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ring: stream import options, direct sdp usage

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This commit is contained in:
Koushik Dutta
2022-03-10 09:47:47 -08:00
parent 0da966afd3
commit eae580708b
5 changed files with 106 additions and 28 deletions
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12
common/src/listen-cluster.ts
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@@ -1,4 +1,4 @@
import net, { AddressInfo } from 'net';
import net from 'net';
import { once } from 'events';
import dgram from 'dgram';
@@ -8,6 +8,16 @@ export async function listenZero(server: net.Server) {
return (server.address() as net.AddressInfo).port;
}
export function closeQuiet(socket: dgram.Socket) {
if (!socket)
return;
try {
socket.close()
}
catch (e) {
}
}
export async function bindUdp(server: dgram.Socket, usePort: number) {
server.bind(usePort);
await once(server, 'listening');

8
common/src/sdp-utils.ts
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@@ -1,3 +1,11 @@
export function replacePorts(sdp: string, audioPort: number, videoPort: number) {
let outputSdp = sdp
.replace(/c=IN .*/, `c=IN IP4 127.0.0.1`)
.replace(/m=audio \d+/, `m=audio ${audioPort}`)
.replace(/m=video \d+/, `m=video ${videoPort}`);
return outputSdp;
}
export function addTrackControls(sdp: string) {
let lines = sdp.split('\n').map(line => line.trim());
lines = lines.filter(line => !line.includes('a=control:'));

4
plugins/ring/package-lock.json generated
View File

@@ -1,12 +1,12 @@
{
"name": "@scrypted/ring",
"version": "0.0.60",
"version": "0.0.65",
"lockfileVersion": 2,
"requires": true,
"packages": {
"": {
"name": "@scrypted/ring",
"version": "0.0.60",
"version": "0.0.65",
"dependencies": {
"@koush/ring-client-api": "file:../../external/ring-client-api",
"@types/node": "^16.6.1",

2
plugins/ring/package.json
View File

@@ -37,5 +37,5 @@
"@scrypted/sdk": "file:../../sdk",
"typescript": "^4.6.2"
},
"version": "0.0.60"
"version": "0.0.65"
}

