[wip] now display user's camera

next tricky thing is connecting users

priv/static/js/dist/webrtc.d.ts

@@ -10,5 +10,13 @@
  * @module
  */
 declare function main(): Promise<void>;
-declare function handle_join(init: HTMLDivElement, init_name: HTMLInputElement, peers: HTMLDivElement, ws: WebSocket): Promise<void>;
-declare function ws_send_json(ws: WebSocket, x: any): void;
+/**
+ * Try to get the user's camera
+ *
+ * tries to get screenshare if user camera not available for whatever reason
+ *
+ * chimps out if
+ * - not on https
+ * - user says no
+ */
+declare function getUserCamera(): Promise<MediaStream>;

priv/static/js/dist/webrtc.js

@@ -12,34 +12,30 @@
  */
 main();
 async function main() {
-    // start websocket immediately
-    let ws = new WebSocket('/ws/webrtc');
-    // grab document elements
-    let init = document.getElementById('init');
-    let peers = document.getElementById('peers');
-    let init_name = document.getElementById('init-name');
-    let init_join = document.getElementById('init-join');
-    // handle button click
-    init_join.addEventListener('click', function () {
-        handle_join(init, init_name, peers, ws);
-    });
-    // handle message from ws
-    ws.onopen = function (e) { console.log('ws open:', e); };
-    ws.onclose = function (e) { console.warn('ws closed:', e); };
-    ws.onerror = function (e) { console.error('ws error:', e); };
-    ws.onmessage = function (e) { console.log('ws message', e); };
+    console.log('peepy');
+    let video = await getUserCamera();
+    console.log('poopy');
+    console.log('poopu');
+    let moi = document.getElementById('me');
+    moi.srcObject = video;
+    console.log('poopa');
 }
-async function handle_join(init, init_name, peers, ws) {
-    console.log('connecting...');
-    let user_name = init_name.value.trim();
-    console.log('username:', user_name);
-    ws_send_json(ws, ['username', user_name]);
-    init.hidden = true;
-    peers.hidden = false;
-}
-function ws_send_json(ws, x) {
-    let s = JSON.stringify(x, undefined, 4);
-    console.log('sending:\n', s);
-    ws.send(s);
+/**
+ * Try to get the user's camera
+ *
+ * tries to get screenshare if user camera not available for whatever reason
+ *
+ * chimps out if
+ * - not on https
+ * - user says no
+ */
+async function getUserCamera() {
+    if (!(navigator.mediaDevices)) {
+        console.error('navigator.mediaDevices is null; user not on https or something');
+    }
+    let availableDevices = await navigator.mediaDevices.enumerateDevices();
+    console.log(availableDevices);
+    return await navigator.mediaDevices.getUserMedia({ video: true, audio: false });
+    // return await navigator.mediaDevices.getDisplayMedia();
 }
 //# sourceMappingURL=webrtc.js.map

