This is an example implementation of something that might become WebTorrent DHT. It is based on BitTorrent DHT and is extended in a way that will work both in Node.js and modern browsers while keeping BitTorrent DHT interface.
npm install webtorrent-dhtThis project is WIP and not ready for production use.
Also K in WebTorrent DHT defaults to 2, which is much more reasonable for WebRTC realities.
Additional options specific to WebTorrent DHT are:
* extensions - Array of strings with extensions supported by current node
* simple_peer_opts - Object as in simple-peer constructor, used by webrtc-socket
* simple_peer_constructor - Custom implementation of simple-peer, in case it is necessary
* http_address - Object with keys address and port that corresponds to running HTTP server (specify this in case when publicly accessible address/port are different from those where HTTP server is listening on), used by webrtc-socket
There are simple demos in demo directory, you can run them in browser and on the server and then use dht key on window or global for DHT queries.
For debugging purposes run on Node as following:
``bash`
DEBUG=webtorrent-dht node demo/node-server.js
And set on client localStorage key as following:`javascript`
localStorage.debug = 'webtorrent-dht'
This will output a lot of useful debugging information.
inherits bittorrent-dht
* inherits k-rpc
* inherits k-rpc-socket
* (superset of the subset of dgram interface implemented from scratch)For high level protocol extension description in form of potential BEP take a look at bep.rts file in this repository.
$3
Uses k-rpc-webrtc instance instead of k-rpc in constructor.Patches its
toJSON() method of bittorrent-dht so that nodes property of returned object contains bootstrap nodes with HTTP servers only. This ensures that bootstrapping will be possible from those nodes in future (WebRTC-only nodes can't be used for this since they require 2-way signaling before connection can be established).Also
listen() method is now used to start HTTP server, but this will be detailed in webrtc-socket component.Other than mentioned changes
webtorrent-dht API is exactly the same as bittorrent-dht.This is the components exported by this package.
$3
Uses k-rpc-socket-webrtc instance instead of k-rpc-socket in constructor.Uses other default bootstrap nodes than
k-rpc.Forgets about disconnected nodes as soon as disconnection happens (since we know when this happens exactly).
$3
Uses socket-webrtc instance instead of dgram in constructor.Patches
send() method for logging in debug mode.Patches
query() method for find_node, get_peer and get queries by injecting signals argument with WebRTC offer signaling data before making query and establishes WebRTC connection when response to the query with WebRTC answer signaling data is received.
Also each peer in nodes and values keys from response is updated with IP address and port of the actual established WebRTC connection (since response will contain details of the queried node and those details will likely be slightly different).Patches
response() method for find_node, get_peer and get queries by re-sending WebRTC offer signaling data to the peers queried node is about to respond with (using peer_connection query) and injects signals key into the response with WebRTC answer signaling data and same number of entries and order as corresponding nodes or values key.Patches
emit method to capture firing query event for peer_connection query and handles it itself (while not canceling query event as such) by consuming WebRTC offer signaling data and responding with WebRTC answer signaling data.
Also associates nodes IDs with corresponding WebRTC peers connections.$3
Implements the subset of dgram interface (address, bind, close, emit, on and send methods) as used by k-rpc-socket (exactly what is used, nothing more) and superset on top of it with features used by k-rpc-socket-webrtc and webtorrent-dht.*
address method returns information about where HTTP server is listening, object with keys address and port
* method starts HTTP server on specified address and port (both should be specified explicitly)
* close method closes all WebRTC connections and stops HTTP server if it is running
* emit method emits an event with arguments
* on method adds handler for an event
* send method first checks if there is an established WebRTC connection to specified address and port, if not - assumes HTTP address and port were specified, so that it will establish HTTP connection, use it for establishing WebRTC connection, will close HTTP connection and create an alias to use WebRTC connection instead next timeThere are a few of public methods exposed by
webrtc-socket:
* get_id_mapping(id : string) - Allows to get simple-peer instance by node's ID if connection is already established
* add_tag(id : string, association : string) - Allows to tag connection to specified node ID so that connection will not be closed until at least one tag is left
* del_tag(id : string, association : string) - Remove tag from connection to specified node ID, if no tags left on connection - it will be closedThere are also a few events exposed by
webrtc-socket:
* node_connected with string argument id - Is fired when there is a new connection established
* node_disconnected with string argument id - Is fired when there is a connection was closed
* extensions_received with arguments peer_connection (simple-peer instance) and an array extensions - If fired when extensions are received from the other peer
* http_peer_connection_alias with arguments host, port and peer_connection` - Is fired when peer connection is aliased by HTTP host and port (from bootstrap node)When reading LiveScript code make sure to configure 1 tab to be 4 spaces (GitHub uses 8 by default), otherwise code might be hard to read.
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