feat(webrtc): WebRTC-Direct v2 flow (libp2p/specs#715) — listener accepts v2, dialer opt-in - #1459
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feat(webrtc): WebRTC-Direct v2 flow (libp2p/specs#715) — listener accepts v2, dialer opt-in#1459yashksaini-coder wants to merge 8 commits into
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…rity by DTLS role go-libp2p (go-msgio pbio fallback) writes the uvarint prefix and the protobuf body as two SCTP messages, so every py<->go WebRTC-Direct connection died at Noise msg#1 with "malformed handshake frame length". New stream._decode_frames pops complete frames off an accumulating buffer (frames may be split or batched across SCTP messages, bounded by MAX_MESSAGE_SIZE); DataChannelReadWriter and WebRTCStream.on_data both use it. A malformed frame now resets the stream instead of dropping one message and desynchronising. With the handshake passing, a py listener then hit aiortc's `assert stream_id not in self._data_channels`: aiortc picks DCEP stream id parity from the ICE role, which is inverted for a WebRTC-Direct listener (ICE-controlled, DTLS server), so it used even ids like go's dialer. _create_channel now passes an explicit id by DTLS role (RFC 8832: client even, server odd). Verified live against go-libp2p v0.49.0: go->py and py->go, v1 and v2. Refs libp2p#1437
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- keep aiortc allocating (and recycling) SCTP stream ids; only re-seed the parity from the DTLS role (explicit ids tied the 16-bit space to our never-recycled counter) - on_data: deliver frames decoded before a malformed one, reset the peer (RESET + cleanup) instead of a local-only reset, ignore bytes after reset, and apply a whole SCTP message in one trio hop - _decode_frames: derive the max prefix length from MAX_MESSAGE_SIZE Refs libp2p#1437
The test performs two full dials and two bounded peer-connection closes; on Windows each close can take ~5 s when a datagram write is in flight, so 30 s was flaky (seen once on CI, no socket errors logged). Refs libp2p#1437
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… collides Harness mode binds TCP on the same port number as the OS-chosen UDP mux port; Windows reserves port ranges per protocol, so that TCP bind can fail with WinError 10013 (seen on CI in the harness loopback test). Retry up to five times with a new UDP port before giving up; explicit ports still fail loudly. Refs libp2p#1437
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…epts v2, dialer opt-in Listener: stop dropping libp2p+webrtc+v2/ first contacts. The server ufrag minus the prefix is the dialer's ICE pwd (must be a valid one, 22..256 ice-chars, else rejected in parse_direct_username); the inferred offer uses client_ufrag + that pwd, and our local ufrag/pwd stay server_ufrag verbatim as in v1. _accept_v1 -> _accept(client_pwd=...). Dialer: new WebRTCTransportConfig.webrtc_direct_dial_version (default 1). v2 keeps the aioice ufrag/pwd (no munging) and sets the synthetic answer's ufrag == pwd to "libp2p+webrtc+v2/" + local_password. Default stays v1 while specs#715 is unmerged; unknown values raise WebRTCConnectionError. Tests: v2 parse/reject cases, SDP shape, v2 loopback echo, no-munge check, short-suffix rejection, mixed v1+v2 dialers on one listener. Refs libp2p#1437
- validate webrtc_direct_dial_version in the config (was only checked mid-dial after the PC was built) - parse_direct_username returns the validated client pwd; make_v2_credential; one pwd check instead of two version branches - inline the v1 credential writes; trim the dial() comment Refs libp2p#1437
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What
The v2 flow from libp2p/specs#715, on top of the v1 listener that landed in #1449.
libp2p+webrtc+v2/first contacts alongside v1: recoversclient_pwdby stripping the prefix (rejected unless a valid ice-pwd, 22–256), infers the offer withice-ufrag=client_ufrag/ice-pwd=client_pwd, and sets its own ufrag and pwd toserver_ufragverbatim. Unknown/missing prefix still rejected; v1 unchanged (pwd = client half).WebRTCTransportConfig(webrtc_direct_dial_version=2): no munging — it reads its aioice-generated pwd and encodes it in the synthetic answer's ufrag/pwd aslibp2p+webrtc+v2/<client_pwd>. Default stays v1 until specs#715 merges.Verified
Stacked on #1460. Refs #1437 (v2 item); go/js interop tests come next.