priv/static/js/dist/webrtc.js.map

@@ -1 +1 @@
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+{"version":3,"file":"webrtc.js","sourceRoot":"","sources":["../ts/webrtc.ts"],"names":[],"mappings":";AAAA;;;;;;;;;;GAUG;AAEH,IAAI,EAAE,CAAC;AAEP,KAAK,UACL,IAAI;IAIA,OAAO,CAAC,GAAG,CAAC,OAAO,CAAC,CAAC;IACrB,IAAI,KAAK,GAAsB,MAAM,aAAa,EAAE,CAAC;IACrD,OAAO,CAAC,GAAG,CAAC,OAAO,CAAC,CAAC;IACrB,OAAO,CAAC,GAAG,CAAC,OAAO,CAAC,CAAC;IACrB,IAAI,GAAG,GAAG,QAAQ,CAAC,cAAc,CAAC,IAAI,CAAqB,CAAC;IAC5D,GAAG,CAAC,SAAS,GAAG,KAAK,CAAC;IACtB,OAAO,CAAC,GAAG,CAAC,OAAO,CAAC,CAAC;AACzB,CAAC;AAGD;;;;;;;;GAQG;AACH,KAAK,UACL,aAAa;IAIT,IAAI,CAAC,CAAC,SAAS,CAAC,YAAY,CAAC,EAAE;QAC3B,OAAO,CAAC,KAAK,CAAC,gEAAgE,CAAC,CAAC;KACnF;IAED,IAAI,gBAAgB,GAChB,MAAM,SAAS,CAAC,YAAY,CAAC,gBAAgB,EAAE,CAAC;IACpD,OAAO,CAAC,GAAG,CAAC,gBAAgB,CAAC,CAAC;IAE9B,OAAO,MAAM,SAAS,CAAC,YAAY,CAAC,YAAY,CAAC,EAAC,KAAK,EAAE,IAAI,EAAE,KAAK,EAAE,KAAK,EAAC,CAAC,CAAC;IAC9E,yDAAyD;AAC7D,CAAC"}

priv/static/js/ts/webrtc.ts

@@ -17,57 +17,38 @@ main
     ()
     : Promise<void>
 {
-    // start websocket immediately
-    let ws = new WebSocket('/ws/webrtc');
-
-    // grab document elements
-    let init  = document.getElementById('init')    as HTMLDivElement;
-    let peers = document.getElementById('peers')   as HTMLDivElement;
-
-    let init_name = document.getElementById('init-name') as HTMLInputElement;
-    let init_join = document.getElementById('init-join') as HTMLButtonElement;
-
-    // handle button click
-    init_join.addEventListener('click',
-        function() {
-            handle_join(init, init_name, peers, ws);
-        }
-    );
-
-    // handle message from ws
-    ws.onopen    = function(e) { console.log('ws open:', e); };
-    ws.onclose   = function(e) { console.warn('ws closed:', e); };
-    ws.onerror   = function(e) { console.error('ws error:', e); };
-    ws.onmessage = function(e) { console.log('ws message', e); };
+    console.log('peepy');
+    let video : MediaStream      = await getUserCamera();
+    console.log('poopy');
+    console.log('poopu');
+    let moi = document.getElementById('me') as HTMLVideoElement;
+    moi.srcObject = video;
+    console.log('poopa');
 }
 
+
+/**
+ * Try to get the user's camera
+ *
+ * tries to get screenshare if user camera not available for whatever reason
+ *
+ * chimps out if
+ * - not on https
+ * - user says no
+ */
 async function
-handle_join
-    (init      : HTMLDivElement,
-     init_name : HTMLInputElement,
-     peers     : HTMLDivElement,
-     ws        : WebSocket)
-    : Promise<void>
+getUserCamera
+    ()
+    : Promise<MediaStream>
 {
-    console.log('connecting...');
-    let user_name: string = init_name.value.trim();
-    console.log('username:', user_name);
+    if (!(navigator.mediaDevices)) {
+        console.error('navigator.mediaDevices is null; user not on https or something');
+    }
 
-    ws_send_json(ws, ['username', user_name]);
+    let availableDevices: Array<MediaDeviceInfo> =
+        await navigator.mediaDevices.enumerateDevices();
+    console.log(availableDevices);
 
-    init.hidden = true;
-    peers.hidden = false;
-}
-
-
-function
-ws_send_json
-    (ws : WebSocket,
-     x  : any)
-    : void
-{
-    let s: string = JSON.stringify(x, undefined, 4);
-    console.log('sending:\n', s);
-
-    ws.send(s);
+    return await navigator.mediaDevices.getUserMedia({video: true, audio: false});
+    // return await navigator.mediaDevices.getDisplayMedia();
 }

priv/static/webrtc.html

@@ -15,17 +15,10 @@
     <div class="content">
         <h1 class="content-title">FEWD: webrtc demo</h1>
 
