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quic: add initial RTT option to session options · nodejs/node@84d3100 · GitHub

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quic: add initial RTT option to session options
Signed-off-by: James M Snell <jasnell@gmail.com> Assisted-by: OpenCode:Opus 4.6 PR-URL: #63267 Backport-PR-URL: #64675 Reviewed-By: Matteo Collina <matteo.collina@gmail.com>
1 parent a2b6a81 commit 84d3100

6 files changed

Lines changed: 109 additions & 4 deletions

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‎doc/api/quic.md‎

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@@ -2723,6 +2723,23 @@ added: v23.8.0
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Specifies the maximum number of milliseconds a TLS handshake is permitted to take
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to complete before timing out.
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#### `sessionOptions.initialRtt`
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<!-- YAML
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added: REPLACEME
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-->
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* Type: {bigint|number}
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* **Default:** `0` (use ngtcp2 default of 333ms)
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Specifies the initial round-trip time estimate in milliseconds. This value is
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used for probe timeout (PTO) computation, initial pacing, and early loss
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detection before the first actual RTT sample is collected from the connection.
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The default of 333ms is appropriate for the general internet. For low-latency
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environments such as loopback or same-rack deployments, setting a value closer
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to the actual RTT (e.g., `1`) avoids unnecessarily conservative initial
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behavior.
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#### `sessionOptions.keepAlive`
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<!-- YAML

‎lib/internal/quic/quic.js‎

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@@ -404,6 +404,9 @@ const endpointRegistry = new SafeSet();
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* @property {ArrayBufferView} [token] An opaque address validation token
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* previously received from the server via `onnewtoken` (client only).
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* @property {bigint|number} [handshakeTimeout] The handshake timeout
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* @property {bigint|number} [initialRtt] The initial round-trip time estimate in milliseconds.
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* Used for PTO computation and initial pacing before the first RTT sample. Default uses
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* ngtcp2's built-in default of 333ms. Set lower for low-latency environments.
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* @property {bigint|number} [keepAlive] The keep-alive timeout in milliseconds. When set,
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* PING frames will be sent automatically to prevent idle timeout.
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* @property {bigint|number} [maxStreamWindow] The maximum stream window
@@ -4875,6 +4878,7 @@ function processSessionOptions(options, config = kEmptyObject) {
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maxPayloadSize,
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unacknowledgedPacketThreshold = 0,
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handshakeTimeout,
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initialRtt,
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keepAlive,
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maxStreamWindow,
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maxWindow,
@@ -4982,6 +4986,7 @@ function processSessionOptions(options, config = kEmptyObject) {
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maxPayloadSize,
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unacknowledgedPacketThreshold,
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handshakeTimeout,
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initialRtt,
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keepAlive,
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maxStreamWindow,
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maxWindow,

‎src/quic/bindingdata.h‎

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@@ -93,6 +93,7 @@ class SessionManager;
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V(groups, "groups") \
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V(handshake_timeout, "handshakeTimeout") \
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V(http3_alpn, &NGHTTP3_ALPN_H3[1]) \
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V(initial_rtt, "initialRtt") \
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V(keep_alive_timeout, "keepAlive") \
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V(initial_max_data, "initialMaxData") \
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V(initial_max_stream_data_bidi_local, "initialMaxStreamDataBidiLocal") \

