fix(webrtc): wire the WebRTC-Direct inbound path; spec Noise roles; real loopback test - #1447
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tox commands_pre installed no extras, so aiortc was absent in CI and every test under tests/core/transport/webrtc was skipped on GitHub. aiortc 1.15 is pure Python and its native deps (av, pylibsrtp) ship manylinux/win wheels, so no system packages are needed. Refs libp2p#1437
…oles The listener's inbound completion was a stub: after ICE it logged and returned, so no inbound connection ever reached the handler. dial() was broken too - get_remote_fingerprint read attributes aiortc does not have. - listener: own a trio nursery (system task, like TCP), hop asyncio->trio without blocking the loop, run Noise XX as *initiator* (spec: server initiates, dialer responds), then hand the authenticated connection to the handler; bound in-flight unauthenticated inbounds. - transport.dial(): Noise responder; verify authenticated peer ID against /p2p/ after the handshake. - noise: role-ordered prologue (dialer fingerprint, then server); PatternXX.handshake_outbound(remote_peer=None) skips only the ID equality check; DataChannelReadWriter.read(n) honours n (the Noise packet reader asks for the 2-byte prefix alone). - helpers: get_remote_fingerprint reads pc.sctp.transport._ssl peer cert; noise send waits for data channel 0 to open. - tests: transport-level dial->listen->stream echo loopback, wrong /p2p/ rejection, e2e Noise role test, buffered read test, fingerprint test. Refs libp2p#1437
- Frame Noise handshake bytes as uvarint-prefixed webrtc.pb.Message stream frames on channel 0 (spec 'Multiplexing'; what go/js do) instead of raw bytes - required for interop; py<->py was symmetric so tests did not notice. - Listener: guard the handler call - the nursery lives in a trio system task, so an escaping exception aborted the whole trio run. - dial(): bound the Noise phase (responder's first step is a read) with handshake_timeout; close the PC on every failure path, not just the two explicit mismatches; do not pass config.ice_servers (previously no STUN servers were used; the default Google STUN added ~5s to offline dials). - Tests: framing round-trip/chunking/FIN/malformed, handler-exception regression. Refs libp2p#1437
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@seetadev @acul71 Rebased on latest main; CI fully green (incl. Windows). This PR makes WebRTC-Direct actually connect end-to-end: the inbound path was a stub and |
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What
First code step for #1437. Makes WebRTC-Direct actually connect end to end (it did not):
_complete_inboundlogged and returned; handler never called). Now: listener-owned trio nursery (system task, like TCP), non-blocking asyncio→trio hop, Noise XX as initiator (spec: server initiates, dialer responds), authenticatedWebRTCConnectionhanded to the handler. In-flight unauthenticated inbounds bounded bymax_in_flight_connections.dial()was broken:get_remote_fingerprintreadpc._dtlsTransport._remote_certificate, which aiortc doesn't have. Now reads the DTLS peer cert frompc.sctp.transport._ssl. Dialer is the Noise responder and verifies the authenticated ID against/p2p/.dialer_fp ‖ server_fp, per spec) instead of local/remote.PatternXX.handshake_outbound(remote_peer: ID | None)— skips only the equality check whenNone(server doesn't know the dialer's ID); signature verification unchanged. Existing callers unaffected.DataChannelReadWriter.read(n)now honoursn— the Noise packet reader asks for the 2-byte length prefix alone; returning a whole channel message broke framing.sendwaits for data channel 0 to open (initiator sends right after DTLS, before SCTP opens the channel →InvalidStateError).Tests
test_dial_listen_open_stream_echo: dial → ICE/DTLS → Noise → handler → stream echo, both sides see the other's peer ID. Plus wrong-/p2p/rejection, e2e Noise role test, buffered-read test, fingerprint test,remote_peer=Nonepattern test.tests/core/transport/webrtc+tests/core/security/noise: 319 passed locally; loopback run 5× no flake. mypy/pyrefly clean.Stacked on #1446 (CI needs aiortc installed for these to run); will rebase once that merges.
Refs #1437