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rtp

A userspace reliable transport protocol whose congestion window is governed only by the delivery rate (i.e., bandwidth/throughput) and the data loss rate.

Features

  • an async-free reliable layer
    • a piece of pure algorithm
    • main entry: [[src/reliable_layer.rs]]
    • send window: [[src/packet_send_space.rs]]
    • recv window: [[src/packet_recv_space.rs]]
    • send rate limiter: [[src/token_bucket.rs]]
    • SACK managing: [[src/sack.rs]]
  • a dead simple codec for packet encoding/decoding
    • main entry: [[src/codec.rs]]
    • wireshark dissector: [[wireshark/rtp.dissector.lua]]
  • an async-based I/O-agnostic transport layer
    • gluing the unreliable layer and the reliable layer together
    • main entry: [[src/transport_layer.rs]]
  • a user-facing I/O-agnostic socket wrapper
    • managing opening, closing, timer, async read/write for the transport layer
    • perk: You are allowed to wait until or check if the send buf is empty!
    • main entry: [[src/socket.rs]]
  • a user-facing over-UDP implementation
    • exposing listening, accepting, and connecting APIs
    • main entry: [[src/udp.rs]]
  • a user-facing keyed-streams over-single-UDP-connection implementation
    • exposing listening, accepting, and connecting APIs
    • main entry: [[src/keyed_udp.rs]]

How to use

  • as a high-level socket user: Refer to [[examples]] directory and the test sections in [[src/udp.rs]] and [[src/keyed_udp.rs]].
  • as a low-level reliable layer user: Refer to the test sections in [[src/socket.rs]].

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A userspace reliable transport protocol; send rate governed only by delivery rate and data loss; similar to TCP performance.

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