Ya/replace rtmp with srt by yagarwal1307 · Pull Request #2246 · Mentra-Community/MentraOS

yagarwal1307 · 2026-03-19T14:33:05Z

generalise streaming infra away from rtmp
Implement SRT streaming on glasses
Implement webrtc streaming on glasses
Implement signed URLs to watch the webrtc stream

github-actions · 2026-03-19T14:33:17Z

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coderabbitai · 2026-03-19T14:33:19Z

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aisraelov · 2026-03-23T01:59:45Z

Hey awesome work so far. A few things to note:

rtmpUrl/srtUrl confusion in CloudflareStreamService

Right now createLiveInput() computes the real RTMP URL into a local variable, then throws it away and stuffs the SRT URL into the rtmpUrl field:

const rtmpUrl = ${liveInput.rtmps.url}${liveInput.rtmps.streamKey}; // computed, never used

  const result: LiveInputResult = {
    rtmpUrl: srtUrl!,  // ← this is an SRT URL, not RTMP
    srtUrl,
  };

This means managed streaming can never send an actual RTMP URL anymore. Every other part of the code reads rtmpUrl expecting RTMP but gets srt://.... which works but isn't documented well.

WebRtcStreamingService isn't wired up

Here's how I'd finish the PR (probably just feed this to claude and it can 1shot... lmk what you think):

Streaming Protocol Refactor: Unified Multi-Protocol Support

Context

The current streaming pipeline was built entirely around RTMP — every field is named rtmpUrl, every message type is START_RTMP_STREAM, every handler is RtmpCommandHandler. PR #2246 tries to add SRT and WebRTC support by stuffing SRT URLs into the rtmpUrl field, which works by accident but creates a fragile, confusing system.

We need to properly support 5 streaming modes:

Livestream (managed) — SRT to Cloudflare under the hood, Cloudflare restreams via RTMP to YouTube/Twitch/etc, provides HLS/DASH playback
Direct RTMP — direct RTMP to user's endpoint
Direct SRT — direct SRT to user's endpoint
Direct WebRTC — direct WHIP to user's endpoint
Livestream WebRTC (managed) — WHIP to Cloudflare's WebRTC endpoint

WebRTC runs only on asg_client (Android glasses). Phone is command passthrough only.

This is a clean break. All RTMP-specific naming is replaced. No deprecated aliases, no backwards compat shims.

SDK Public API

// Direct streaming — you provide the URL, glasses push directly to it
session.camera.startDirectStream({
  streamUrl: "rtmp://your-server.com/live/key",   // or srt:// or https:// (WHIP)
  video?: VideoConfig,
  audio?: AudioConfig,
  sound?: boolean,
})

// Livestreaming — Cloudflare handles infra, restreaming, HLS/DASH playback
session.camera.startLivestream({
  quality?: "720p" | "1080p",
  enableWebRTC?: boolean,        // use WHIP to Cloudflare instead of SRT
  restreamDestinations?: [       // Cloudflare restreams to these
    { url: "rtmp://youtube.com/live/KEY", name: "YouTube" },
    { url: "rtmp://twitch.tv/app/KEY", name: "Twitch" },
  ],
  video?: VideoConfig,
  audio?: AudioConfig,
  sound?: boolean,
})
// Returns: { hlsUrl, dashUrl, webrtcPlaybackUrl?, streamId }

session.camera.stopDirectStream()
session.camera.stopLivestream()

Design Principle

The rtmpUrl field was always a generic "stream ingest URL." Rather than creating parallel pipelines for each protocol, we generalize the existing pipeline: rename the field, detect protocol from the URL prefix, and let the glasses-side router pick the right streamer.

Architecture

SDK App
  │
  ├─ startDirectStream({ streamUrl: "rtmp://..." })  ← direct RTMP
  ├─ startDirectStream({ streamUrl: "srt://..." })   ← direct SRT
  ├─ startDirectStream({ streamUrl: "https://..." }) ← direct WebRTC (WHIP)
  ├─ startLivestream({ restreamDestinations: [...] })← livestream (cloud picks protocol)
  │
  ▼
Cloud (DirectStreamingExtension or LivestreamExtension)
  │  validates URL prefix, creates lifecycle
  │
  ▼
WebSocket message: start_stream { streamUrl, protocol, streamId, ... }
  │
  ▼
Mobile SocketComms (passthrough) → CoreModule.startStream(params)
  │
  ▼
asg_client: StreamCommandHandler routes by protocol field
  ├─ "rtmp" / "rtmps" → RtmpStreamingService (StreamPackLite RTMP)
  ├─ "srt"            → RtmpStreamingService (StreamPackLite SRT mode)
  └─ "webrtc"         → WebRtcStreamingService (WHIP)

