[wip] now display user's camera

next tricky thing is connecting users

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

@@ -10,7 +10,13 @@
  * @module
  */
 declare function main(): Promise<void>;
-type ws_msg = ["username", string] | ["users", Array<string>];
-declare function handle_ws_msg(message: ws_msg, roster_ul: HTMLUListElement, whoami: HTMLHeadingElement): 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

@@ -12,55 +12,30 @@
  */
 main();
 async function main() {
-    // start websocket immediately
-    let ws = new WebSocket('/ws/webrtc');
-    // grab document elements
-    let init = document.getElementById('init');
-    let roster = document.getElementById('roster');
-    let roster_ul = document.getElementById('roster-ul');
-    let whoami = document.getElementById('whoami');
-    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, roster, 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.data);
-            let message = JSON.parse(e.data);
-            handle_ws_msg(message, roster_ul, whoami);
-        };
+    console.log('peepy');
+    let video = await getUserCamera();
+    console.log('poopy');
+    console.log('poopu');
+    let moi = document.getElementById('me');
+    moi.srcObject = video;
+    console.log('poopa');
 }
-function handle_ws_msg(message, roster_ul, whoami) {
-    switch (message[0]) {
-        case "username":
-            whoami.innerText = 'Whoami: ' + message[1];
-            break;
-        case "users":
-            for (let uname of message[1]) {
-                let thisli = document.createElement('li');
-                thisli.innerText = uname;
-                roster_ul.appendChild(thisli);
+/**
+ * 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');
     }
-            break;
-    }
-}
-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);
+    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 @@
-{"version":3,"file":"webrtc.js","sourceRoot":"","sources":["../ts/webrtc.ts"],"names":[],"mappings":";AAAA;;;;;;;;;;GAUG;AAEH,IAAI,EAAE,CAAC;AAEP,KAAK,UACL,IAAI;IAIA,8BAA8B;IAC9B,IAAI,EAAE,GAAG,IAAI,SAAS,CAAC,YAAY,CAAC,CAAC;IAErC,yBAAyB;IACzB,IAAI,IAAI,GAAQ,QAAQ,CAAC,cAAc,CAAC,MAAM,CAAwB,CAAC;IACvE,IAAI,MAAM,GAAM,QAAQ,CAAC,cAAc,CAAC,QAAQ,CAAsB,CAAC;IACvE,IAAI,SAAS,GAAG,QAAQ,CAAC,cAAc,CAAC,WAAW,CAAqB,CAAC;IACzE,IAAI,MAAM,GAAM,QAAQ,CAAC,cAAc,CAAC,QAAQ,CAA0B,CAAC;IAE3E,IAAI,SAAS,GAAG,QAAQ,CAAC,cAAc,CAAC,WAAW,CAAqB,CAAC;IACzE,IAAI,SAAS,GAAG,QAAQ,CAAC,cAAc,CAAC,WAAW,CAAsB,CAAC;IAE1E,sBAAsB;IACtB,SAAS,CAAC,gBAAgB,CAAC,OAAO,EAC9B;QACI,WAAW,CAAC,IAAI,EAAE,SAAS,EAAE,MAAM,EAAE,EAAE,CAAC,CAAC;IAC7C,CAAC,CACJ,CAAC;IAEF,yBAAyB;IACzB,EAAE,CAAC,MAAM,GAAM,UAAS,CAAC,IAAI,OAAO,CAAC,GAAG,CAAC,UAAU,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;IAC3D,EAAE,CAAC,OAAO,GAAK,UAAS,CAAC,IAAI,OAAO,CAAC,IAAI,CAAC,YAAY,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;IAC9D,EAAE,CAAC,OAAO,GAAK,UAAS,CAAC,IAAI,OAAO,CAAC,KAAK,CAAC,WAAW,EAAE,CAAC,CAAC,CAAC,CAAC,CAAC,CAAC;IAC9D,EAAE,CAAC,SAAS;QACR,UAAS,CAAC;YACN,sCAAsC;YACtC,IAAI,OAAO,GAAG,IAAI,CAAC,KAAK,CAAC,CAAC,CAAC,IAAI,CAAW,CAAC;YAC3C,aAAa,CAAC,OAAO,EAAE,SAAS,EAAE,MAAM,CAAC,CAAC;QAC9C,CAAC,CAAC;AACV,CAAC;AAKD,SACA,aAAa,CACR,OAAkB,EAClB,SAA4B,EAC5B,MAA8B;IAG/B,QAAO,OAAO,CAAC,CAAC,CAAC,EAAE;QACf,KAAK,UAAU;YACX,MAAM,CAAC,SAAS,GAAG,UAAU,GAAG,OAAO,CAAC,CAAC,CAAC,CAAC;YAC3C,MAAM;QACV,KAAK,OAAO;YACR,KAAK,IAAI,KAAK,IAAI,OAAO,CAAC,CAAC,CAAC,EAAE;gBAC1B,IAAI,MAAM,GAAG,QAAQ,CAAC,aAAa,CAAC,IAAI,CAAC,CAAC;gBAC1C,MAAM,CAAC,SAAS,GAAG,KAAK,CAAC;gBACzB,SAAS,CAAC,WAAW,CAAC,MAAM,CAAC,CAAC;aACjC;YACD,MAAM;KACb;AAEL,CAAC;AAID,KAAK,UACL,WAAW,CACN,IAA0B,EAC1B,SAA4B,EAC5B,KAA0B,EAC1B,EAAqB;IAGtB,OAAO,CAAC,GAAG,CAAC,eAAe,CAAC,CAAC;IAC7B,IAAI,SAAS,GAAW,SAAS,CAAC,KAAK,CAAC,IAAI,EAAE,CAAC;IAC/C,OAAO,CAAC,GAAG,CAAC,WAAW,EAAE,SAAS,CAAC,CAAC;IAEpC,YAAY,CAAC,EAAE,EAAE,CAAC,UAAU,EAAE,SAAS,CAAC,CAAC,CAAC;IAE1C,IAAI,CAAC,MAAM,GAAG,IAAI,CAAC;IACnB,KAAK,CAAC,MAAM,GAAG,KAAK,CAAC;AACzB,CAAC;AAGD,SACA,YAAY,CACP,EAAc,EACd,CAAQ;IAGT,IAAI,CAAC,GAAW,IAAI,CAAC,SAAS,CAAC,CAAC,EAAE,SAAS,EAAE,CAAC,CAAC,CAAC;IAChD,OAAO,CAAC,GAAG,CAAC,YAAY,EAAE,CAAC,CAAC,CAAC;IAE7B,EAAE,CAAC,IAAI,CAAC,CAAC,CAAC,CAAC;AACf,CAAC"}
+{"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