‎src/quic/session.cc‎

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@@ -513,6 +513,12 @@ Session::Config::Config(Environment* env,
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options.handshake_timeout == UINT64_MAX
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? UINT64_MAX
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: options.handshake_timeout * NGTCP2_MILLISECONDS;
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// The initial_rtt option is in milliseconds; ngtcp2 expects nanoseconds.
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// A value of 0 leaves the ngtcp2 default (333ms) unchanged.
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if (options.initial_rtt > 0)
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settings.initial_rtt = options.initial_rtt * NGTCP2_MILLISECONDS;
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settings.max_stream_window = options.max_stream_window;
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settings.max_window = options.max_window;
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settings.ack_thresh = options.unacknowledged_packet_threshold;
@@ -604,10 +610,11 @@ Maybe<Session::Options> Session::Options::From(Environment* env,
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if (!SET(version) || !SET(min_version) || !SET(preferred_address_strategy) ||
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!SET(transport_params) || !SET(tls_options) || !SET(qlog) ||
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!SET(handshake_timeout) || !SET(keep_alive_timeout) ||
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!SET(max_stream_window) || !SET(max_window) || !SET(max_payload_size) ||
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!SET(unacknowledged_packet_threshold) || !SET(cc_algorithm) ||
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!SET(draining_period_multiplier) || !SET(max_datagram_send_attempts)) {
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!SET(handshake_timeout) || !SET(initial_rtt) ||
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!SET(keep_alive_timeout) || !SET(max_stream_window) || !SET(max_window) ||
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!SET(max_payload_size) || !SET(unacknowledged_packet_threshold) ||
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!SET(cc_algorithm) || !SET(draining_period_multiplier) ||
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!SET(max_datagram_send_attempts)) {
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return Nothing<Options>();
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}
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@@ -726,6 +733,12 @@ std::string Session::Options::ToString() const {
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res += prefix + "handshake timeout: " + std::to_string(handshake_timeout) +
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" nanoseconds";
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}
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if (initial_rtt > 0) {
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res += prefix + "initial rtt: " + std::to_string(initial_rtt) +
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" milliseconds";
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} else {
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res += prefix + "initial rtt: <default>";
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}
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res += prefix + "max stream window: " + std::to_string(max_stream_window);
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res += prefix + "max window: " + std::to_string(max_window);
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res += prefix + "max payload size: " + std::to_string(max_payload_size);

‎src/quic/session.h‎

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@@ -163,6 +163,15 @@ class Session final : public AsyncWrap, private SessionTicket::AppData::Source {
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static constexpr uint64_t DEFAULT_HANDSHAKE_TIMEOUT = 10'000;
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uint64_t handshake_timeout = DEFAULT_HANDSHAKE_TIMEOUT;
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// The initial round-trip time estimate in milliseconds. ngtcp2 uses this
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// for PTO computation, initial pacing, and early loss detection before
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// the first RTT sample is collected. The default of 0 uses ngtcp2's
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// built-in default of 333ms, which is appropriate for the general
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// internet. For low-latency environments (e.g., loopback or same-rack
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// deployments), setting a value closer to the actual RTT avoids
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// unnecessarily conservative initial behavior.
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uint64_t initial_rtt = 0;
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// The keep-alive timeout in milliseconds. When set to a non-zero value,
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// ngtcp2 will automatically send PING frames to keep the connection alive
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// before the idle timeout fires. Set to 0 to disable (default).

‎test/parallel/test-quic-session-initial-rtt.mjs‎

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// Flags: --experimental-quic --experimental-stream-iter --no-warnings
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// Test: initialRtt session option is accepted and the session functions
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// correctly with a custom initial RTT estimate.
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import { hasQuic, skip, mustCall } from '../common/index.mjs';
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import assert from 'node:assert';
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const { ok } = assert;
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if (!hasQuic) {
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skip('QUIC is not enabled');
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}
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const { listen, connect } = await import('../common/quic.mjs');
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const { bytes } = await import('stream/iter');
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const encoder = new TextEncoder();
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const payload = encoder.encode('hello rtt');
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const serverDone = Promise.withResolvers();
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// Use a low initialRtt (1ms) to simulate a low-latency environment.
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// The session should complete successfully and the smoothed RTT in
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// stats should converge to a value well below the default 333ms.
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const serverEndpoint = await listen(mustCall((serverSession) => {
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serverSession.onstream = mustCall(async (stream) => {
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const data = await bytes(stream);
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ok(data.byteLength > 0);
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stream.writer.endSync();
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await stream.closed;
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serverSession.close();
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serverDone.resolve();
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});
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}), {
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initialRtt: 1, // 1ms
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});
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const clientSession = await connect(serverEndpoint.address, {
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initialRtt: 1, // 1ms
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});
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await clientSession.opened;
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const stream = await clientSession.createBidirectionalStream({
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body: payload,
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});
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for await (const _ of stream) { /* drain */ } // eslint-disable-line no-unused-vars
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await stream.closed;
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await serverDone.promise;
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// After data exchange, the smoothed RTT should have converged to a
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// realistic value. On loopback it should be well under 10ms (10,000,000ns).
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// The stat is in nanoseconds.
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const smoothedRtt = clientSession.stats.smoothedRtt;
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ok(smoothedRtt > 0n, 'smoothedRtt should be non-zero after data exchange');
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ok(smoothedRtt < 10_000_000n,
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`smoothedRtt should be under 10ms on loopback, got ${smoothedRtt}ns`);
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await clientSession.close();
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await serverEndpoint.close();

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