Livestream (Managed) Flow

Glasses ──[SRT]──> Cloudflare Live Input ──> HLS/DASH playback URLs (for apps/web)
                                           ├─> Output: rtmp://youtube.com/live/KEY
                                           ├─> Output: rtmp://twitch.tv/app/KEY
                                           └─> Output: rtmps://facebook.com/rtmp/KEY

Cloud picks ingest protocol (SRT by default, WHIP if enableWebRTC). Cloudflare handles all restreaming. Restream destinations, createOutputs(), OutputStatus, and the REST API for managing outputs all stay exactly as they are.

Changes by Layer

Layer 1: SDK Types (`cloud/packages/sdk/src/types/`)

message-types.ts — Replace RTMP message types:

// Replace these:
//   START_RTMP_STREAM = "start_rtmp_stream"
//   STOP_RTMP_STREAM = "stop_rtmp_stream"
//   KEEP_RTMP_STREAM_ALIVE = "keep_rtmp_stream_alive"
// With:
START_STREAM = "start_stream",
STOP_STREAM = "stop_stream",
KEEP_STREAM_ALIVE = "keep_stream_alive",

Update all references throughout the codebase.

messages/cloud-to-glasses.ts — Replace StartRtmpStream with:

type StreamProtocol = "rtmp" | "rtmps" | "srt" | "webrtc";

interface StartStream extends BaseMessage {
  type: CloudToGlassesMessageType.START_STREAM;
  streamUrl: string;        // ingest URL (rtmp://, srt://, https://...whip)
  protocol: StreamProtocol;  // explicit protocol
  appId: string;
  streamId?: string;
  video?: any;
  audio?: any;
  stream?: any;
  flash?: boolean;
  sound?: boolean;
}

interface StopStream extends BaseMessage {
  type: CloudToGlassesMessageType.STOP_STREAM;
  appId: string;
  streamId?: string;
}

interface KeepStreamAlive extends BaseMessage {
  type: CloudToGlassesMessageType.KEEP_STREAM_ALIVE;
  streamId: string;
  ackId: string;
}

Delete StartRtmpStream, StopRtmpStream, KeepRtmpStreamAlive.

messages/glasses-to-cloud.ts — Replace RtmpStreamStatus with:

// Replace RTMP_STREAM_STATUS = "rtmp_stream_status" with:
STREAM_STATUS = "stream_status"

Rename the interface from RtmpStreamStatus → StreamStatus. Same payload.

messages/app-to-cloud.ts — Rename request types:

// Replace RtmpStreamRequest with:
interface DirectStreamRequest {
  packageName: string;
  streamUrl: string;  // was rtmpUrl
  video?: VideoConfig;
  audio?: AudioConfig;
  stream?: StreamConfig;
  sound?: boolean;
}

// Replace ManagedStreamRequest with:
interface LivestreamRequest {
  packageName: string;
  quality?: "720p" | "1080p";
  enableWebRTC?: boolean;
  restreamDestinations?: RestreamDestination[];  // unchanged
  video?: VideoConfig;
  audio?: AudioConfig;
  stream?: StreamConfig;
  sound?: boolean;
}

Layer 2: SDK Camera Module (`cloud/packages/sdk/src/app/session/modules/camera.ts`)

Replace public API:

// Direct streaming (was startRtmpStream)
interface DirectStreamOptions {
  streamUrl: string;  // rtmp://, rtmps://, srt://, or https:// (WHIP)
  video?: VideoConfig;
  audio?: AudioConfig;
  stream?: StreamConfig;
  sound?: boolean;
}

async startDirectStream(options: DirectStreamOptions): Promise<void>
async stopDirectStream(): Promise<void>
isDirectStreaming(): boolean
getDirectStreamStatus(): StreamStatus | undefined
onDirectStreamStatus(handler: StreamStatusHandler): () => void

// Livestreaming (was startManagedStream)
interface LivestreamOptions {
  quality?: "720p" | "1080p";
  enableWebRTC?: boolean;
  restreamDestinations?: RestreamDestination[];
  video?: VideoConfig;
  audio?: AudioConfig;
  stream?: StreamConfig;
  sound?: boolean;
}

interface LivestreamResult {
  hlsUrl?: string;
  dashUrl?: string;
  webrtcPlaybackUrl?: string;
  previewUrl?: string;
  streamId: string;
}

async startLivestream(options?: LivestreamOptions): Promise<LivestreamResult>
async stopLivestream(): Promise<void>
isLivestreamActive(): boolean
getLivestreamUrls(): LivestreamResult | undefined
onLivestreamStatus(handler: (status: LivestreamStatus) => void): () => void

Delete RtmpStreamOptions, startRtmpStream(), ManagedStreamOptions, startManagedStream().

camera-managed-extension.ts — Rename file to camera-livestream-extension.ts. Rename class/types accordingly.

