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themusicmanthemusicman
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adding rtcp processing example
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‎examples/rtcp-processing/.formatter.exs

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# Used by "mix format"
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[
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inputs: ["{mix,.formatter}.exs", "{config,lib,test}/**/*.{ex,exs}"]
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]

‎examples/rtcp-processing/.gitignore

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# The directory Mix will write compiled artifacts to.
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/_build/
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# If you run "mix test --cover", coverage assets end up here.
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/cover/
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# The directory Mix downloads your dependencies sources to.
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/deps/
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# Where third-party dependencies like ExDoc output generated docs.
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/doc/
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# Ignore .fetch files in case you like to edit your project deps locally.
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/.fetch
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# If the VM crashes, it generates a dump, let's ignore it too.
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erl_crash.dump
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# Also ignore archive artifacts (built via "mix archive.build").
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*.ez
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# Ignore package tarball (built via "mix hex.build").
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echo-*.tar
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# Temporary files, for example, from tests.
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/tmp/

‎examples/rtcp-processing/README.md

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# RTCP Processing
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Receive video and audio from a browser and display the RTCP packets in the browser
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rtcp-processing demonstrates the Public API for processing RTCP packets in ExWebRTC.
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RTCP is used for statistics and control information for media in WebRTC. Using these messages you can get information about the quality of the media, round trip time and packet loss. You can also craft messages to influence the media quality.
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While in `examples/rtcp-processing` directory
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1. Run `mix deps.get`
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2. Run `mix run --no-halt`
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3. Visit `http://127.0.0.1:8829/index.html` in your browser and press the `play` button.

‎examples/rtcp-processing/assets/index.html

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<!DOCTYPE html>
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<html lang="en">
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<head>
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<meta charset="UTF-8">
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<meta name="viewport" content="width=device-width, initial-scale=1.0">
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<meta http-equiv="X-UA-Compatible" content="ie=edge">
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<title>Elixir WebRTC RTCP Example</title>
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</head>
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<body>
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<main>
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<h1>Elixir WebRTC RTCP Example</h1>
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</main>
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<video id="videoPlayer" autoplay controls></video>
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<h2>RTCP Packets from the server:</h2>
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<div id="rtcpPackets"></div>
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<script src="script.js"></script>
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</body>
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</html>

‎examples/rtcp-processing/assets/script.js

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const pcConfig = { 'iceServers': [{ 'urls': 'stun:stun.l.google.com:19302' },] };
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const mediaConstraints = { video: true, audio: true }
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const proto = window.location.protocol === "https:" ? "wss:" : "ws:"
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const ws = new WebSocket(`${proto}//${window.location.host}/ws`);
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ws.onopen = _ => start_connection(ws);
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ws.onclose = event => console.log("WebSocket connection was terminated:", event);
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const start_connection = async (ws) => {
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const videoPlayer = document.getElementById("videoPlayer");
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const rtcpPacketsEl = document.getElementById("rtcpPackets");
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videoPlayer.srcObject = new MediaStream();
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const pc = new RTCPeerConnection(pcConfig);
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pc.ontrack = event => videoPlayer.srcObject.addTrack(event.track);
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pc.onicecandidate = event => {
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if (event.candidate === null) return;
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console.log("Sent ICE candidate:", event.candidate);
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ws.send(JSON.stringify({ type: "ice", data: event.candidate }));
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};
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const localStream = await navigator.mediaDevices.getUserMedia(mediaConstraints);
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for (const track of localStream.getTracks()) {
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pc.addTrack(track, localStream);
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}
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pc.addEventListener(
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"connectionstatechange",
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(_event) => {
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if (pc.connectionState == "connected") {
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setInterval(function() {
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ws.send(JSON.stringify({ type: "getRTCP" }))
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}, 1000)
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}
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},
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false,
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);
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ws.onmessage = async event => {
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console.debug('[message] Data received from server: ', event.data);
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try {
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const { type, data } = JSON.parse(event.data);
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switch (type) {
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case "answer":
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console.log("Received SDP answer:", data);
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await pc.setRemoteDescription(data)
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break;
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case "ice":
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console.log("Recieved ICE candidate:", data);
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await pc.addIceCandidate(data);
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case "rtcpPackets":
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console.log("Recieved rtcp packets:", data);
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var p = document.createElement('p');
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p.innerHTML = JSON.stringify(data)
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rtcpPacketsEl.appendChild(p)
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}
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} catch (error) {
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console.error(error);
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}
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};
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const offer = await pc.createOffer();
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await pc.setLocalDescription(offer);
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console.log("Sent SDP offer:", offer)
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ws.send(JSON.stringify({ type: "offer", data: offer }));
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};

‎examples/rtcp-processing/lib/rtcp_processing/application.ex

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defmodule RtcpProcessing.Application do
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use Application
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require Logger
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@ip {127, 0, 0, 1}
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@port 8829
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@impl true
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def start(_type, _args) do
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Logger.configure(level: :info)
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children = [
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{Bandit, plug: RtcpProcessing.Router, ip: @ip, port: @port}
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]
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Supervisor.start_link(children, strategy: :one_for_one)
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end
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end