-        <div id="init">
-            <label for="init-name">username:</label>
-            <input autofocus id="init-name" type="text" value='alice'>
-            <button id="init-join">connect</button>
-        </div>
+        <div class="content-body">
+            <textarea disabled id="wfc-output"></textarea>
 
-
-        <div id="peers" hidden>
-            <h1>Peers</h1>
-
-            <ul id="peers-ul"></ul>
+            <video id="me" autoplay muted></video>
         </div>
     </div>
 

scratch/grok-webrtc.html

@@ -0,0 +1,166 @@
+<!DOCTYPE html>
+<html lang="en">
+<head>
+  <meta charset="UTF-8" />
+  <title>Super Simple WebRTC – Manual Signaling</title>
+  <style>
+    body { font-family: sans-serif; margin: 20px; }
+    textarea { width: 100%; height: 90px; font-family: monospace; }
+    button { margin: 6px 0; padding: 8px 16px; }
+    video { max-width: 320px; border: 1px solid #444; margin: 8px; background: #000; }
+  </style>
+</head>
+<body>
+
+<h2>WebRTC 1:1 test (copy-paste signaling)</h2>
+
+<div>
+  <button id="btnCreateOffer">1. Create Offer (Peer A)</button>
+  <button id="btnSetRemoteOffer">2. Paste offer → Set remote description</button><br>
+  <textarea id="offerSdp" placeholder="Offer SDP will appear here (Peer A) or paste offer here (Peer B)"></textarea>
+</div>
+
+<div style="margin-top: 24px;">
+  <button id="btnCreateAnswer">3. Create Answer (Peer B)</button>
+  <button id="btnSetRemoteAnswer">4. Paste answer → Set remote description</button><br>
+  <textarea id="answerSdp" placeholder="Answer SDP will appear here (Peer B) or paste answer here (Peer A)"></textarea>
+</div>
+
+<div style="margin-top: 20px;">
+  <button id="btnAddIce">Add ICE candidate manually (if needed)</button><br>
+  <textarea id="iceInput" placeholder="Paste remote ICE candidate here"></textarea>
+  <div id="iceLog"></div>
+</div>
+
+<div style="margin-top: 20px;">
+  <video id="localVideo" autoplay playsinline muted></video>
+  <video id="remoteVideo" autoplay playsinline></video>
+</div>
+
+<pre id="status">Status: ready</pre>
+
+<script>
+// ────────────────────────────────────────────────
+const $ = s => document.querySelector(s);
+const status = $('pre#status');
+const offerArea  = $('#offerSdp');
+const answerArea = $('#answerSdp');
+const iceInput   = $('#iceInput');
+const iceLog     = $('#iceLog');
+
+let pc = null;
+let localStream = null;
+
+// ────────────────────────────────────────────────
+async function initPeerConnection() {
+  if (pc) pc.close();
+
+  pc = new RTCPeerConnection({
+    iceServers: [
+      { urls: "stun:stun.l.google.com:19302" },
+      { urls: "stun:stun.stunprotocol.org" }
+    ]
+  });
+
+  pc.onicecandidate = e => {
+    if (e.candidate) {
+      console.log("New ICE candidate:", e.candidate);
+      iceLog.innerHTML += "<div style='color:#c33'>→ " + e.candidate.candidate + "</div>";
+    }
+  };
+
