webrtc: allow dialing without listening first

Description

Currently, the WebRTC implementation requires us to establish a listener before we can dial a connection. See https://github.com/libp2p/rust-libp2p/blob/14530af2611b0cb17179f83802d31c0f28222e91/transports/webrtc/src/tokio/transport.rs#L134.

Not only is this a general short-coming of the implementation, it also complicates things in our interop-tests. In particular, we need to do a Swarm::listen_on before we can perform a dial: https://github.com/zvolin/rust-libp2p/blob/3c833dd6e9d1c069826c889e090f3b11cd4ad355/interop-tests/src/lib.rs#L44-L51

This makes the interop-tests for dial-only transports such as webtransport-websys unnecessarily complicated. This is the reason why we need to feature-gate the listening on cfg(target_arch = "wasm32"). Ideally, we would just move the listening to the is_dialer = false branch but due to the above limitation of the WebRTC transport, that is not possible.

For QUIC, we solve this with a dedicated dialer state: https://github.com/libp2p/rust-libp2p/blob/14530af2611b0cb17179f83802d31c0f28222e91/transports/quic/src/transport.rs#L73-L74

Are you planning to do it yourself in a pull request?

No.

Start in transports/webrtc/src/tokio/transport.rs at the listener-dependent dialing path, then compare the dedicated dialer state in transports/quic/src/transport.rs. Review interop-tests/src/lib.rs to understand the current listen-before-dial workaround. Done means WebRTC can dial without establishing a listener first and the interop-test workaround is no longer required.