108
plugins/ring/src/main.ts
View File

@@ -3,11 +3,19 @@ import { SipSession, isStunMessage, LiveCallNegotiation, clientApi, generateUuid
import { StorageSettings } from '@scrypted/common/src/settings';
import { startRTCSignalingSession } from '@scrypted/common/src/rtc-signaling';
import { RefreshPromise } from "@scrypted/common/src/promise-utils"
import { ChildProcess } from 'child_process';
import { createBindZero, listenZeroSingleClient } from '@scrypted/common/src/listen-cluster';
import { closeQuiet, createBindZero, listenZeroSingleClient } from '@scrypted/common/src/listen-cluster';
import { replacePorts } from '@scrypted/common/src/sdp-utils';
import { RtspServer } from '@scrypted/common/src/rtsp-server'
import dgram from 'dgram';
import { createCryptoLine, getPayloadType, isRtpMessagePayloadType } from './srtp-utils';
import { createCryptoLine, encodeSrtpOptions, getPayloadType, isRtpMessagePayloadType } from './srtp-utils';
import child_process, { ChildProcess } from 'child_process';
enum CaptureModes {
Default = 'Default',
UDP = 'RTSP+UDP',
TCP = 'RTSP+TCP',
FFmpeg = 'FFmpeg Direct Capture',
}
const STREAM_TIMEOUT = 120000;
@@ -42,10 +50,11 @@ class RingCameraLight extends ScryptedDeviceBase implements OnOff {
class RingCameraDevice extends ScryptedDeviceBase implements Intercom, Settings, DeviceProvider, Camera, MotionSensor, BinarySensor, RTCSignalingChannel, VideoCamera {
storageSettings = new StorageSettings(this, {
ffmpegDirectCapture: {
title: 'SIP FFmpeg Direct Capture',
description: 'Experimental: May be faster. May not work.',
type: 'boolean',
captureMode: {
title: 'SIP Gateway',
description: 'The gateway used to import the stream.',
choices: Object.values(CaptureModes),
value: CaptureModes.Default,
}
});
buttonTimeout: NodeJS.Timeout;
@@ -67,10 +76,56 @@ class RingCameraDevice extends ScryptedDeviceBase implements Intercom, Settings,
}
async startIntercom(media: MediaObject): Promise<void> {
}
async stopIntercom(): Promise<void> {
if (!this.session)
throw new Error("not in call");
this.stopIntercom();
const ffmpegInput: FFMpegInput = JSON.parse((await mediaManager.convertMediaObjectToBuffer(media, ScryptedMimeTypes.FFmpegInput)).toString());
const ringRtpOptions = this.rtpDescription;
let cameraSpeakerActive = false;
const audioOutForwarder = await createBindZero();
this.audioOutForwarder = audioOutForwarder.server;
audioOutForwarder.server.on('message', message => {
if (!cameraSpeakerActive) {
cameraSpeakerActive = true;
this.session.activateCameraSpeaker().catch(e => this.console.error('camera speaker activation error', e))
}
this.session.audioSplitter.send(message, ringRtpOptions.audio.port, ringRtpOptions.address);
return null;
});
const args = ffmpegInput.inputArguments.slice();
args.push(
'-vn', '-dn', '-sn',
'-acodec', 'pcm_mulaw',
'-flags', '+global_header',
'-ac', '1',
'-ar', '8k',
'-f', 'rtp',
'-srtp_out_suite', 'AES_CM_128_HMAC_SHA1_80',
'-srtp_out_params', encodeSrtpOptions(this.session.rtpOptions.audio),
`srtp://127.0.0.1:${audioOutForwarder.port}?pkt_size=188`,
);
const cp = child_process.spawn(await mediaManager.getFFmpegPath(), args);
this.audioOutProcess = cp;
cp.on('exit', () => this.console.log('two way audio ended'));
this.session.onCallEnded.subscribe(() => {
closeQuiet(audioOutForwarder.server);
cp.kill('SIGKILL');
});
}
async stopIntercom(): Promise<void> {
closeQuiet(this.audioOutForwarder);
this.audioOutProcess?.kill('SIGKILL');
this.audioOutProcess = undefined;
this.audioOutForwarder = undefined;
}
resetStreamTimeout() {
this.console.log('starting/refreshing stream');
@@ -103,7 +158,8 @@ class RingCameraDevice extends ScryptedDeviceBase implements Intercom, Settings,
const { clientPromise: playbackPromise, port: playbackPort, url: clientUrl } = await listenZeroSingleClient();
const useRtsp = !this.storageSettings.values.ffmpegDirectCapture;
const useRtsp = this.storageSettings.values.captureMode !== CaptureModes.FFmpeg;
const useRtspTcp = this.storageSettings.values.captureMode === CaptureModes.TCP;
const playbackUrl = useRtsp ? `rtsp://127.0.0.1:${playbackPort}` : clientUrl;
@@ -122,11 +178,7 @@ class RingCameraDevice extends ScryptedDeviceBase implements Intercom, Settings,
}
catch (e) {
}
try {
udp.close();
}
catch (e) {
}
closeQuiet(udp);
}
client.on('close', cleanup);
@@ -135,6 +187,7 @@ class RingCameraDevice extends ScryptedDeviceBase implements Intercom, Settings,
sip = await camera.createSipSession(undefined);
sip.onCallEnded.subscribe(cleanup);
this.rtpDescription = await sip.start();
this.console.log('ring sdp', this.rtpDescription.sdp)
const videoPort = useRtsp ? 0 : sip.videoSplitter.address().port;
const audioPort = useRtsp ? 0 : sip.audioSplitter.address().port;
@@ -159,7 +212,10 @@ class RingCameraDevice extends ScryptedDeviceBase implements Intercom, Settings,
'a=rtcp-mux'
];
const sdp = inputSdpLines.filter((x) => Boolean(x)).join('\n');
const proposedSdp = inputSdpLines.filter((x) => Boolean(x)).join('\n');
const sdp = replacePorts(this.rtpDescription.sdp, audioPort, videoPort);
this.console.log('proposed sdp', sdp);
if (useRtsp) {
const rtsp = new RtspServer(client, sdp, udp);
rtsp.console = this.console;
@@ -170,21 +226,23 @@ class RingCameraDevice extends ScryptedDeviceBase implements Intercom, Settings,
sip.videoSplitter.on('message', message => {
if (!isStunMessage(message)) {
const isRtpMessage = isRtpMessagePayloadType(getPayloadType(message));
if (!isRtpMessage)
this.console.log('rtcp')
if (!isRtpMessage) {
this.console.log('rtcp');
}
rtsp.sendVideo(message, !isRtpMessage);
}
});
sip.audioSplitter.on('message', message => {
if (!isStunMessage(message)) {
const isRtpMessage = isRtpMessagePayloadType(getPayloadType(message));
if (!isRtpMessage)
this.console.log('rtcp')
if (!isRtpMessage) {
this.console.log('rtcp');
}
rtsp.sendAudio(message, !isRtpMessage);
}
});
// sip.requestKeyFrame();
sip.requestKeyFrame();
this.session = sip;
try {
@@ -233,7 +291,7 @@ class RingCameraDevice extends ScryptedDeviceBase implements Intercom, Settings,
}
});
// sip.requestKeyFrame();
sip.requestKeyFrame();
this.session = sip;
await packetWaiter;
@@ -260,7 +318,7 @@ class RingCameraDevice extends ScryptedDeviceBase implements Intercom, Settings,
}),
inputArguments: [
...(useRtsp
? ['-rtsp_transport', 'udp']
? ['-rtsp_transport', useRtspTcp ? 'tcp' : 'udp']
: ['-f', 'sdp']),
'-i', playbackUrl,
],
@@ -272,13 +330,15 @@ class RingCameraDevice extends ScryptedDeviceBase implements Intercom, Settings,
}
getSipMediaStreamOptions(): MediaStreamOptions {
const useRtsp = this.storageSettings.values.captureMode !== CaptureModes.FFmpeg;
return {
id: 'sip',
name: 'SIP',
// note that the rtsp stream comes from scrypted,
// can bypass ffmpeg parsing.
// tool: "scrypted",
container: 'rtsp',
container: useRtsp ? 'rtsp' : 'sdp',
video: {
codec: 'h264',
},