‎examples/rtcp-processing/lib/rtcp_processing/peer_handler.ex

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defmodule RtcpProcessing.PeerHandler do
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require Logger
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alias ExWebRTC.{
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ICECandidate,
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MediaStreamTrack,
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PeerConnection,
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RTPCodecParameters,
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SessionDescription
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}
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@behaviour WebSock
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@ice_servers [
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%{urls: "stun:stun.l.google.com:19302"}
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]
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@video_codecs [
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%RTPCodecParameters{
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payload_type: 96,
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mime_type: "video/VP8",
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clock_rate: 90_000
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}
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]
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@audio_codecs [
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%RTPCodecParameters{
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payload_type: 111,
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mime_type: "audio/opus",
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clock_rate: 48_000,
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channels: 2
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}
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]
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@impl true
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def init(_) do
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{:ok, pc} =
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PeerConnection.start_link(
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ice_servers: @ice_servers,
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video_codecs: @video_codecs,
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audio_codecs: @audio_codecs
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)
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video_track = MediaStreamTrack.new(:video)
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audio_track = MediaStreamTrack.new(:audio)
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{:ok, _sender} = PeerConnection.add_track(pc, video_track)
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{:ok, _sender} = PeerConnection.add_track(pc, audio_track)
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state = %{
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peer_connection: pc,
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out_video_track_id: video_track.id,
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out_audio_track_id: audio_track.id,
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in_video_track_id: nil,
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in_audio_track_id: nil,
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rtcp_packets: []
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}
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{:ok, state}
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end
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@impl true
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def handle_in({msg, [opcode: :text]}, state) do
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msg
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|> Jason.decode!()
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|> handle_ws_msg(state)
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end
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@impl true
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def handle_info({:ex_webrtc, _from, msg}, state) do
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handle_webrtc_msg(msg, state)
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end
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@impl true
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def terminate(reason, _state) do
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Logger.warning("WebSocket connection was terminated, reason: #{inspect(reason)}")
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end
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defp handle_ws_msg(%{"type" => "offer", "data" => data}, state) do
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Logger.info("Received SDP offer: #{inspect(data)}")
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offer = SessionDescription.from_json(data)
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:ok = PeerConnection.set_remote_description(state.peer_connection, offer)
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{:ok, answer} = PeerConnection.create_answer(state.peer_connection)
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:ok = PeerConnection.set_local_description(state.peer_connection, answer)
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answer_json = SessionDescription.to_json(answer)
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msg =
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%{"type" => "answer", "data" => answer_json}
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|> Jason.encode!()
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Logger.info("Sent SDP answer: #{inspect(answer_json)}")
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{:push, {:text, msg}, state}
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end
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defp handle_ws_msg(%{"type" => "ice", "data" => data}, state) do
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Logger.info("Received ICE candidate: #{inspect(data)}")
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candidate = ICECandidate.from_json(data)
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:ok = PeerConnection.add_ice_candidate(state.peer_connection, candidate)
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{:ok, state}
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end
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defp handle_ws_msg(%{"type" => "getRTCP"}, state) do
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Logger.info("Received request for RTCP packets")
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packets =
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Enum.map(state.rtcp_packets, fn packet ->
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"packet: #{inspect(packet)}"
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end)
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msg = Jason.encode!(%{type: "rtcpPackets", data: packets})
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{:push, {:text, msg}, %{state | rtcp_packets: []}}
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end
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defp handle_webrtc_msg({:ice_candidate, candidate}, state) do
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candidate_json = ICECandidate.to_json(candidate)
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msg =
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%{"type" => "ice", "data" => candidate_json}
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|> Jason.encode!()
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Logger.info("Sent ICE candidate: #{inspect(candidate_json)}")
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{:push, {:text, msg}, state}
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end
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defp handle_webrtc_msg({:track, track}, state) do
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%MediaStreamTrack{kind: kind, id: id} = track
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state =
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case kind do
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:video -> %{state | in_video_track_id: id}
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:audio -> %{state | in_audio_track_id: id}
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end
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{:ok, state}
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end
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defp handle_webrtc_msg({:rtp, id, packet}, %{in_audio_track_id: id} = state) do
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PeerConnection.send_rtp(state.peer_connection, state.out_audio_track_id, packet)
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{:ok, state}
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end
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defp handle_webrtc_msg({:rtp, id, packet}, %{in_video_track_id: id} = state) do
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PeerConnection.send_rtp(state.peer_connection, state.out_video_track_id, packet)
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{:ok, state}
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end
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defp handle_webrtc_msg({:rtcp, packets}, state) do
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{:ok, %{state | rtcp_packets: state.rtcp_packets ++ packets}}
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end
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defp handle_webrtc_msg(_msg, state), do: {:ok, state}
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end

‎examples/rtcp-processing/lib/rtcp_processing/router.ex

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defmodule RtcpProcessing.Router do
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use Plug.Router
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plug(Plug.Static, at: "/", from: "assets")
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plug(:match)
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plug(:dispatch)
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get "/ws" do
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WebSockAdapter.upgrade(conn, RtcpProcessing.PeerHandler, %{}, [])
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end
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match _ do
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send_resp(conn, 404, "not found")
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end
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end

‎examples/rtcp-processing/mix.exs

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defmodule RtcpProcessing.MixProject do
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use Mix.Project
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def project do
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[
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app: :rtcp_processing,
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version: "0.1.0",
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elixir: "~> 1.15",
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start_permanent: Mix.env() == :prod,
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deps: deps()
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]
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end
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def application do
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[
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extra_applications: [:logger],
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mod: {RtcpProcessing.Application, []}
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]
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end
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defp deps do
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[
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{:plug, "~> 1.15.0"},
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{:bandit, "~> 1.2.0"},
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{:websock_adapter, "~> 0.5.0"},
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{:jason, "~> 1.4.0"},
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{:ex_webrtc, path: "../../."}
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]
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end
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end

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