Bolt.Net.Media 1.0.0-dev.213

Bolt Media Streaming — Voice/Video Calls over Bolt Protocol

Overview

Bolt Media is an experimental extension of the Bolt binary RPC protocol for audio/video call research. The repository contains protocol and processing primitives, but the audited end-to-end media path is not production-ready and is not a replacement for WebRTC.

For standard XFramework module RPC, prefer the generated [BoltHandler] plus IBoltRequest<TRequest, TResponse> pattern documented in BOLT.md. Bolt Media is the specialized media-streaming layer, not the default pattern for CRUD or feature-command handlers.

Current status: quarantined pending remediation. The audit identified broken browser stream wiring, unbounded peer-controlled work, incomplete FEC/NACK behavior, unauthenticated key exchange, and lifecycle leaks. Existing unit tests do not establish a secure or operational end-to-end media path.

Deployment Containment

Bolt Hub enforces BoltConfiguration:MediaEnabled, which defaults to false and is explicitly disabled in every XFramework Hub environment and Compose deployment. While quarantined, deployments must not override it, instantiate BoltMediaClient or BoltMediaService, expose media UI, route production media clients to the Hub, or advertise Bolt Media capability. Recognition of media frame types by the protocol does not constitute production enablement.

QUIC/WebTransport and direct P2P are not wired as supported end-to-end transports. They must remain absent from negotiated capabilities and production documentation until secure browser and server integration tests pass.

Architecture

Caller                    Bolt Hub (SFU)                    Callee
  |                           |                               |
  |-- CallSignal(Initiate) -->|-- CallSignal(Initiate) ------>|
  |<-- CallSignal(Ring) ------|                               |
  |                           |<-- CallSignal(Answer) --------|
  |<-- CallSignal(Answer) ----|                               |
  |                           |                               |
  |-- MediaConfig (audio) --->|-- MediaConfig (audio) ------->|
  |-- MediaConfig (video) --->|-- MediaConfig (video) ------->|
  |                           |                               |
  |== MediaFrame (audio) ====>|== MediaFrame (audio) ========>|
  |== MediaFrame (video) ====>|== MediaFrame (video) ========>|
  |<== MediaFrame (audio) ====|<== MediaFrame (audio) ========|
  |<== MediaFrame (video) ====|<== MediaFrame (video) ========|
  |                           |                               |
  |                      Media Tap                            |
  |                    (non-blocking copy)                    |
  |                           |                               |
  |                    IMediaProcessor                        |
  |                  (recording, transcription, AI)           |

Group Calls (SFU Mode)

  Participant A ──MediaFrame──> Bolt Hub ──MediaFrame──> Participant B
  Participant B ──MediaFrame──> Bolt Hub ──MediaFrame──> Participant A
  Participant C ──MediaFrame──> Bolt Hub ──MediaFrame──> Participant A
                                         ──MediaFrame──> Participant B

The intended SFU path forwards encoded payloads without codec decoding. The current implementation still performs copies and has not passed the required bounded-memory fanout tests.

Wire Protocol

Frame Types

MediaFrame Header (30 bytes)

[1:type=0x21] [16:streamId] [4:sequenceNumber] [4:timestamp] [1:flags] [4:payloadLen] [payload]

• sequenceNumber — monotonic per-stream, for ordering + gap detection
• timestamp — RTP-style media clock (48kHz for audio, 90kHz for video)
• flags — bit 0: keyframe, bit 1: end-of-picture, bit 2: marker, bit 3: FEC-protected, bit 4-5: priority, bit 6: drop-eligible, bit 7: compressed

CallSignal Types

Call State Machine

Initiating --> Ringing --> Active --> Ended
                       \-> Rejected   /\ (from Active or Held)
                       \-> Missed     Active <-> Held

• Initiating → Ringing: callee is online and receives the signal
• Ringing → Active: callee answers
• Ringing → Rejected/Missed: callee rejects or 30-second timeout
• Active ↔ Held: either party holds/unholds
• Active/Held → Ended: either party ends

Experimental Components

The sections below describe implementation primitives, not production-ready capabilities. Their behavior remains subject to the deployment containment above.

