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mediasoup - npm Package Compare versions

Comparing version 3.13.3 to 3.13.4

worker/scripts/clang-format.mjs

2

node/lib/ortc.js

@@ -659,3 +659,3 @@ "use strict";

channels: matchedCapCodec.channels,
parameters: codec.parameters,
parameters: codec.parameters, // Keep the Producer codec parameters.
rtcpFeedback: matchedCapCodec.rtcpFeedback

@@ -662,0 +662,0 @@ };

@@ -188,3 +188,3 @@ "use strict";

kind: 'audio',
uri: 'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time',
uri: 'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time', // eslint-disable-line max-len
preferredId: 4,

@@ -195,3 +195,3 @@ preferredEncrypt: false

kind: 'video',
uri: 'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time',
uri: 'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time', // eslint-disable-line max-len
preferredId: 4,

@@ -198,0 +198,0 @@ preferredEncrypt: false

@@ -76,3 +76,3 @@ "use strict";

kind: 'audio',
uri: 'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time',
uri: 'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time', // eslint-disable-line max-len
preferredId: 4,

@@ -79,0 +79,0 @@ preferredEncrypt: false

@@ -22,4 +22,4 @@ "use strict";

transport = await router.createPlainTransport({
listenIp: '127.0.0.1',
comedia: true,
listenIp: '127.0.0.1', // https://github.com/nodejs/node/issues/14900
comedia: true, // So we don't need to call transport.connect().
enableSctp: true,

@@ -44,4 +44,4 @@ numSctpStreams: { OS: 256, MIS: 256 }

sctpSocket = sctp.connect({
localPort: 5000,
port: 5000,
localPort: 5000, // Required for SCTP over UDP in mediasoup.
port: 5000, // Required for SCTP over UDP in mediasoup.
OS: OS,

@@ -58,4 +58,4 @@ MIS: MIS,

sctpSocket = sctp.connect({
localPort: 5000,
port: 5000,
localPort: 5000, // Required for SCTP over UDP in mediasoup.
port: 5000, // Required for SCTP over UDP in mediasoup.
OS: OS,

@@ -62,0 +62,0 @@ MIS: MIS,

@@ -20,3 +20,3 @@ "use strict";

mimeType: 'video/VP8',
preferredPayloadType: 125,
preferredPayloadType: 125, // Let's force it.
clockRate: 90000

@@ -42,3 +42,3 @@ },

mimeType: 'audio/opus',
preferredPayloadType: 100,
preferredPayloadType: 100, // 100 is the first available dynamic PT.
clockRate: 48000,

@@ -74,3 +74,3 @@ channels: 2,

mimeType: 'video/rtx',
preferredPayloadType: 101,
preferredPayloadType: 101, // 101 is the second available dynamic PT.
clockRate: 90000,

@@ -86,3 +86,3 @@ parameters: {

mimeType: 'video/H264',
preferredPayloadType: 102,
preferredPayloadType: 102, // 102 is the third available dynamic PT.
clockRate: 90000,

@@ -89,0 +89,0 @@ parameters: {

{
"name": "mediasoup",
"version": "3.13.3",
"version": "3.13.4",
"description": "Cutting Edge WebRTC Video Conferencing",

@@ -29,3 +29,3 @@ "contributors": [

"worker/include",
"worker/scripts/*.js",
"worker/scripts/*.mjs",
"worker/scripts/*.json",

@@ -109,6 +109,6 @@ "worker/scripts/*.py",

"@types/debug": "^4.1.12",
"@types/jest": "^29.5.8",
"@types/node": "^20.9.2",
"@typescript-eslint/eslint-plugin": "^6.11.0",
"@typescript-eslint/parser": "^6.11.0",
"@types/jest": "^29.5.9",
"@types/node": "^20.9.3",
"@typescript-eslint/eslint-plugin": "^6.12.0",
"@typescript-eslint/parser": "^6.12.0",
"eslint": "^8.54.0",

@@ -122,4 +122,4 @@ "eslint-plugin-jest": "^27.6.0",

"ts-jest": "^29.1.1",
"typescript": "^5.2.2"
"typescript": "^5.3.2"
}
}

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