[juliusmarminge]

serve-sim can be locked to WebRTC, as EAS Simulator always is. It then answers AVCC with 409 stream_transport_locked and refuses MJPEG, so the Device panel cannot show those simulators.

The stream client still asks for AVCC first, so hubs that serve it are unchanged. It switches to WebRTC on that 409, or when its MJPEG image fails in a browser that has RTCPeerConnection:

• Signalling: reads ICE servers from serve-sim's /api (streamSettings or stream), then posts one fully gathered offer (serve-sim does not take trickled ICE) to POST /helper/<udid>/webrtc/offer through the proxy. Gathering that times out with no reflexive or relay candidate (a cloud simulator needs one) is retried once with twice the deadline, 2 s then 4 s; after that the offer goes as gathered, all inside the first-frame deadline. A 409 from another viewer still negotiating is retried. stop() closes the peer and releases the session with webrtc/close.
• Rendering: the video only feeds the existing canvas sink, through requestVideoFrameCallback on a hidden in-document <video>, so the web panel and the mobile WebView keep one surface.
• New onCanvasVideo event: lets an owner that fell back to MJPEG show the canvas again. The web panel and the mobile WebView handle it. The switch shows the stream as connecting again, so a viewer that already had MJPEG frames does not stare at a blank canvas until the first WebRTC frame. After the first frame, a feed that shows none for the first-frame deadline (15 s, checked every 5 s) reconnects through the usual retry path, so a peer left disconnected cannot sit frozen while the status says streaming.
• Session ids: a v4 UUID built from getRandomValues, because crypto.randomUUID is missing on plain-HTTP LAN origins.
• Proxy: allows webrtc/offer and webrtc/close with read scope (opening a view, like the AVCC stream). It also keeps a request's content type: it was sending every proxied body as application/octet-stream, which serve-sim refuses for offers. The new proxy test caught this.

Part of built-in EAS Simulator support (stack #21847). Layer 2 of 6.

Testing:

• New stream tests (fake WebRTC browser APIs):
  • the 409 switch, the offer shape (UUID v4 session id, H.264, ICE servers from /api), frames reaching the sink, and the close with the same session id on stop(), and the gathering retry (no candidate at 2 s → retry, none at 4 s → the offer goes anyway);
  • a busy offer retried;
  • an MJPEG viewer moved back to the canvas;
  • a feed that keeps delivering frames is left alone, and one that stops is reconnected.
• All 40 stream tests, 101 server device tests and the proxy tests pass. The client-runtime, web and mobile typechecks pass.
• Against a real EAS session (from layer 5's live check): AVCC answered 409 stream_transport_locked, and the offer route answered through the proxy.
• Not yet watched in a real browser. The handoff note on feat(web): set up EAS Simulator hosts in Settings #21846 covers that, including Safari and the iOS WebView.

Done with Claude Opus 5.5 in Claude Code, running in T3 Code.

🤖 Generated with Claude Code

[github-actions]

Thread transfer impact

✅ Thread transfer remains within every enforced ceiling.

ℹ️ No successful main baseline artifact is available yet. This run establishes the initial measurement.

Provider	Metric	Main baseline	This PR	Impact	PR ceiling
Codex	Total thread wire	—	5.0 KiB	—	6.8 KiB	✅
Codex	Thread snapshot wire	—	3.8 KiB	—	4.9 KiB	✅
Codex	Live turn WebSocket wire	—	1.2 KiB	—	2.0 KiB	✅
Codex	Live turn WebSocket decoded	—	20.9 KiB	—	29.3 KiB	✅
Codex	Live turn messages	—	2	—	8	✅
Claude	Total thread wire	—	5.0 KiB	—	6.8 KiB	✅
Claude	Thread snapshot wire	—	3.8 KiB	—	4.9 KiB	✅
Claude	Live turn WebSocket wire	—	1.2 KiB	—	2.0 KiB	✅
Claude	Live turn WebSocket decoded	—	21.2 KiB	—	29.3 KiB	✅
Claude	Live turn messages	—	1	—	8	✅

Baseline: unavailable · PR result: 93e88d1 · Source CI: success

Scenario and decoded snapshot size

10 historical turns, 5 command tools per turn, 878.9 KiB retained MCP result per historical turn, and a 1.05 MiB retained result in the measured turn.

• Codex decoded thread snapshot: 108.5 KiB
• Claude decoded thread snapshot: 108.8 KiB

Updated in place by a trusted workflow. PR artifacts are strictly validated and never executed.

[macroscopeapp]

Approvability

Verdict: Needs Human Approval

Macroscope's review found this PR requires human approval — This PR adds a substantial WebRTC signaling and rendering path for iOS devices, including ICE negotiation, retries, proxy authorization, and session cleanup across client, server, web, and mobile code. Its cross-layer runtime impact and new user-facing capability warrant human review.

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• packages/client-runtime/src/device/stream.test.ts
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📝 Walkthrough

The client adds iOS WebRTC fallback, negotiation, playback, retries, and cleanup. The server proxy forwards WebRTC signaling requests. Mobile and web views restore canvas presentation after MJPEG fallback.