Layer 3: Cloud Extensions

CloudflareStreamService.ts — Fix LiveInputResult:

interface LiveInputResult {
  liveInputId: string;
  rtmpUrl: string;              // actual RTMP URL (rtmps://...)
  srtUrl?: string;              // SRT URL if available
  whipUrl?: string;             // WHIP URL if available (from liveInput.webRTC.url)
  hlsUrl: string;
  dashUrl: string;
  webrtcPlaybackUrl?: string;   // playback URL
}

Return ALL URLs from Cloudflare. Let the caller pick. No more rtmpUrl: srtUrl! hack.

Rename ManagedStreamingExtension.ts → LivestreamExtension.ts:

Class: LivestreamExtension
Choose protocol and send StartStream:
- Default: use srtUrl for ingest (better for unstable connections)
- If enableWebRTC is true AND whipUrl available: use WHIP
- Fallback to rtmpUrl if SRT not available
- Send StartStream with streamUrl + protocol

Rename UnmanagedStreamingExtension.ts → DirectStreamingExtension.ts:

Class: DirectStreamingExtension
Detect protocol from URL:
- rtmp:// → protocol "rtmp"
- rtmps:// → protocol "rtmps"
- srt:// → protocol "srt"
- https:// → protocol "webrtc" (WHIP)
Rename internal field from rtmpUrl to streamUrl
Send StartStream message

Rename ManagedStreamStatus → LivestreamStatus in cloud-to-app messages.

Layer 4: Mobile Bridge

SocketComms.ts — Replace handlers:

// Replace handle_start_rtmp_stream with:
private handle_start_stream(msg: any) {
  const streamUrl = msg.streamUrl || ""
  if (streamUrl) {
    CoreModule.startStream(msg)
  } else {
    console.log("Invalid stream request: missing streamUrl")
  }
}

// Replace handle_stop_rtmp_stream with:
private handle_stop_stream(msg: any) {
  CoreModule.stopStream()
}

// Replace handle_keep_rtmp_stream_alive with:
private handle_keep_stream_alive(msg: any) {
  CoreModule.keepStreamAlive(msg)
}

Update the message type routing map to use start_stream, stop_stream, keep_stream_alive.

Replace sendRtmpStreamStatus() → sendStreamStatus().

CoreModule.ts — Replace methods:

startStream(params: Record<string, any>): Promise<void>
stopStream(): Promise<void>
keepStreamAlive(params: Record<string, any>): Promise<void>

CoreModule.kt — Replace bridge functions:

AsyncFunction("startStream") { params: Map<String, Any> ->
  coreManager?.startStream(params.toMutableMap())
}
AsyncFunction("stopStream") {
  coreManager?.stopStream()
}
AsyncFunction("keepStreamAlive") { params: Map<String, Any> ->
  coreManager?.keepStreamAlive(params.toMutableMap())
}

CoreManager.kt — Replace delegation:

fun startStream(message: MutableMap<String, Any>) {
  sgc?.startStream(message)
}
fun stopStream() {
  sgc?.stopStream()
}
fun keepStreamAlive(message: MutableMap<String, Any>) {
  sgc?.keepStreamAlive(message)
}

Layer 5: asg_client (Glasses)

Replace RtmpCommandHandler.java with StreamCommandHandler.java:

Handles start_stream, stop_stream, keep_stream_alive
Reads protocol field from message
Routes to RtmpStreamingService (for rtmp/rtmps/srt) or WebRtcStreamingService (for webrtc)
Falls back to URL prefix detection if protocol field missing

RtmpStreamingService.java — Rename mRtmpUrl → mStreamUrl, setRtmpUrl() → setStreamUrl(). Already supports SRT via StreamPackLite internally.

WebRtcStreamingService.java — Already written in PR #2246. Wire into StreamCommandHandler routing.