+  pc.oniceconnectionstatechange = () => {
+    status.textContent = "ICE state: " + pc.iceConnectionState;
+  };
+
+  pc.ontrack = e => {
+    console.log("ontrack", e);
+    const remoteVideo = $('#remoteVideo');
+    remoteVideo.srcObject = e.streams[0];
+    status.textContent = "ICE state: " + pc.iceConnectionState + " (receiving media!)";
+  };
+
+  // Add local stream if we have camera/mic
+  if (localStream) {
+    localStream.getTracks().forEach(track => {
+      pc.addTrack(track, localStream);
+    });
+  }
+}
+
+// ────────────────────────────────────────────────
+async function startLocalCamera() {
+  try {
+    localStream = await navigator.mediaDevices.getUserMedia({
+      video: true,
+      audio: false    // change to true if you want audio too
+    });
+    $('#localVideo').srcObject = localStream;
+  } catch (err) {
+    console.error("Camera access failed", err);
+    status.textContent = "Camera access failed: " + err.message;
+  }
+}
+
+// ────────────────────────────────────────────────
+$('#btnCreateOffer').onclick = async () => {
+  await initPeerConnection();
+  await startLocalCamera();
+
+  const offer = await pc.createOffer();
+  await pc.setLocalDescription(offer);
+
+  offerArea.value = JSON.stringify(pc.localDescription, null, 2);
+  status.textContent = "Offer created – copy and send to peer B";
+};
+
+// ────────────────────────────────────────────────
+$('#btnCreateAnswer').onclick = async () => {
+  await initPeerConnection();
+  await startLocalCamera();
+
+  const offer = JSON.parse(offerArea.value);
+  await pc.setRemoteDescription(offer);
+
+  const answer = await pc.createAnswer();
+  await pc.setLocalDescription(answer);
+
+  answerArea.value = JSON.stringify(pc.localDescription, null, 2);
+  status.textContent = "Answer created – copy and send to peer A";
+};
+
+// ────────────────────────────────────────────────
+$('#btnSetRemoteOffer').onclick = async () => {
+  if (!pc) await initPeerConnection();
+
+  const remote = JSON.parse(offerArea.value);
+  await pc.setRemoteDescription(remote);
+  status.textContent = "Remote offer set";
+};
+
+// ────────────────────────────────────────────────
+$('#btnSetRemoteAnswer').onclick = async () => {
+  const remote = JSON.parse(answerArea.value);
+  await pc.setRemoteDescription(remote);
+  status.textContent = "Remote answer set – waiting for ICE/media";
+};
+
+// ────────────────────────────────────────────────
+$('#btnAddIce').onclick = async () => {
+  if (!iceInput.value.trim()) return;
+  try {
+    const candidate = new RTCIceCandidate(JSON.parse(iceInput.value));
+    await pc.addIceCandidate(candidate);
+    iceLog.innerHTML += "<div style='color:#3a3'>← added ICE candidate</div>";
+    iceInput.value = "";
+  } catch (err) {
+    console.error("Bad ICE candidate", err);
+  }
+};
+
+// Optional: auto-start camera when page loads
+// startLocalCamera();
+
+</script>
+</body>
+</html>