Changes: Device streaming

Layer / File(s)	Summary
WebRTC signaling routes apps/server/src/device/DeviceHubProxy.ts, apps/server/src/device/DeviceHubProxy.test.ts, packages/client-runtime/src/device/stream.ts	The proxy allows read-scoped WebRTC offer and close requests and preserves request content types. Client schemas define WebRTC settings and signaling responses.
WebRTC negotiation and lifecycle packages/client-runtime/src/device/stream.ts, packages/client-runtime/src/device/stream.test.ts	The client probes transport settings, switches from AVCC or MJPEG to WebRTC, negotiates H.264 playback, retries stalled or failed connections, and closes peer and server sessions. Tests cover negotiation, stalls, ICE timing, retries, and cleanup.
Canvas restoration and fallback validation packages/client-runtime/src/device/stream.ts, packages/client-runtime/src/device/stream.test.ts, apps/mobile/src/features/devices/device-stream.browser.ts, apps/web/src/components/device/DeviceStreamView.tsx, docs/internals/devices.md	The client signals when canvas video resumes. Mobile and web views restore canvas presentation after MJPEG. Documentation describes the fallback flow. Tests cover MJPEG-to-canvas replacement.

Priority: ➖ Normal

Estimated code review effort: 4 (Complex) | ~45 minutes

Change: Feature

Sequence Diagram(s)

sequenceDiagram
  participant DeviceStream
  participant DeviceHubProxy
  participant ServeSimHub
  participant CanvasSink
  DeviceStream->>ServeSimHub: Probe transport settings
  ServeSimHub-->>DeviceStream: Return WebRTC settings
  DeviceStream->>DeviceStream: Create H.264 offer and gather ICE candidates
  DeviceStream->>DeviceHubProxy: POST session-tagged WebRTC offer
  DeviceHubProxy->>ServeSimHub: Forward offer with JSON content type
  ServeSimHub-->>DeviceHubProxy: Return SDP answer
  DeviceHubProxy-->>DeviceStream: Return SDP answer
  DeviceStream->>CanvasSink: Paint received video frames
  DeviceStream-->>CanvasSink: Notify canvas video resumed

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Suggested reviewers: maria-rcks

Merge Risk: 🟡 Moderate · up to 93e88

A static simulator screen may cause unnecessary stream reconnections, and connections needing a later network candidate may fail. Resolve or explicitly accept these streaming risks before merging.

Pre-merge checks: 4

Check name	Status	Explanation
Title check		The title is concise, uses the conventional commit format, and clearly identifies the main change: WebRTC-only streaming for iOS devices.
Description check		The description clearly explains the problem, implementation, scope, verification results, and known limitation that real-browser behavior was not tested. It also identifies the related work and agent…
Linked Issues check		Check skipped because no linked issues were found for this pull request.
Out of Scope Changes check		Check skipped because no linked issues were found for this pull request.

Full details: Description check

Explanation

The description clearly explains the problem, implementation, scope, verification results, and known limitation that real-browser behavior was not tested. It also identifies the related work and agent used. The scope section references stack #21847 and handoff note #21846, although it does not include an explicit maintainer approval comment.

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Actionable comments posted: 1

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• 🪄 Fix comments: Fix CodeRabbit comments on this PR

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Inline comments:
Review comments at @packages/client-runtime/src/device/stream.ts:
- Around line 856-961: The readIosWebRtc flow needs a bounded watchdog for
streams that stop producing frames after the initial paint. Track the latest
frame time in presentFrame, start a periodic check after the first frame, and
invoke the existing retry path when no frame arrives within the bounded timeout,
including disconnected peers without retrying immediately. Clear the watchdog
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🟡 Minor · Guard frame callbacks without resetting the stall watchdog on animation… · stream.ts:898-939

packages/client-runtime/src/device/stream.ts:898-939
🩺 Stability & Availability | 🟡 Minor | ⚡ Quick win

Guard frame callbacks without resetting the stall watchdog on animation ticks.

readIosWebRtc can run in an Android WebView that supports RTCPeerConnection but does not support requestVideoFrameCallback. Guard both callback calls and use requestAnimationFrame only as a polling fallback. Reset checksWithoutFrame only when a feature-detected frame-progress signal advances, such as a playback-quality or inbound-RTP frame counter. Do not reset it on every animation tick. video.currentTime is playback position, not a reliable frame counter.

If no reliable frame-progress signal exists, leave checksWithoutFrame unchanged until the watchdog reconnects. A fallback that resets it on every animation tick would allow a frozen stream to avoid reconnection.

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Review comment at @packages/client-runtime/src/device/stream.ts around lines 898
- 939:
Update readIosWebRtc’s presentFrame and ontrack flow to feature-detect
requestVideoFrameCallback, using it when available and requestAnimationFrame
only as a polling fallback. Do not reset checksWithoutFrame on animation ticks;
reset it only when a reliable feature-detected playback-quality or inbound-RTP
frame-progress counter advances, otherwise leave it unchanged so the watchdog
can reconnect stalled streams.

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Outside diff comments:
Review comments at @packages/client-runtime/src/device/stream.ts:
- Around line 898-939: Update readIosWebRtc’s presentFrame and ontrack flow to
feature-detect requestVideoFrameCallback, using it when available and
requestAnimationFrame only as a polling fallback. Do not reset
checksWithoutFrame on animation ticks; reset it only when a reliable
feature-detected playback-quality or inbound-RTP frame-progress counter
advances, otherwise leave it unchanged so the watchdog can reconnect stalled
streams.

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Actionable comments posted: 1

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Inline comments:
Review comments at @packages/client-runtime/src/device/stream.ts:
- Around line 965-967: Update the candidate-readiness guard in the device stream
flow to retry whenever ICE gathering has not completed, using the existing
gathered state rather than requiring the local SDP to lack candidates. Keep the
existing retry message and behavior unchanged, and do not add a separate guard
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