ServiceConstants.java — Replace:

COMMAND_START_STREAM = "start_stream"
COMMAND_STOP_STREAM = "stop_stream"
COMMAND_KEEP_STREAM_ALIVE = "keep_stream_alive"

Layer 6: Status Reporting

Status message type: Replace RTMP_STREAM_STATUS → STREAM_STATUS ("stream_status").

Status flows back through same path:

asg_client StreamingStatusCallback
  → WebSocket → cloud glasses-message-handler
    → handleStreamStatus()
      → appManager.sendMessageToApp()

Rename sendRtmpStatusResponse → sendStreamStatusResponse in asg_client.

Files to Modify

File	Change
`cloud/packages/sdk/src/types/message-types.ts`	Replace START_RTMP_STREAM → START_STREAM, etc.
`cloud/packages/sdk/src/types/messages/cloud-to-glasses.ts`	Replace StartRtmpStream → StartStream
`cloud/packages/sdk/src/types/messages/glasses-to-cloud.ts`	Replace RtmpStreamStatus → StreamStatus
`cloud/packages/sdk/src/types/messages/app-to-cloud.ts`	RtmpStreamRequest → DirectStreamRequest, ManagedStreamRequest → LivestreamRequest
`cloud/packages/sdk/src/types/messages/cloud-to-app.ts`	ManagedStreamStatus → LivestreamStatus
`cloud/packages/sdk/src/app/session/modules/camera.ts`	startRtmpStream → startDirectStream, startManagedStream → startLivestream
`cloud/packages/sdk/src/app/session/modules/camera-managed-extension.ts`	Rename to camera-livestream-extension.ts
`cloud/packages/cloud/src/services/streaming/CloudflareStreamService.ts`	Fix LiveInputResult, return all URLs
`cloud/packages/cloud/src/services/streaming/ManagedStreamingExtension.ts`	Rename to LivestreamExtension.ts, use StartStream
`cloud/packages/cloud/src/services/session/UnmanagedStreamingExtension.ts`	Rename to DirectStreamingExtension.ts, use StartStream
`cloud/packages/cloud/src/services/session/handlers/glasses-message-handler.ts`	Rename handler, use stream_status
`cloud/packages/cloud/src/services/streaming/StreamRegistry.ts`	Update field names
`mobile/src/services/SocketComms.ts`	Replace handle_start_rtmp_stream → handle_start_stream
`mobile/modules/core/src/CoreModule.ts`	Replace startRtmpStream → startStream
`mobile/modules/core/android/.../CoreModule.kt`	Replace bridge functions
`mobile/modules/core/android/.../CoreManager.kt`	Replace delegation methods
`asg_client/.../handlers/StreamCommandHandler.java`	New file, replaces RtmpCommandHandler
`asg_client/.../handlers/RtmpCommandHandler.java`	Delete
`asg_client/.../services/RtmpStreamingService.java`	Rename mRtmpUrl → mStreamUrl
`asg_client/.../services/WebRtcStreamingService.java`	Wire into StreamCommandHandler
`asg_client/.../utils/ServiceConstants.java`	Replace RTMP constants

Verification

SDK build: cd cloud && bun run build — verify no type errors
Cloud lint: cd cloud/packages/cloud && bun run lint
Cloud tests: cd cloud && bun run test
asg_client build: cd asg_client && ./gradlew assembleDebug
Grep for remnants: rg "rtmpUrl|startRtmpStream|START_RTMP|RTMP_STREAM|ManagedStream|UnmanagedStream" --type ts --type java --type kotlin — should return zero hits outside of comments/docs
Manual: Test each protocol with asg_client test scripts

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aisraelov · 2026-03-29T00:40:52Z

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chatgpt-codex-connector · 2026-03-29T00:47:29Z

cloud/packages/cloud/src/services/streaming/ManagedStreamingExtension.ts

+      type: CloudToGlassesMessageType.START_STREAM,
      sessionId: userSession.sessionId,
-      rtmpUrl: liveInput.rtmpUrl, // Cloudflare ingest URL
+      streamUrl: liveInput.srtUrl!, // Cloudflare ingest URL


Guard against missing SRT ingest URL before start

startManagedStream always sends streamUrl: liveInput.srtUrl!, but CloudflareStreamService.createLiveInput() makes srtUrl optional and can return undefined when Cloudflare does not provide SRT fields. In that case we send an invalid START_STREAM payload to glasses, so managed streaming fails to start even though an ingest URL (rtmpUrl) is available. Add an explicit null check with fallback/error handling before building the start message.