scratch/webrtc01.ts

@@ -0,0 +1,127 @@
+/**
+ * webrtc page script
+ *
+ * Author: Peter Harpending <peterharpending@qpq.swiss>
+ * Date: 2026-02-04
+ * Copyright: Copyright (c) 2026 QPQ AG
+ *
+ * Reference: https://git.qpq.swiss/QPQ-AG/research-megadoc/src/commit/c7c4592d4b21ad120145ef63334471a1a7ec1e60/paste/2026-02/grok-webrtc.html
+ *
+ * @module
+ */
+
+type state =
+    {whoami: string,
+     roster: Array<string>};
+
+let st: state =
+    {whoami: "",
+     roster: []};
+
+main();
+
+async function
+main
+    ()
+    : Promise<void>
+{
+    // start websocket immediately
+    let ws: WebSocket;
+    let init = document.getElementById('init') as HTMLDivElement;
+    let init_join = document.getElementById('init-join') as HTMLButtonElement;
+
+    // handle button click
+    init_join.addEventListener('click',
+        function() {
+            init.hidden = true;
+            document.getElementById('run')!.hidden = false;
+            ws = new WebSocket('/ws/webrtc');
+            ws.onopen    = function(e) { console.log('ws open'); };
+            ws.onclose   = function(e) { console.warn('ws closed'); };
+            ws.onerror   = function(e) { console.error('ws error:', e); };
+            ws.onmessage =
+                function(e) {
+                    // console.log('ws message:', e.data);
+                    let message = JSON.parse(e.data) as ws_msg;
+                    handle_ws_msg(message);
+                };
+        }
+    );
+}
+
+type ws_msg = ["whoami", string]
+            | ["roster", Array<string>];
+
+function
+handle_ws_msg
+    (message: ws_msg)
+    : void
+{
+    console.log('message from server:', message);
+    switch(message[0]) {
+        case "whoami":
+            st.whoami = message[1];
+            break;
+        case "roster":
+            st.roster = (message[1] as Array<string>);
+            break;
+        default:
+            console.warn("unknown message", message);
+    }
+
+    render_state();
+}
+
+
+
+function
+ws_send_json
+    (ws : WebSocket,
+     x  : any)
+    : void
+{
+    let s: string = JSON.stringify(x, undefined, 4);
+    console.log('sending:\n', s);
+
+    ws.send(s);
+}
+
+
+
+function
+render_state
+    ()
+    : void
+{
+    console.log('st', st);
+    // whoami
+    document.getElementById('whoami')!.innerText = st.whoami;
+    //
+    let roster_ul = document.getElementById('roster-ul') as HTMLUListElement;
+    let newChildren : Array<HTMLElement> = [];
+    for (let nick of st.roster) {
+        if (!(nick === st.whoami)) {
+            let li = nickkk(nick);
+            newChildren.push(li);
+        }
+    }
+    roster_ul.replaceChildren(...newChildren);
+}
+
+function
+nickkk
+    (nick: string)
+    : HTMLElement
+{
+    let li = document.createElement('li');
+
+    li.innerText += nick;
+    li.innerText += ' ';
+
+    let call_a = document.createElement('button');
+    call_a.innerText = 'call';
+
+    li.appendChild(call_a);
+
+    return li;
+}

src/fd_httpd.erl

@@ -23,6 +23,12 @@ start_link() ->
 
 init([]) ->
     RestartStrategy = {one_for_one, 1, 60},
+    WebRTC    = {fd_httpd_webrtc,
+                 {fd_httpd_webrtc, start_link, []},
+                 permanent,
+                 5000,
+                 worker,
+                 [fd_httpd_webrtc]},
     FileCache = {fd_httpd_sfc,
                  {fd_httpd_sfc, start_link, []},
                  permanent,
@@ -35,5 +41,6 @@ init([]) ->
                  5000,
                  supervisor,
                  [fd_httpd_clients]},
-    Children  = [FileCache, Clients],
+    Children  = [WebRTC, FileCache, Clients],
+    %Children  = [FileCache, Clients],
     {ok, {RestartStrategy, Children}}.

src/fd_httpd_client.erl

@@ -138,14 +138,16 @@ loop(Parent, Debug, State = #s{socket = Socket, next = Next0}) ->
                     %% FIXME: unfuck received logic here
                     %% handle_request should eventually call back into
                     %% loop/3 or close the socket
+                    %%
+                    %% also loop/3 should be loop/1, because we need to pass
+                    %% the state down the callchains
                     Next2 =
                         case Next1 of
                             none -> <<>>;
                             Bin  -> Bin
                         end,
-                    Next3 = handle_request(Socket, Req, Next2),
-                    NewState = State#s{next = Next3},
-                    loop(Parent, Debug, NewState);
+                    NewState = State#s{next = Next2},
+                    handle_request(Req, Parent, Debug, NewState);
                 Error ->
                     %% should trigger bad request
                     tell(error, "~p QHL parse error: ~tp", [?LINE, Error]),
@@ -218,41 +220,44 @@ system_replace_state(StateFun, State) ->
 
 %%% http request handling
 
--spec handle_request(Sock, Request, Received) -> NewReceived
-    when Sock        :: gen_tcp:socket(),
-         Request     :: request(),
-         Received    :: binary(),
-         NewReceived :: binary().
+-spec handle_request(Request, Parent, Debug, State) -> no_return()
+    when Request :: request(),
+         Parent  :: pid(),
+         Debug   :: [sys:dbg_opt()],
+         State   :: state().
 
