feat: Add WebSocket and WebRTC transports to relay server example #6030
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elenaf9
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May 22, 2025
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Thanks @r-zig. Some high-level comments, but please see #6029 (comment) for the general discussion.
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| RUST_LOG=info cargo run -- --port 4884 --secret-key-seed 0 --websocket-port 8080 --webrtc-port 8081 | |||
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I appreciate the effort, but I don't think we should add run scripts that hard-code the parameters. Most user want to change these parameters anyway, and having to think about them IMO also helps the general understanding.
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Sure. I will move them to a new repo that provide examples.
examples/relay-server/Cargo.toml
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| libp2p-webrtc = { workspace = true, features = ["tokio"] } | ||
| libp2p-noise = { workspace = true } | ||
| libp2p-tls = { workspace = true } |
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Why not use them through the libp2p meta crate?
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Removed the tls & noise , but webrtc is not exposed via the libp2p meta crate
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Description
This PR addresses issue #6029 by adding WebSocket and WebRTC transport support to the relay server example.
While full WebRTC interoperability is not yet supported, these changes lay the groundwork for ongoing efforts to enable and test connectivity between peers behind different private NATs and VPNs.
This work is particularly relevant for use cases involving interoperability with other libp2p implementations, such as js-libp2p-example-webrtc-private-to-private, and is a step toward seamless connectivity between browser-based JS peers, Rust processes, and both Rust and JS relays.
Related:
libp2p/js-libp2p-examples#226.
libp2p/js-libp2p-examples#227.
Draft: WebRTC Implementation for Browser to Browser #5978
Fixes: libp2p/rust-libp2p#6029.
Notes & open questions
Change checklist