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webrtc: add transport logging, add container hint

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This commit is contained in:
Koushik Dutta
2022-05-28 18:59:35 -07:00
parent 7cf9daabe3
commit 8b2fcd8cc5
5 changed files with 22 additions and 17 deletions
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4
plugins/webrtc/package-lock.json generated
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@@ -1,12 +1,12 @@
{
"name": "@scrypted/webrtc",
"version": "0.0.39",
"version": "0.0.40",
"lockfileVersion": 2,
"requires": true,
"packages": {
"": {
"name": "@scrypted/webrtc",
"version": "0.0.39",
"version": "0.0.40",
"dependencies": {
"@koush/werift": "file:../../external/werift/packages/webrtc",
"@scrypted/common": "file:../../common",

2
plugins/webrtc/package.json
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@@ -1,6 +1,6 @@
{
"name": "@scrypted/webrtc",
"version": "0.0.39",
"version": "0.0.40",
"scripts": {
"prepublishOnly": "NODE_ENV=production scrypted-webpack",
"prescrypted-vscode-launch": "scrypted-webpack",

13
plugins/webrtc/src/ffmpeg-to-wrtc.ts
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@@ -5,13 +5,12 @@ import { connectRTCSignalingClients } from "@scrypted/common/src/rtc-signaling";
import { RtspServer } from "@scrypted/common/src/rtsp-server";
import { createSdpInput, parseSdp } from "@scrypted/common/src/sdp-utils";
import sdk, { FFmpegInput, Intercom, MediaStreamDestination, MediaStreamTool, RTCAVSignalingSetup, RTCSignalingSession } from "@scrypted/sdk";
import ip from 'ip';
import { WeriftOutputSignalingSession } from "./output-signaling-session";
import { waitConnected } from "./peerconnection-util";
import { getFFmpegRtpAudioOutputArguments, RtpTrack, RtpTracks, startRtpForwarderProcess } from "./rtp-forwarders";
import { ScryptedSessionControl } from "./session-control";
import { requiredAudioCodecs, requiredVideoCodec } from "./webrtc-required-codecs";
import { isPeerConnectionAlive } from "./werift-util";
import { isPeerConnectionAlive, logIsPrivateIceTransport } from "./werift-util";
const { mediaManager } = sdk;
@@ -169,15 +168,7 @@ export async function createRTCPeerConnectionSink(
await waitConnected(pc);
console.log('connected', Date.now() - timeStart);
let isPrivate = true;
for (const ice of pc.iceTransports) {
const [address, port] = ice.connection.remoteAddr;
isPrivate = isPrivate && ip.isPrivate(address);
console.log('ice transport ip', address);
}
console.log('Connection is local network:', isPrivate);
const isPrivate = logIsPrivateIceTransport(console, pc);
// should really inspect the session description here.

12
plugins/webrtc/src/werift-util.ts
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@@ -1,4 +1,5 @@
import { RTCIceServer, RTCPeerConnection, RTCSessionDescription } from "@koush/werift";
import ip from 'ip';
export function createRawResponse(response: RTCSessionDescription): RTCSessionDescriptionInit {
return {
@@ -40,3 +41,14 @@ export function getWeriftIceServers(configuration: RTCConfiguration): RTCIceServ
return ret;
}
export function logIsPrivateIceTransport(console: Console, pc: RTCPeerConnection) {
let isPrivate = true;
for (const ice of pc.iceTransports) {
const [address, port] = ice.connection.remoteAddr;
isPrivate = isPrivate && ip.isPrivate(address);
console.log('ice transport ip', address);
}
console.log('Connection is local network:', isPrivate);
return isPrivate;
}

8
plugins/webrtc/src/wrtc-to-rtsp.ts
View File

@@ -1,7 +1,7 @@
import { BundlePolicy, Output, Pipeline, RTCPeerConnection, RtcpPacket, RtcpPayloadSpecificFeedback, RTCRtpTransceiver, RTCSessionDescription, RtpPacket, uint16Add } from "@koush/werift";
import { BundlePolicy, Pipeline, RTCPeerConnection, RtcpPacket, RtcpPayloadSpecificFeedback, RTCRtpTransceiver, RtpPacket, uint16Add } from "@koush/werift";
import { FullIntraRequest } from "@koush/werift/lib/rtp/src/rtcp/psfb/fullIntraRequest";
import { listenZeroSingleClient } from "@scrypted/common/src/listen-cluster";
import { findH264NaluType, findH264NaluTypeInNalu, getNaluTypesInNalu, H264_NAL_TYPE_IDR, RtspServer } from "@scrypted/common/src/rtsp-server";
import { getNaluTypesInNalu, RtspServer } from "@scrypted/common/src/rtsp-server";
import { createSdpInput, parseSdp } from '@scrypted/common/src/sdp-utils';
import sdk, { FFmpegInput, Intercom, MediaObject, MediaStreamUrl, ResponseMediaStreamOptions, RTCAVSignalingSetup, RTCSessionControl, RTCSignalingChannel, RTCSignalingOptions, RTCSignalingSendIceCandidate, RTCSignalingSession, ScryptedMimeTypes } from "@scrypted/sdk";
import dgram from 'dgram';
@@ -9,7 +9,7 @@ import { Socket } from "net";
import { waitConnected } from "./peerconnection-util";
import { getFFmpegRtpAudioOutputArguments, startRtpForwarderProcess } from "./rtp-forwarders";
import { requiredAudioCodecs, requiredVideoCodec } from "./webrtc-required-codecs";
import { createRawResponse, getWeriftIceServers, isPeerConnectionAlive } from "./werift-util";
import { createRawResponse, getWeriftIceServers, isPeerConnectionAlive, logIsPrivateIceTransport } from "./werift-util";
const { mediaManager } = sdk;
@@ -98,6 +98,7 @@ export async function createRTCPeerConnectionSource(options: {
waitConnected(pc).then(() => {
console.log('connected', Date.now() - timeStart);
logIsPrivateIceTransport(console, pc);
sessionControl.startSession().catch(() => { });
});
@@ -297,6 +298,7 @@ export async function createRTCPeerConnectionSource(options: {
const url = `rtsp://127.0.0.1:${port}`;
const mediaStreamUrl: MediaStreamUrl = {
url,
container: 'rtsp',
mediaStreamOptions,
};