Adaptive Bitrate

The receiver sends MediaFeedback every 250ms with quality metrics:

Bitrate floor: audio 16kbps, video 100kbps. Ceiling: originally negotiated bitrate.

Forward Error Correction (FEC)

XOR-based parity. For every K source frames, one parity frame is generated.

Enabled by default on TCP (WebSocket). Dynamic: enable when loss > 3%, disable when loss < 0.5%.

Dynamic Throughput Maintenance

Multi-layer strategy to maintain target throughput under degrading networks:

Server-Side Media Hooks

public interface IMediaProcessor
{
    bool Accepts(Guid callId, MediaType mediaType);
    ValueTask ProcessFrameAsync(Guid callId, Guid streamId,
        ReadOnlyMemory<byte> frameData, uint timestamp, uint sequenceNumber);
    ValueTask OnCallStartedAsync(Guid callId);
    ValueTask OnCallEndedAsync(Guid callId);
}

// Registration
services.AddBoltServer(options =>
{
    options.MediaProcessors.Add(new RecordingProcessor());
    options.MediaProcessors.Add(new TranscriptionProcessor());
});

Codec Support

Audio

Video

Usage

The following snippets are design sketches and are not a supported production quick start. The audited high-level browser path does not currently complete encode-to-remote-decode media flow. Use these APIs only in isolated remediation tests until the quarantine is lifted.

.NET Client

var client = new BoltClient(serverUri, "my-service", "My App", options, logger);
await client.ConnectAsync(ct);

// Start a call
var callId = await client.StartCallAsync("other-service", video: true);

// Handle incoming calls
client.OnIncomingCall += async (info) =>
{
    await client.AnswerCallAsync(info.CallId);
};

// Send media frames
var stream = client.GetMediaStream(audioStreamId);
await stream.SendFrameAsync(opusEncodedAudio, isKeyframe: false);

// Receive media frames
await foreach (var frame in stream.ReadFramesAsync(ct))
{
    // frame.Data contains encoded audio/video
    // frame.IsKeyframe, frame.SequenceNumber, frame.Timestamp
}

// End call
await client.EndCallAsync(callId);

Browser Client (TypeScript)

import { BoltBrowserClient, AudioCodecHelper, VideoCodecHelper } from '@xframework/bolt-browser';

const client = new BoltBrowserClient('ws://bolt-hub:7000/bolt', 'browser-1', 'Browser');
await client.connect();

// Handle incoming calls
client.onIncomingCall = (callId, callerClientId) => {
    client.answerCall(callId);
};

// Start a call with media
const callId = client.startCall('other-user');
const audioStream = client.sendMediaConfig(crypto.randomUUID(), callId, true, 64);

// Encode audio via WebCodecs
const audio = new AudioCodecHelper();
await audio.initEncoder(48000, 1, 64000);
audio.onEncodedChunk = (data) => audioStream.sendFrame(data);

// Decode received audio
const decoder = new AudioCodecHelper();
await decoder.initDecoder();
audioStream.onFrame = (event) => decoder.decode(event.data, event.timestamp);

Comparison vs WebRTC

Intended architectural properties

Where WebRTC is ahead

Overall Assessment

No defensible feature-parity percentage is currently available. WebRTC provides mature mandatory encryption, congestion control, NAT traversal, interoperability, and browser validation that Bolt Media has not demonstrated. Bolt Media remains experimental until its security, correctness, browser, loss/reordering, and soak gates pass.

Implementation Phases

Remaining Work

• Keep the server-enforced, disabled-by-default media gate closed until all release gates pass.
• Repair the bounded NACK/FEC/sequence pipeline and deterministic resource cleanup.
• Implement authenticated, transcript-bound, fail-closed per-call and group encryption.
• Repair browser stream registration, answer-side setup, timestamps, codec metadata, and playback.
• Validate congestion control, jitter, probing, simulcast, and hold behavior end to end.
• Either implement secure negotiated QUIC/WebTransport and P2P transports or keep them unadvertised.
• Pass real-browser, multi-peer, adversarial, loss/reordering, and long-running soak gates before release.