-handle_request(Sock, R = #request{method = M, path = P}, Received) when M =/= undefined, P =/= undefined ->
-    tell("~tp ~tp ~ts", [self(), M, P]),
-    route(Sock, M, P, R, Received).
+handle_request(Req = #request{method = Method, path = Path}, Parent, Debug, State)
+        when Method =/= undefined,
+             Path   =/= undefined ->
+    tell("~tp ~tp ~ts", [self(), Method, Path]),
+    route(Method, Path, Req, Parent, Debug, State).
 
 
-
--spec route(Sock, Method, Route, Request, Received) -> NewReceived
-    when Sock        :: gen_tcp:socket(),
-         Method      :: get | post,
-         Route       :: binary(),
-         Request     :: request(),
-         Received    :: binary(),
-         NewReceived :: binary().
-
-route(Sock, get, Route, Request, Received) ->
+% hardcode routes first
+route(get, Route, Request, P, D, S = #s{socket = Sock}) ->
     case Route of
-         <<"/ws/echo">>   -> ws_echo(Sock, Request)                 , Received;
-         <<"/ws/webrtc">> -> ws_webrtc(Sock, Request, Received)     , Received;
-         <<"/">>          -> route_static(Sock, <<"/index.html">>)  , Received;
-         _                -> route_static(Sock, Route)              , Received
+         <<"/ws/echo">>   -> ws_echo(Sock, Request);
+         <<"/ws/webrtc">> ->
+            ws_webrtc(Request, S);
+         <<"/">> ->
+            route_static(Sock, <<"/index.html">>),
+            loop(P, D, S);
+         _ ->
+            route_static(Sock, Route),
+            loop(P, D, S)
     end;
-route(Sock, post, Route, Request, Received) ->
+route(post, Route, Request, P, D, S = #s{socket = Sock}) ->
     case Route of
-         <<"/wfcin">>       -> wfcin(Sock, Request) , Received;
-         _                  -> fd_httpd_utils:http_err(Sock, 404)  , Received
+         <<"/wfcin">> ->
+            wfcin(Sock, Request),
+            loop(P, D, S);
+         _ ->
+            fd_httpd_utils:http_err(Sock, 404),
+            loop(P, D, S)
     end;
-route(Sock, _, _, _, Received) ->
+route(_, _, _, P, D, S = #s{socket = Sock}) ->
     fd_httpd_utils:http_err(Sock, 404),
-    Received.
+    loop(P, D, S).
 
 
 
@@ -292,46 +297,63 @@ respond_static(Sock, not_found) ->
 %% webrtc
 %% ------------------------------
 
--record(rs,
-        {received = <<>>      :: binary(),
-         username = undefined :: undefined | string(),
-         peers    = undefined :: undefined | [string()]}).
-
--type webrtc_state() :: #rs{}.
-
-ws_webrtc(Sock, Request, Received) ->
+ws_webrtc(Request, State = #s{socket = Sock}) ->
     try
         case qhl_ws:handshake(Request) of
             {ok, Response} ->
                 fd_httpd_utils:respond(Sock, Response),
-                ws_webrtc_loop(Sock, #rs{received = Received});
+                % wait for username
+                ok = fd_httpd_webrtc:join(),
+                ws_webrtc_loop(State);
             Error ->
                 tell("ws_webrtc: error: ~tp", [Error]),
-                fd_httpd_utils:http_err(Sock, 400)
+                fd_httpd_utils:http_err(Sock, 400),
+                exit(bad_request)
         end
     catch
-        X:Y:Z ->
-            tell(error, "CRASH ws_webrtc: ~tp:~tp:~tp", [X, Y, Z]),
-            fd_httpd_utils:http_err(Sock, 500)
+        exit:Y:Z ->
+            tell("~tp ws_webrtc: dying peacefully: ~tp:~tp", [self(), Y, Z]),
+            gen_tcp:close(Sock);
+        error:Y:Z ->
+            tell(error, "~tp ws_webrtc: ERROR ~tp:~tp", [self(), Y, Z]),
+            fd_httpd_utils:http_err(Sock, 500),
+            error(internal_server_error)
     end.
 