@@ -17,91 +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 roster    = document.getElementById('roster')    as HTMLDivElement;
-    let roster_ul = document.getElementById('roster-ul') as HTMLUListElement;
-    let whoami    = document.getElementById('whoami')    as HTMLHeadingElement;
-
-    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, roster, 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.data);
-            let message = JSON.parse(e.data) as ws_msg;
-            handle_ws_msg(message, roster_ul, whoami);
-        };
-}
-
-type ws_msg = ["username", string]
-            | ["users", Array<string>];
-
-function
-handle_ws_msg
-    (message   : ws_msg,
-     roster_ul : HTMLUListElement,
-     whoami    : HTMLHeadingElement)
-    : void
-{
-    switch(message[0]) {
-        case "username":
-            whoami.innerText = 'Whoami: ' + message[1];
-            break;
-        case "users":
-            for (let uname of message[1]) {
-                let thisli = document.createElement('li');
-                thisli.innerText = uname;
-                roster_ul.appendChild(thisli);
-            }
-            break;
-    }
-
+    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);
-
-    ws_send_json(ws, ['username', user_name]);
-
-    init.hidden = true;
-    peers.hidden = false;
+    if (!(navigator.mediaDevices)) {
+        console.error('navigator.mediaDevices is null; user not on https or something');
     }
 
+    let availableDevices: Array<MediaDeviceInfo> =
+        await navigator.mediaDevices.enumerateDevices();
+    console.log(availableDevices);
 
-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

@@ -15,19 +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="roster" hidden>
-            <h2 id="whoami">Whoami: </h2>
-
-            <h2>Roster</h2>
-
-            <ul id="roster-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_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
-    end;
-route(Sock, _, _, _, Received) ->
+         <<"/wfcin">> ->
+            wfcin(Sock, Request),
+            loop(P, D, S);
+         _ ->
             fd_httpd_utils:http_err(Sock, 404),
-    Received.
+            loop(P, D, S)
+    end;
+route(_, _, _, P, D, S = #s{socket = Sock}) ->
+    fd_httpd_utils:http_err(Sock, 404),
+    loop(P, D, S).
 
