Merge branch 'master' into feature/error-handle

Author: RogerHardiman

.vscode/launch.json

@@ -29,9 +29,14 @@
         {
             "type": "node",
             "request": "launch",
-            "name": "Wowza Example",
-            "program": "${workspaceFolder}/examples/wowza.js",
+            "name": "Demo Example",
+            "program": "${workspaceFolder}/examples/demo.js",
             "preLaunchTask": "tsc: build - tsconfig.json",
+            "args": [
+                "-u","myusername",
+                "-p","mypassword",
+                "rtsp://8.8.8.8/stream1"
+            ],
             "outFiles": [
                 "${workspaceFolder}/dist/**/*.js"
             ]

CHANGELOG.md

@@ -1,3 +1,16 @@
+# Next Release
+
+**New features**:
+Output AV1 streams to a .obu file that 'ffplay' can replay. Tested with MediaMTX with ffmpeg sending AV1 via RTSP to MediaMTX
+Remove wowza demo. They no longer host an online RTSP server we can use
+Add new examples/demo where the URL, username and password can be passed on the Command Line
+Show Wall Clock Time for RTP packets after the RTCP Sender Report is received
+Initial RTSPS support, tested with Bosch CCTV Cameras. (did not work with Axis CCTV Cameras that want to use SRTP)
+Add support for SHA256 hash (in addition to existing MD5)
+
+**Bug fixes**:
+Fix parsing of H265 with no 'fmtp' in the SDP
+
 # v3.0.4
 
 **New features**:

README.md

@@ -30,18 +30,20 @@ Yellowstone was co-developed by Michael Bullington and Roger Hardiman.
 
 ## Examples
 
-An example of most API features can be found at [examples/wowza.js](examples/wowza.js), which will create
-a file named bigbuckbunny.264 in the project's root directory. To test this file with a
-video player, you either need a video player that supports raw H264 frames, or wrap it
-in a container like MP4.
-You can play the video with
-```ffplay bigbuckbunny.264```
+An example of most API features can be found at [examples/demo.js](examples/demo.js), which will
+connect to a RTSP Stream and dump H264, H265 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```
+
+To testthe output file with a video player you can use FFMPEG's ffplay command
+```ffplay outfile.264```
 
 While yellowstone is /**not**/ dependent on ffmpeg, converting the file to an .mp4 can be easily
 accomplished with the following command.
 
 ```sh
-ffmpeg -f h264 -i bigbuckbunny.264 -vcodec copy bigbuckbunny.mp4
+ffmpeg -f h264 -i outfile.264 -vcodec copy outfile.mp4
 ```
 
 

dist/RTSPClient.d.ts

@@ -1,8 +1,12 @@
 /// <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 {
@@ -34,6 +38,9 @@ type Detail = {
     isH264: boolean;
     rtpChannel: number;
     rtcpChannel: number;
+    sr_ntpMSW?: number;
+    sr_ntpLSW?: number;
+    sr_rtptimestamp?: number;
 };
 export default class RTSPClient extends EventEmitter {
     username: string;
@@ -44,7 +51,7 @@ export default class RTSPClient extends EventEmitter {
     isConnected: boolean;
     closed: boolean;
     _url?: string;
-    _client?: net.Socket;
+    _client?: SocketUnion;
     _cSeq: number;
     _unsupportedExtensions?: string[];
     _session?: string;
@@ -60,15 +67,16 @@ export default class RTSPClient extends EventEmitter {
     rtspPacket: Buffer;
     rtspPacketPointer: number;
     clientSSRC: number;
-    tcpSocket: net.Socket;
+    tcpSocket: SocketUnion;
     setupResult: Array<Detail>;
     constructor(username: string, password: string, headers?: {
         [key: string]: string;
     });
-    _netConnect(hostname: string, port: number): Promise<this>;
-    connect(url: string, { keepAlive, connection, }?: {
+    _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;
@@ -83,6 +91,8 @@ export default class RTSPClient extends EventEmitter {
     _sendInterleavedData(channel: number, buffer: Buffer): void;
     _sendUDPData(host: string, port: number, buffer: Buffer): void;
     _emptyReceiverReport(): Buffer;
-    _socketWrite(socket: net.Socket, data: Buffer): Promise<any>;
+    _socketWrite(socket: SocketUnion, data: Buffer): Promise<any>;
+    ntpBaseDate_ms: number;
+    GetWallClockTime(packet: util.RTPPacket, detail: Detail): Date | undefined;
 }
 export { RTPPacket, RTCPPacket } from "./util";