 
--define(WEBRTC_TIMEOUT, 30*qhl_ws:min()).
+-spec ws_webrtc_loop(state()) -> no_return().
+
+% we have no tcp bytes waiting to be parsed
+ws_webrtc_loop(State = #s{socket = Socket, next = <<>>}) ->
+    ok = inet:setopts(Socket, [{active, once}]),
+    receive
+        {tcp, Socket, Message} ->
+            NewState = State#s{next = Message},
+            ws_webrtc_loop(NewState);
+        {webrtc, Message} ->
+            ok = ws_webrtc_relay(Socket, s2c, Message),
+            ws_webrtc_loop(State);
+        {tcp_closed, Socket} ->
+            ok = tell("~p Socket closed, retiring.~n", [self()]),
+            exit(normal)
+    end;
+% we have TCP bytes sitting and waiting to be parsed
+ws_webrtc_loop(State = #s{socket = Sock, next = Recv}) ->
+    {ok, Message, NewRecv} = qhl_ws:recv_json(Sock, Recv, 5000),
+    ok = ws_webrtc_relay(Sock, c2s, Message),
+    ws_webrtc_loop(State#s{next = NewRecv}).
+
+
+% relay
+% message is a zj:value()
+ws_webrtc_relay(Socket, Direction, ZJValue) ->
+    tell("~p ws_webrtc_relay ~p: ~p", [self(), Direction, ZJValue]),
+    case Direction of
+        c2s -> fd_httpd_webrtc:hi(ZJValue);
+        s2c -> qhl_ws:send_json(Socket, ZJValue)
+    end,
+    ok.
 
-%% first thing is to get username
-ws_webrtc_loop(Socket, State = #rs{received = Recv,
-                                   username = undefined}) ->
-    %Foo = qhl_ws:recv_json(Socket, Recv, ?WEBRTC_TIMEOUT),
-    %tell("~p recv_json: ~p", [self(), Foo]),
-    %error(ur_mom);
-    {ok, ["username", Username], NewRecv} = qhl_ws:recv_json(Socket, Recv, ?WEBRTC_TIMEOUT),
-    tell("~p ws_webrtc_loop: request username: ~p", [self(), Username]),
-    NewState = State#rs{received = NewRecv,
-                        username = Username},
-    ws_webrtc_loop(Socket, NewState);
-ws_webrtc_loop(Socket, State = _) ->
-    tell("~p ws_webrtc_loop nyi state: ~p", [self(), State]),
-    error(nyi).
 
