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Adding webrtc-aside.
This transport is yet an other webrtc transport but instead of current it doesn't use any meeting server or any protocol. Instead it use a libp2p stream so it can use circuit or ws for SDP trickle exchange. The only drawback is it require routing. Then its DTLS for security and SCTP for muxing.
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protocols.csv

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@@ -28,5 +28,6 @@ code, size, name, comment
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277, 0, p2p-stardust,
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275, 0, p2p-webrtc-star,
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276, 0, p2p-webrtc-direct,
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278, 0, p2p-webrtc-aside,
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290, 0, p2p-circuit,
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777, V, memory, in memory transport for self-dialing and testing; arbitrary

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