webrtc_coordinator 0.2.2

A peer-to-peer transport for peer_coordinator using WebRTC with data streams.

webrtc_coordinator

webrtc_coordinator is a peer-to-peer transport for peer_coordinator. Coordination messages and data travel directly between devices over WebRTC data channels, and a hub is used only for discovery and connection setup.

The package is pure Dart and contains no WebRTC implementation. An application supplies an RtcPeerAdapter: webrtc_coordinator_flutter provides one based on flutter_webrtc, and package:webrtc_coordinator/testing.dart provides a fake adapter for tests that run without a device.

API documentation

Comparison with the WebSocket transport

The WebSocket transport of peer_coordinator relays every message through the hub, so each message takes two network hops. With this transport, peers exchange an offer and an answer through the hub and then communicate directly, in one hop.

A data channel can also be unreliable and unordered, which a relay cannot provide. A stream for which latency matters more than completeness can then avoid the cost of retransmissions.

Installation

dart pub add webrtc_coordinator

A Flutter application adds webrtc_coordinator_flutter as well.

Usage

import 'package:peer_coordinator/peer_coordinator.dart';
import 'package:webrtc_coordinator/transports/webrtc.dart';
import 'package:webrtc_coordinator_flutter/webrtc_coordinator_flutter.dart';

final config = CoordinationConfig(
  name: 'my_session',
  transportConfig: RtcTransportConfig(
    hubUri: Uri.parse('ws://hub.local:8080'),
    adapterFactory: flutterWebrtcAdapterFactory,
    // Data streams only; the coordination stream is always reliable and
    // ordered, because a lost election message splits the session.
    dataOrdered: false,
    dataMaxRetransmits: 1,
  ),
);

The hub is the one used by the WebSocket transport:

dart run peer_coordinator:hub --host 0.0.0.0 --port 8080

What passes through the hub

Traffic	Path
Endpoint registration, peer queries, election	Hub
WebRTC offers, answers and ICE candidates	Hub
Coordination messages	Direct
Data samples	Direct

test/transports/rtc_transport_test.dart verifies this: after a complete data exchange, the routing table of the hub is empty for every stream.

ICE and NAT

iceServers is empty by default, which gives host candidates only: peers on one local network connect directly and no third party is involved. STUN servers can be added for peers behind NAT.

A connection relayed by a TURN server passes through that server, in the same way as traffic through the hub. The single-hop property described above therefore holds for host and server-reflexive (STUN) connectivity, and does not hold for TURN.

Design

There is one RTCPeerConnection for each pair of peers, shared by all streams between them. It is keyed on the node uId, which is the only identifier that remains stable when the role of a node changes in an election.

Each stream has one data channel, opened with negotiated: true and an id derived from the stream name (rtcChannelIdFor), so that the two ends do not exchange channel metadata. Collisions of derived ids are checked when a channel is opened.

When two peers dial each other at the same time, the node with the lexicographically lower uId makes the offer.

Routing is local to each peer. RtcMesh.subscribersFor takes the place of the hub's RelayRouting and is filled by the open signal that a subscriber sends. A producer sends on exactly the channels whose far end has asked for them.

Samples use the WsSampleFrame format of the WebSocket transport, so the encoders and decodeChannels are shared and covered by the tests of that transport. The slot fields of the frame are written as zero and are not read, since a data channel identifies its sender.

The adapter

flutter_webrtc cannot run in a dart test process, so every call to it passes through lib/src/rtc/rtc_adapter.dart, and tests use FakeRtcPeerAdapter in its place. No flutter_webrtc type appears in a signature of that interface: session descriptions and ICE candidates cross it as JSON maps, and the six values of RTCPeerConnectionState cross as the three values of RtcLinkState. This makes the dial state machine, the resolution of simultaneous dials, the routing and the framing testable without a device.

Contracts for implementers

Contract	Reason
discoverOnce waits for its full timeout when nothing matches	The election concludes that no better candidate exists from an empty result after the timeout. An early return makes every node elect itself and splits the session
PeerHandle.take() transfers ownership exactly once, and addInlet calls it before using the handle	A discovery cycle must not free a resource that a stream is using
inbox is a broadcast stream	Several subscribers read it
Delivery by the fake adapter is asynchronous, as in InMemoryBus	A synchronous fake hides ordering errors and validates behaviour that no real transport has

Testing

dart analyze
dart test
dart run benchmark/rtc_latency_bench.dart 60 5

test/transports/participation_modes_test.dart runs the shared runParticipationScenarios suite of peer_coordinator, which the in-memory and WebSocket transports also run. A participation mode is a property of the coordination layer, so a scenario that passes there and fails here indicates a defect in this transport.

The WebRTC arm of the benchmark runs against the fake adapter. It measures the overhead of the library with the network removed and serves as a regression check. Real peer-to-peer latency is measured with two devices on one local network and the Flutter binding, for example with transport_timing.

Known limitations

Limitation	Detail
Tests without a device stop at the adapter	ICE restarts, packet loss and the difference between reliable and unreliable delivery are exercised only on a device
maxRetransmits: 0 cannot be expressed through flutter_webrtc	RTCDataChannelInit.toMap writes the field only when it is positive, so zero produces a reliable channel. The binding logs a warning; 1 gives nearly unreliable delivery
WsControl.peerGone is not used	The hub broadcasts it when a socket closes, which would signal a departure sooner than a failed ICE connection. RtcMesh.peerLost currently depends on the link state alone
Mobile networks and NAT	Multiple network interfaces, Wi-Fi client isolation and carrier NAT can make host candidates fail in ways that a test on a local network does not show