 
 %% ------------------------------

src/fd_httpd_webrtc.erl

@@ -0,0 +1,165 @@
+% @doc webrtc peer pool
+-module(fd_httpd_webrtc).
+
+-behavior(gen_server).
+
+%% api (caller context)
+-export([
+    join/0, hi/1
+]).
+
+% startup/gen_server callbacks
+-export([
+    % caller context
+    start_link/0,
+    % process context
+    init/1, handle_call/3, handle_cast/2, handle_info/2,
+    code_change/3, terminate/2
+]).
+
+-include("$zx_include/zx_logger.hrl").
+
+
+-record(u, {pid :: pid()}).
+-type user() :: #u{}.
+
+-record(s, {roster = [] :: [user()]}).
+-type state() :: #s{}.
+
+
+%%-----------------------------------------------------------------------------
+%% caller context
+%%-----------------------------------------------------------------------------
+
+
+start_link() ->
+    gen_server:start_link({local, ?MODULE}, ?MODULE, none, []).
+
+
+
+-spec join() -> ok | {error, any()}.
+
+join() ->
+    gen_server:call(?MODULE, join).
+
+
+
+-spec hi(Message) -> ok
+    when Message :: zj:value().
+
+hi(Message) ->
+    gen_server:cast(?MODULE, {hi, self(), Message}).
+
+
+
+%%-----------------------------------------------------------------------------
+%% process context
+%%-----------------------------------------------------------------------------
+
+%% gen_server callbacks
+
+init(none) ->
+    tell("starting fd_httpd_webrtc", []),
+    InitState = #s{},
+    {ok, InitState}.
+
+
+handle_call(join, _From = {PID, _Tag}, State) ->
+    {Reply, NewState} = do_join(PID, State),
+    {reply, Reply, NewState};
+handle_call(Unexpected, From, State) ->
+    tell("~tp: unexpected call from ~tp: ~tp", [?MODULE, Unexpected, From]),
+    {noreply, State}.
+
+
+handle_cast(Unexpected, State) ->
+    tell("~tp: unexpected cast: ~tp", [?MODULE, Unexpected]),
+    {noreply, State}.
+
+
+handle_info({'DOWN', _Ref, process, PID, _Reason}, State) ->
+    NewState = do_down(PID, State),
+    {noreply, NewState};
+handle_info(Unexpected, State) ->
+    tell("~tp: unexpected info: ~tp", [?MODULE, Unexpected]),
+    {noreply, State}.
+
+
+code_change(_, State, _) ->
+    {ok, State}.
+
+terminate(_, _) ->
+    ok.
+
+
+%%---------------------
+%% doers
+%%---------------------
+
+-spec do_join(PID, State) -> {Reply, NewState}
+    when PID      :: pid(),
+         State    :: state(),
+         Reply    :: ok
+                   | {error, any()},
+         NewState :: state().
+% @private join a user to a pool
+
+do_join(PID, State = #s{roster = Users}) ->
+    % see if pid is already there
+    case lists:keymember(PID, #u.pid, Users) of
+        true -> {error, already_joined};
+        false -> do_join2(PID, State)
+    end.
+
+do_join2(PID, State = #s{roster = Users}) ->
+    monitor(process, PID),
+    NewUser = #u{pid = PID},
+    NewRoster = usort([NewUser | Users]),
+    NewState  = State#s{roster = NewRoster},
+    ok = whisper(whoami, PID),
+    ok = gossip(roster, NewState),
+    {ok, NewState}.
+
+usort(Users) ->
+    lists:keysort(#u.pid, Users).
+
+
+-spec do_down(PID, State) -> NewState
+    when PID      :: pid(),
+         State    :: state(),
+         NewState :: state().
+% @private handle a user dying
+
+do_down(PID, State = #s{roster = Users}) ->
+    % remove from roster
+    NewUsers = lists:keydelete(PID, #u.pid, Users),
+    NewState = State#s{roster = NewUsers},
+    % broadcast username
+    ok = gossip(roster, NewState),
+    NewState.
+
+
+% @doc send a message to a specific person
+whisper(whoami, PID) ->
+    PID ! {webrtc, ["whoami", pidstr(PID)]},
+    ok.
+
+
+% @doc send a message to everyone
+gossip(roster, State = #s{roster = Us}) ->
+    Pids = [pidstr(PID) || #u{pid = PID} <- Us],
+    gossip_json(["roster", Pids], State).
+
+pidstr(PID) ->
+    unicode:characters_to_list(io_lib:format("~tp", [PID])).
+
+
+-spec gossip_json(zj:value(), state()) -> ok.
+% @private gossip a message to everyone
+
+gossip_json(Message, #s{roster = Users}) ->
+    GossipTo =
+        fun(#u{pid = PID}) ->
+            PID ! {webrtc, Message}
+        end,
+    lists:foreach(GossipTo, Users).