rtp

A userspace reliable transmission 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/pkt_send_space.rs]]
  • recv window: [[src/pkt_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 transmission layer
  • gluing the unreliable layer and the reliable layer together
  • main entry: [[transmission_layer.rs]]
• a user-facing I/O-agnostic socket wrapper
  • managing opening, closing, timer, async read/write for the transmission 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]].