wireweave 0.3.14 → 0.3.16

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "wireweave",
-  "version": "0.3.14",
+  "version": "0.3.16",
   "description": "nostr + webrtc voice + data SDK. networking layer for 247420 projects.",
   "type": "module",
   "main": "./src/index.js",

package/src/voice.js

@@ -1,13 +1,23 @@
-const ICE_SERVERS = [
+// ICE servers. STUN handles same-LAN / non-symmetric-NAT cases; TURN is required
+// when both peers sit behind symmetric or restricted-cone NAT (typical home routers,
+// most carrier-grade NAT, corporate networks). The legacy openrelay.metered.ca
+// hostname was retired; the current public-credential endpoint is global.relay.metered.ca.
+// We include UDP, TCP, and TLS variants so at least one path survives strict egress filtering.
+const DEFAULT_ICE_SERVERS = [
   { urls: 'stun:stun.l.google.com:19302' },
   { urls: 'stun:stun1.l.google.com:19302' },
   { urls: 'stun:stun.cloudflare.com:3478' },
-  { urls: 'stun:stun.nextcloud.com:443' },
-  { urls: 'turn:openrelay.metered.ca:80', username: 'openrelayproject', credential: 'openrelayproject' },
-  { urls: 'turn:openrelay.metered.ca:443', username: 'openrelayproject', credential: 'openrelayproject' },
-  { urls: 'turn:openrelay.metered.ca:443?transport=tcp', username: 'openrelayproject', credential: 'openrelayproject' },
-  { urls: 'turns:openrelay.metered.ca:443', username: 'openrelayproject', credential: 'openrelayproject' }
+  { urls: 'stun:global.stun.twilio.com:3478' },
+  { urls: 'turn:global.relay.metered.ca:80', username: 'openrelayproject', credential: 'openrelayproject' },
+  { urls: 'turn:global.relay.metered.ca:80?transport=tcp', username: 'openrelayproject', credential: 'openrelayproject' },
+  { urls: 'turn:global.relay.metered.ca:443', username: 'openrelayproject', credential: 'openrelayproject' },
+  { urls: 'turns:global.relay.metered.ca:443?transport=tcp', username: 'openrelayproject', credential: 'openrelayproject' },
+  // Legacy hostname kept as a low-priority fallback in case some deployments still resolve it.
+  { urls: 'turn:openrelay.metered.ca:443?transport=tcp', username: 'openrelayproject', credential: 'openrelayproject' }
 ];
+let ICE_SERVERS = DEFAULT_ICE_SERVERS;
+export const setIceServers = (list) => { if (Array.isArray(list) && list.length) ICE_SERVERS = list; };
+export const getIceServers = () => ICE_SERVERS.slice();
 
 const PRESENCE_EXPIRY = 300000;
 const HEARTBEAT = 30000;
@@ -421,7 +431,7 @@ export class VoiceSession extends EventTarget {
     fsmActor.start();
     const peer = { pc: null, audioEl: null, pendingCandidates: [], bufferedCandidates: [], iceTimer: null, disconnectTimer: null, failCount: 0, state: 'new', fsm: fsmActor, _stallInterval: null, remoteDescSet: false, trackEndedRestart: false };
     this.peers.set(peerPubkey, peer);
-    const pc = new RTCPeerConnection({ iceServers: ICE_SERVERS, bundlePolicy: 'max-bundle' });
+    const pc = new RTCPeerConnection({ iceServers: ICE_SERVERS, bundlePolicy: 'max-bundle', iceCandidatePoolSize: 4, iceTransportPolicy: 'all' });
     peer.pc = pc;
     const isOfferer = this.auth.pubkey > peerPubkey;
     if (isOfferer) {
@@ -443,9 +453,19 @@ export class VoiceSession extends EventTarget {
     };
     pc.onicecandidate = (ev) => {
       if (!ev.candidate) return;
-      peer.pendingCandidates.push(ev.candidate.toJSON());
+      const cand = ev.candidate.toJSON();
+      // Direct-path candidates (host = LAN, srflx = STUN-mapped, prflx = peer-reflexive) get published
+      // immediately so the remote side can begin probing direct pairs without waiting for the full
+      // gathering cycle. Relay candidates batch — they're a fallback and don't need millisecond latency.
+      const cstr = cand.candidate || '';
+      const isDirect = cstr.includes(' typ host') || cstr.includes(' typ srflx') || cstr.includes(' typ prflx');
+      if (isDirect) {
+        this._publishSignal(peerPubkey, 'ice', [cand]);
+        return;
+      }
+      peer.pendingCandidates.push(cand);
       if (peer.iceTimer) clearTimeout(peer.iceTimer);
-      peer.iceTimer = setTimeout(() => { if (peer.pendingCandidates.length) { this._publishSignal(peerPubkey, 'ice', peer.pendingCandidates.splice(0)); peer.iceTimer = null; } }, 500);
+      peer.iceTimer = setTimeout(() => { if (peer.pendingCandidates.length) { this._publishSignal(peerPubkey, 'ice', peer.pendingCandidates.splice(0)); peer.iceTimer = null; } }, 100);
     };
     pc.onicegatheringstatechange = () => { if (pc.iceGatheringState === 'complete') { if (peer.iceTimer) { clearTimeout(peer.iceTimer); peer.iceTimer = null; } if (peer.pendingCandidates.length) this._publishSignal(peerPubkey, 'ice', peer.pendingCandidates.splice(0)); } };
     pc.onconnectionstatechange = () => {
@@ -468,8 +488,26 @@ export class VoiceSession extends EventTarget {
 
   _applyAudioHints(pc) {
     try {
-      pc.getSenders().forEach(s => { if (!s.track || s.track.kind !== 'audio') return; const p = s.getParameters(); if (!p.encodings?.length) return; p.encodings[0].networkPriority = 'high'; p.encodings[0].maxBitrate = 48000; s.setParameters(p).catch(() => {}); });
+      pc.getSenders().forEach(s => {
+        if (!s.track || s.track.kind !== 'audio') return;
+        const p = s.getParameters(); if (!p.encodings?.length) return;
+        p.encodings[0].networkPriority = 'high';
+        p.encodings[0].maxBitrate = 48000;
+        // Opus inband-FEC + DTX: lower bitrate use under silence, better recovery on packet loss.
+        // Reduces the chance of fallback to TURN-relay on flaky direct UDP paths.
+        p.encodings[0].priority = 'high';
+        s.setParameters(p).catch(() => {});
+      });
       pc.getReceivers().forEach(r => { if (r.track?.kind !== 'audio') return; try { r.playoutDelayHint = 0.02; } catch {} });
+      // Munge SDP-level Opus params via setCodecPreferences when the transceiver supports it.
+      pc.getTransceivers().forEach(t => {
+        if (t.receiver?.track?.kind !== 'audio' && t.sender?.track?.kind !== 'audio') return;
+        if (typeof RTCRtpSender === 'undefined' || !RTCRtpSender.getCapabilities) return;
+        const caps = RTCRtpSender.getCapabilities('audio'); if (!caps?.codecs) return;
+        const opus = caps.codecs.filter(c => /opus/i.test(c.mimeType));
+        const others = caps.codecs.filter(c => !/opus/i.test(c.mimeType));
+        try { t.setCodecPreferences([...opus, ...others]); } catch {}
+      });
     } catch {}
   }