Struct workflow_rpc::client::WebSocketConfig

pub struct WebSocketConfig {
    pub write_buffer_size: usize,
    pub max_write_buffer_size: usize,
    pub max_message_size: Option<usize>,
    pub max_frame_size: Option<usize>,
    pub accept_unmasked_frames: bool,
    pub receiver_channel_cap: Option<usize>,
    pub sender_channel_cap: Option<usize>,
    pub handshake: Option<Arc<dyn Handshake>>,
    pub resolver: Option<Arc<dyn Resolver>>,
}

Configuration struct for WebSocket client (native Tungstenite and NodeJs connections only)

Fields

write_buffer_size: usize

The target minimum size of the write buffer to reach before writing the data to the underlying stream. The default value is 128 KiB.

If set to 0 each message will be eagerly written to the underlying stream. It is often more optimal to allow them to buffer a little, hence the default value.

Note: flush will always fully write the buffer regardless.

max_write_buffer_size: usize

The max size of the write buffer in bytes. Setting this can provide backpressure in the case the write buffer is filling up due to write errors. The default value is unlimited.

Note: The write buffer only builds up past write_buffer_size when writes to the underlying stream are failing. So the write buffer can not fill up if you are not observing write errors even if not flushing.

Note: Should always be at least write_buffer_size + 1 message and probably a little more depending on error handling strategy.

max_message_size: Option<usize>

The maximum size of a message. None means no size limit. The default value is 64 MiB which should be reasonably big for all normal use-cases but small enough to prevent memory eating by a malicious user.

max_frame_size: Option<usize>

The maximum size of a single message frame. None means no size limit. The limit is for frame payload NOT including the frame header. The default value is 16 MiB which should be reasonably big for all normal use-cases but small enough to prevent memory eating by a malicious user.

accept_unmasked_frames: bool

When set to true, the server will accept and handle unmasked frames from the client. According to the RFC 6455, the server must close the connection to the client in such cases, however it seems like there are some popular libraries that are sending unmasked frames, ignoring the RFC. By default this option is set to false, i.e. according to RFC 6455.

receiver_channel_cap: Option<usize>

The capacity of the channel used to queue incoming messages from WebSocket.

sender_channel_cap: Option<usize>

The capacity of the channel used to queue outgoing messages to WebSocket.

handshake: Option<Arc<dyn Handshake>>

Handshake handler for WebSocket connections. If supplied, it will be called when the connection is established. The handshake handler can be used to perform additional validation or setup before the connection is used.

resolver: Option<Arc<dyn Resolver>>

Resolver for WebSocket connections. If supplied, it will be called to resolve the URL before the connection is established. The resolver can be used as an alternative to supplying the URL and will be invoked each time the websocket needs to be connected or reconnected.

Trait Implementations

impl Clone for WebSocketConfig

fn clone(&self) -> WebSocketConfig

Returns a copy of the value.

fn clone_from(&mut self, source: &Self)

Performs copy-assignment from source.

impl Default for WebSocketConfig

fn default() -> WebSocketConfig

Returns the “default value” for a type.

impl From<WebSocketConfig> for WebSocketConfig

fn from(config: WebSocketConfig) -> WebSocketConfig

Converts to this type from the input type.