
+    sr_rtptimestamp?: number;
+};
+export default class RTSPClient extends EventEmitter {
+    username: string;
+    password: string;
+    headers: {
+        [key: string]: string;
+    };
+    isConnected: boolean;
+    closed: boolean;
+    _url?: string;
+    _client?: SocketUnion;
+    _cSeq: number;
+    _unsupportedExtensions?: string[];
+    _authOpions?: AuthOptions;
+    _session?: string;
+    _keepAliveID?: NodeJS.Timeout;
+    _nextFreeInterleavedChannel: number;
+    _nextFreeUDPPort: number;
+    readState: ReadStates;
+    messageBytes: number[];
+    rtspContentLength: number;
+    rtspStatusLine: string;
+    rtspHeaders: Headers;
+    rtspPacketLength: number;
+    rtspPacket: Buffer;
+    rtspPacketPointer: number;
+    clientSSRC: number;
+    tcpSocket: SocketUnion;
+    setupResult: Array<Detail>;
+    constructor(username: string, password: string, headers?: {
+        [key: string]: string;
+    });
+    _netConnect(hostname: string, port: number, secure?: boolean): Promise<this>;
+    connect(url: string, { keepAlive, connection, secure, }?: {
+        keepAlive: boolean;
+        connection?: Connection;
+        secure: boolean;
+    }): Promise<Detail[]>;
+    request(requestName: string, headersParam?: Headers, url?: string): Promise<{
+        headers: Headers;
+        mediaHeaders?: string[];
+    } | void>;
+    respond(status: string, headersParam?: Headers): void;
+    play(): Promise<void>;
+    pause(): Promise<void>;
+    sendAudioBackChannel(audioChunk: Buffer): Promise<void>;
+    close(isImmediate?: boolean): Promise<void>;
+    _onData(data: Buffer): void;
+    _sendInterleavedData(channel: number, buffer: Buffer): void;
+    _sendUDPData(host: string, port: number, buffer: Buffer): void;
+    _emptyReceiverReport(): Buffer;
+    _socketWrite(socket: SocketUnion, data: Buffer): Promise<any>;
+    private _generateAuthString;
+    ntpBaseDate_ms: number;
+    GetWallClockTime(packet: util.RTPPacket, detail: Detail): Date | undefined;
+}
+export { RTPPacket, RTCPPacket } from "./util";