RTP

Node.js

An output type that allows streaming video and audio from Smelter over RTP.

Usage

import Smelter from "@swmansion/smelter-node";
import { View } from "@swmansion/smelter";

async function run() {
  const smelter = new Smelter();
  await smelter.init();
  await smelter.registerOutput("example", <View />, {
    type: "rtp_stream",
    port: 8001,
    transportProtocol: "tcp_server",
    video: {
      encoder: { type: "ffmpeg_h264" },
      resolution: { width: 1920, height: 1080 },
    }
  });
  // At this point you can connect to 8001 TCP port
  // and start receiving RTP traffic.
}
void run();

Reference

Type definitions

type RegisterRtpOutput = {
  type: "rtp_stream";
  port: string | number;
  ip?: string;
  transportProtocol?: "udp" | "tcp_server";
  video?: RtpVideoOptions;
  audio?: RtpAudioOptions;
}

Parameters for registering an output that sends composed video/audio as an RTP stream.

Properties

port

Depends on the value of the transportProtocol field:

udp - Specifies a UDP port number to which RTP packets will be sent.
tcp_server - Specifies a local TCP port number or a range of ports that Smelter will listen to for incoming connections.

Type: string | number

transportProtocol

Transport layer protocol that will be used to send RTP packets.

Type: "udp" | "tcp_server"
Default value: udp
Supported values:
- udp - UDP protocol.
- tcp_server - TCP protocol where Smelter is the server side of the connection.

ip

IP address to which RTP packets should be sent. This field is only valid if transportProtocol field is set to udp.

Type: string | number

video

Parameters of a video included in the RTP stream.

Type: RtpVideoOptions

audio

Parameters of an audio included in the RTP stream.

Type: RtpAudioOptions

RtpVideoOptions

Type definitions

type RtpVideoOptions = {
  resolution: {
    width: number;
    height: number;
  };
  sendEosWhen?: OutputEndCondition;
  encoder: RtpVideoEncoderOptions;
}

Parameters of a video source included in the RTP stream.

Properties

resolution

Output resolution in pixels.

Type: { width: number; height: number;}

sendEosWhen

Condition for termination of the output stream based on the input streams states. If output includes both audio and video streams, then EOS needs to be sent for every type.

Type: OutputEndCondition

encoder

Video encoder options.

Type: RtpVideoEncoderOptions

RtpVideoEncoderOptions

Type definitions

type RtpVideoEncoderOptions = {
    type: "ffmpeg_h264";
    preset:
      | "ultrafast"
      | "superfast"
      | "veryfast"
      | "faster"
      | "fast"
      | "medium"
      | "slow"
      | "slower"
      | "veryslow"
      | "placebo";
    ffmpegOptions?: Map<string, string>;
  }

Configuration for the video encoder, based on the selected codec.

Properties (type: “ffmpeg_h264”)

preset

Video output encoder preset. Visit FFmpeg docs to learn more.

Type: "ultrafast" | "superfast" | "veryfast" | "faster" | "fast" | "medium" | "slow" | "slower" | "veryslow" | "placebo"
Default value: fast
Supported values:
- ultrafast
- superfast
- veryfast
- faster
- fast
- medium
- slow
- slower
- veryslow
- placebo

ffmpegOptions

Raw FFmpeg encoder options. Visit FFmpeg docs to learn more.

Type: Map<string, string>

RtpAudioOptions

Type definitions

type RtpAudioOptions = {
  mixingStrategy?: "sum_clip" | "sum_scale";
  sendEosWhen?: OutputEndCondition;
  encoder: RtpAudioEncoderOptions;
}

Parameters of an audio source included in the RTP stream.

Properties

mixingStrategy

Specifies how audio should be mixed.

Type: "sum_clip" | "sum_scale"
Default value: "sum_clip"
Supported values:
- sum_clip - First, the input samples are summed. If the result exceeds the i16 PCM range, it is clipped.
- sum_scale - First, the input samples are summed. If the result exceeds the i16 PCM range, the summed samples are scaled down by a factor to fit within the range.

sendEosWhen

Condition for termination of the output stream based on the input streams states. If output includes both audio and video streams, then EOS needs to be sent for every type.

Type: OutputEndCondition

encoder

Audio encoder options.

Type: RtpAudioEncoderOptions

RtpAudioEncoderOptions

Type definitions

type RtpAudioEncoderOptions = {
    type: "opus";
    channels: "mono" | "stereo";
    preset?: "quality" | "voip" | "lowest_latency";
  }

Configuration for the audio encoder, based on the selected codec.

Properties(type: “opus”)

channels

Channels configuration

Type: "mono" | "stereo"
Supported values:
- mono - Mono audio (single channel).
- stereo - Stereo audio (two channels).

preset

Audio output encoder preset.

Type: "quality" | "voip" | "lowest_latency"
Default value: voip
Supported values:
- quality - Recommended for broadcast and high-fidelity applications requiring decoded audio to maintain maximum fidelity to the input signal.
- voip - Recommended for VoIP and videoconferencing applications, prioritizing listening quality and speech intelligibility.
- lowest_latency - Recommended only when achieving the lowest possible latency is the highest priority.

OutputEndCondition

Type definitions

type OutputEndCondition =
  | { anyOf: string[]; }
  | { allOf: string[]; }
  | { anyInput: boolean; }
  | { allInputs: boolean; };

Defines when the output stream should end based on the state of the input streams. Only one of the nested fields can be set at a time.

By default, the input stream is considered finished/ended when:

TCP connection was dropped/closed.
RTCP Goodbye packet (BYE) was received.
MP4 track has ended.
Input was unregistered already (or never registered).

Properties

anyOf

List of the input streams. The output stream will terminate if any stream in the list finishes.

Type: string[]

allOf

List of the input streams. The output stream will terminate when all streams in the list finish.

Type: string[]

anyInput

Terminate the output stream if any of the input streams end, including streams added after the output was registered. Notably, the output stream will not terminate if no inputs were ever connected.

Type: boolean

allInputs

Terminate the output stream only when all input streams have finished. Notably, the output stream will terminate if no inputs were ever connected.

Type: boolean