 
 
@@ -292,103 +297,65 @@ respond_static(Sock, not_found) ->
 %% webrtc
 %% ------------------------------
 
--record(rs,
-        {socket               :: gen_tcp:socket(),
-         received = <<>>      :: binary(),
-         username = 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(#rs{socket = Sock, 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().
 
-
-
--spec ws_webrtc_loop(webrtc_state()) -> no_return().
-
-%% first thing is to get username
-ws_webrtc_loop(State = #rs{socket   = Socket,
-                           received = Recv,
-                           username = undefined}) ->
-    {ok, ["username", Username], NewRecv} =
-        qhl_ws:recv_json(Socket, Recv, ?WEBRTC_TIMEOUT),
-    tell("~p ws_webrtc_loop: request username: ~p", [self(), Username]),
-    {ok, ActualUsername} = fd_httpd_webrtc:join(Username),
-    ok = qhl_ws:send_json(Socket, ["username", ActualUsername]),
-    NewState = State#rs{received = NewRecv,
-                        username = ActualUsername},
-    ws_webrtc_loop(NewState);
 % we have no tcp bytes waiting to be parsed
-ws_webrtc_loop(State = #rs{socket   = Socket,
-                           received = <<>>}) ->
+ws_webrtc_loop(State = #s{socket = Socket, next = <<>>}) ->
     ok = inet:setopts(Socket, [{active, once}]),
     receive
         {tcp, Socket, Message} ->
-            NewState = State#rs{received = Message},
+            NewState = State#s{next = Message},
             ws_webrtc_loop(NewState);
         {webrtc, Message} ->
-            NewState = ws_webrtc_handle_webrtc(Message, State),
-            ws_webrtc_loop(NewState);
+            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 = #rs{socket   = Socket,
-                           received = R}) ->
-    {ok, Message, NewR} = qhl_ws:recv_json(Socket, R, 5000),
-    NewState = ws_webrtc_handle_json(Message, State#rs{received = NewR}),
-    ws_webrtc_loop(NewState).
+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}).
 
 
-
--spec ws_webrtc_handle_webrtc(Message, State) -> NewState
-    when Message  :: any(),
-         State    :: webrtc_state(),
-         NewState :: webrtc_state().
-% @private handle a message from the server
-
-ws_webrtc_handle_webrtc(Message, State = #rs{socket = Sock}) ->
-    tell("~p received message from webrtc: ~p", [self(), Message]),
-    NewState =
-        case Message of
-            {users, Users} ->
-                ok = qhl_ws:send_json(Sock, ["users", Users]),
-                State;
-            _ ->
-                tell("~p unknown webrtc message: ~p", [self, Message])
+% 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,
-    NewState.
+    ok.
 
 
 
--spec ws_webrtc_handle_json(Message, State) -> NewState
-    when Message  :: zj:value(),
-         State    :: webrtc_state(),
-         NewState :: webrtc_state().
-% @private handle a message from the client
-
-ws_webrtc_handle_json(Message, State) ->
-    tell("~p received json message from client: ~p", [self(), Message]),
-    State.
-
-
 %% ------------------------------
 %% echo
 %% ------------------------------

src/fd_httpd_webrtc.erl

@@ -5,7 +5,7 @@
 
 %% api (caller context)
 -export([
-    join/1
+    join/0, hi/1
 ]).
 
 % startup/gen_server callbacks
@@ -20,15 +20,10 @@
 -include("$zx_include/zx_logger.hrl").
 
 
--record(u,
-        {pid      :: pid(),
-         username :: string()}).
-
+-record(u, {pid :: pid()}).
 -type user() :: #u{}.
 