Useful? React with 👍 / 👎.

chatgpt-codex-connector · 2026-03-29T00:47:29Z

asg_client/app/src/main/java/com/mentra/asg_client/service/media/managers/MediaManager.java

+                    String streamId = getActiveStreamId();
+                    if (streamId != null && !streamId.isEmpty()) status.put("streamId", streamId);


Use source service stream IDs in status callbacks

The shared callback tags status events with getActiveStreamId(), which picks whichever service currently has an ID (RTMP, then SRT, then WHIP). During protocol switches, a delayed stop/error callback from the previous service can therefore be emitted with the new stream's ID, causing cloud-side state to mark the wrong stream as stopped/errored. Stream IDs should come from the emitting service (or per-service callbacks), not global active-service lookup.

Useful? React with 👍 / 👎.

chatgpt-codex-connector · 2026-03-29T00:47:29Z

...ient/app/src/main/java/com/mentra/asg_client/service/core/handlers/StreamCommandHandler.java

+            } else if (WhipStreamingService.isStreaming()) {
+                WhipStreamingService.stopStreaming(context);


Stop WHIP sessions even when reconnecting

handleStopCommand only stops WHIP when WhipStreamingService.isStreaming() is true, but that method returns false during reconnecting/starting states. If a user sends stop_stream while WHIP is reconnecting, the handler reports not_streaming and leaves the reconnect loop running, so the stream can come back unexpectedly. Include reconnecting/starting states (or stop unconditionally) for WHIP stop handling.

Useful? React with 👍 / 👎.

yagarwal1307 added 5 commits March 13, 2026 15:34

chore: write a doc

e014f7a

Merge branch 'dev' into ya/replace-rtmp-with-srt

570ab91

chore: add srt implementation

c6d59ae

Merge branch 'dev' into ya/replace-rtmp-with-srt

91e7a0a

fix: fix the audio streaming issue

8e23923

yagarwal1307 requested a review from a team as a code owner March 19, 2026 14:33

yagarwal1307 changed the base branch from main to dev March 19, 2026 14:33

chore: add basic working implementation for webrtc

5f7e0ed

Merge branch 'dev' into ya/replace-rtmp-with-srt

3dbd49e

aisraelov mentioned this pull request Mar 23, 2026

Mentra Live: Overhaul livestreaming system #1974

Open

aisraelov and others added 2 commits March 25, 2026 22:07

Merge remote-tracking branch 'origin/dev' into ya/replace-rtmp-with-srt

be1891e

chore: refactor the codebase and generalise streaming

0bf1250

yagarwal1307 force-pushed the ya/replace-rtmp-with-srt branch from 71afd75 to 0bf1250 Compare March 27, 2026 07:21

yagarwal1307 added 3 commits March 27, 2026 17:39

fix: disable stun and turn to save time

5142d88

chore: delete extra file

9dd0902

chore: use srt url for managed stream

4175eb8

Merge branch 'dev' into ya/replace-rtmp-with-srt

c4f179f

chatgpt-codex-connector bot reviewed Mar 29, 2026

View reviewed changes

aisraelov added 3 commits March 28, 2026 21:20

update docs

7b87db1

minor fixes

2b5a19b

dev ex improvements

d7f4ed9

		String streamId = getActiveStreamId();
		if (streamId != null && !streamId.isEmpty()) status.put("streamId", streamId);

		} else if (WhipStreamingService.isStreaming()) {
		WhipStreamingService.stopStreaming(context);

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Streaming Protocol Refactor: Unified Multi-Protocol Support

Context

SDK Public API

Design Principle

Architecture

Livestream (Managed) Flow

Changes by Layer

Layer 1: SDK Types (cloud/packages/sdk/src/types/)

Layer 2: SDK Camera Module (cloud/packages/sdk/src/app/session/modules/camera.ts)

Layer 3: Cloud Extensions

Layer 4: Mobile Bridge

Layer 5: asg_client (Glasses)

Layer 6: Status Reporting

Files to Modify

Verification

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yagarwal1307 commented Mar 19, 2026 •

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Layer 1: SDK Types (`cloud/packages/sdk/src/types/`)

Layer 2: SDK Camera Module (`cloud/packages/sdk/src/app/session/modules/camera.ts`)

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edited

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cloudflare-workers-and-pages bot commented Mar 23, 2026 •

edited

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cloudflare-workers-and-pages bot commented Mar 23, 2026 •

edited

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