--record(s,
-        {users = [] :: [user()]}).
-
+-record(s, {roster = [] :: [user()]}).
 -type state() :: #s{}.
 
 
@@ -42,10 +37,18 @@ start_link() ->
 
 
 
--spec join(Username :: string()) -> {ok, ActualUsername :: string()} | {error, any()}.
+-spec join() -> ok | {error, any()}.
 
-join(Username) ->
-    gen_server:call(?MODULE, {join, Username}).
+join() ->
+    gen_server:call(?MODULE, join).
+
+
+
+-spec hi(Message) -> ok
+    when Message :: zj:value().
+
+hi(Message) ->
+    gen_server:cast(?MODULE, {hi, self(), Message}).
 
 
 
@@ -61,8 +64,8 @@ init(none) ->
     {ok, InitState}.
 
 
-handle_call({join, Username}, _From = {PID, _Tag}, State) ->
-    {Reply, NewState} = do_join(Username, PID, State),
+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]),
@@ -93,49 +96,32 @@ terminate(_, _) ->
 %% doers
 %%---------------------
 
--spec do_join(Username, PID, State) -> {Reply, NewState}
-    when Username :: string(),
-         PID      :: pid(),
+-spec do_join(PID, State) -> {Reply, NewState}
+    when PID      :: pid(),
          State    :: state(),
-         Reply    :: {ok, ActualUsername :: string()}
+         Reply    :: ok
                    | {error, any()},
          NewState :: state().
 % @private join a user to a pool
 
-do_join(Username, PID, State = #s{users = Users}) ->
+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(Username, PID, State)
+        false -> do_join2(PID, State)
     end.
 
-do_join2(Username, PID, State = #s{users = Users}) ->
+do_join2(PID, State = #s{roster = Users}) ->
     monitor(process, PID),
-    ActualUsername = unique_username(Username, Users),
-    NewUser = #u{pid = PID, username = ActualUsername},
+    NewUser = #u{pid = PID},
     NewRoster = usort([NewUser | Users]),
-    NewState  = State#s{users = NewRoster},
-    ok = gossip_roster(NewState),
-    {{ok, ActualUsername}, NewState}.
+    NewState  = State#s{roster = NewRoster},
+    ok = whisper(whoami, PID),
+    ok = gossip(roster, NewState),
+    {ok, NewState}.
 
 usort(Users) ->
-    lists:keysort(#u.username, Users).
-
-unique_username(Username, Users) ->
-    tell("~p unique_username(~p, ~p)", [?MODULE, Username, Users]),
-    case lists:keymember(Username, #u.username, Users) of
-        true  -> unique_username(Username, 1, Users);
-        false -> Username
-    end.
-
-unique_username(Username, N, Users) ->
-    tell("~p unique_username(~p, ~p ~p)", [?MODULE, Username, N, Users]),
-    U = Username ++ integer_to_list(N),
-    case lists:keymember(U, #u.username, Users) of
-        true  -> unique_username(Username, N + 1, Users);
-        false -> U
-    end.
-
+    lists:keysort(#u.pid, Users).
 
 
 -spec do_down(PID, State) -> NewState
@@ -144,29 +130,34 @@ unique_username(Username, N, Users) ->
          NewState :: state().
 % @private handle a user dying
 
-do_down(PID, State = #s{users = Users}) ->
-    % remove from users
-    NewUsers = usort(lists:keydelete(PID, #u.pid, Users)),
-    NewState = State#s{users = NewUsers},
+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(State),
+    ok = gossip(roster, NewState),
     NewState.
 
 
-
--spec gossip_roster(state()) -> ok.
-% @private gossip the roster to everyone
-
-gossip_roster(State = #s{users = Users}) ->
-    Usernames = [Username || #u{username = Username} <- Users],
-    gossip({users, Usernames}, State).
+% @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).
 
--spec gossip(any(), state()) -> ok.
+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(Message, #s{users = Users}) ->
+gossip_json(Message, #s{roster = Users}) ->
     GossipTo =
         fun(#u{pid = PID}) ->
             PID ! {webrtc, Message}