Files
peerspeak/tests/transport_loopback.rs
T
molluskandClaude Fable 5 554b613466
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chat: attachment cache, download, and transfer hardening (Phase 3)
Phase 3 of docs/chat-hardening-plan.md — attachments can no longer turn
into unbounded memory, bandwidth, decoder, or task pressure (S15 closed;
S14's filename half closed).

Cache and image cost (3A): AttachmentCache now carries encoded- and
decoded-byte budgets (96 MiB / 64 MiB) on top of the count cap, with
per-entry weights, replacement accounting, and oldest-first eviction; an
individually over-budget fetch services any pending Save/Play from the
bytes in hand and is exposed as Evicted instead of retained.
validate_image_bytes prechecks header dimensions (per-side AND a new
14 MP total-pixel limit) before any decode; the renderer only ever
receives a ≤1600 px downscaled RGBA preview whose w*h*4 cost counts
against the decoded budget — originals stay encoded-only for Save.
sanitize_filename strips the bidi/zero-width spoofing set (RTL-override
extension spoof).

Download policy and state (3B): images auto-fetch only when roster-
authored AND declared ≤4 MiB, gated by a new deterministic
AutoFetchBudget (per-author and session request+byte token buckets,
check-then-take, bounded author map) alongside the existing dedup and
four-permit bound. Attachment state is now explicit — absence/Loading/
Ready/Failed/Evicted — driven by a new AttachmentFetchStarted event, so
skipped or evicted images render a "Load image" button instead of an
indefinite "loading…", and repeated clicks can never spawn duplicate
fetch tasks.

Exact transfers and serve store (3C): fetch_blob requires the received
length to equal the declared size (short = local error, overlong =
bounded-read reject, empty keeps meaning "sender no longer has it");
the file picker's unbounded read is replaced by a metadata-prechecked
cap+1 bounded reader; one Arc<Vec<u8>> now backs the UI cache, command
queue, and serve store; served_files is a count- and byte-budgeted FIFO
ServeStore (16 entries / 128 MiB).

37 new tests (568 lib total) including a real two-endpoint loopback
exercising exact/short/overlong/unknown-id transfers. Plan checkboxes
ticked and constant deviations decision-logged. Tests-green-only: the
plan's two-machine field-test section remains open.

Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
2026-07-18 02:09:30 -04:00

664 lines
24 KiB
Rust

//! End-to-end loopback test for the redesigned audio transport.
//!
//! Spins up two real iroh endpoints on localhost (relay disabled, addresses
//! exchanged directly) and drives the actual production path: reactive
//! `connect_peer`, sequenced `broadcast`, `receive_datagrams`, and the
//! per-peer `JitterBuffer` decode. No microphone, speakers, or GUI required.
use std::sync::Arc;
use std::time::Duration;
use tokio::sync::mpsc;
use bytes::Bytes;
use iroh::address_lookup::memory::MemoryLookup;
use iroh::endpoint::presets;
use iroh::protocol::Router;
use iroh::{Endpoint, RelayMode};
use opus::{Application, Channels};
use iroh::endpoint::Connection;
use iroh::protocol::{AcceptError, ProtocolHandler};
use peerspeak::codec::AudioEncoder;
use peerspeak::codec::opus_impl::OpusEncoder;
use peerspeak::core::jitter::{FRAME_SAMPLES, JitterBuffer};
use peerspeak::network::iroh_impl::{AudioRouter, IrohTransport};
use peerspeak::network::{ConnEvent, NetworkTransport};
use peerspeak::protocol::AUDIO_ALPN;
struct Node {
endpoint: Endpoint,
transport: Arc<IrohTransport>,
_router: Router,
lookup: MemoryLookup,
}
async fn spawn_node() -> Node {
spawn_node_with_key(iroh::SecretKey::generate()).await
}
/// A minimal protocol handler for the reconnect test's "peer" side: it hands
/// every accepted audio connection out through a channel so the test can hold
/// it, read datagrams from it, and explicitly `close()` it to drive a drop —
/// all on a single, stable endpoint address that the dialer can re-reach.
#[derive(Clone)]
struct CaptureProtocol {
conns_tx: mpsc::UnboundedSender<Connection>,
}
impl std::fmt::Debug for CaptureProtocol {
fn fmt(&self, f: &mut std::fmt::Formatter<'_>) -> std::fmt::Result {
f.debug_struct("CaptureProtocol").finish_non_exhaustive()
}
}
impl ProtocolHandler for CaptureProtocol {
fn accept(
&self,
connection: Connection,
) -> impl std::future::Future<Output = Result<(), AcceptError>> + Send {
let conns_tx = self.conns_tx.clone();
async move {
// The channel retains the connection, which keeps it open after this
// future resolves (iroh closes accepted connections with no retained
// handle).
let _ = conns_tx.send(connection);
Ok(())
}
}
}
struct CapturePeer {
endpoint: Endpoint,
_router: Router,
lookup: MemoryLookup,
conns_rx: mpsc::UnboundedReceiver<Connection>,
}
async fn spawn_capture_peer(secret: iroh::SecretKey) -> CapturePeer {
let lookup = MemoryLookup::new();
let endpoint = Endpoint::builder(presets::Minimal)
.secret_key(secret)
.relay_mode(RelayMode::Disabled)
.address_lookup(lookup.clone())
.bind()
.await
.expect("bind capture peer");
let (conns_tx, conns_rx) = mpsc::unbounded_channel();
let router = Router::builder(endpoint.clone())
.accept(AUDIO_ALPN, CaptureProtocol { conns_tx })
.spawn();
CapturePeer {
endpoint,
_router: router,
lookup,
conns_rx,
}
}
/// Spawn a node with a specific secret key. Reusing a key gives the respawned
/// node the same `EndpointId`, which is how we simulate a peer dropping off the
/// network and coming back (new address, same identity) for the reconnect test.
async fn spawn_node_with_key(secret: iroh::SecretKey) -> Node {
let lookup = MemoryLookup::new();
let endpoint = Endpoint::builder(presets::Minimal)
.secret_key(secret)
// Direct-only: two endpoints on the same host reach each other via the
// addresses we seed into each other's lookup, with no external relay.
.relay_mode(RelayMode::Disabled)
.address_lookup(lookup.clone())
.bind()
.await
.expect("bind endpoint");
let transport = Arc::new(IrohTransport::new(endpoint.clone()));
// Mirror production: a persistent AudioRouter bound to this session's transport
// is what the router accepts inbound audio links on.
let audio_router = AudioRouter::new();
audio_router.bind(&transport);
let router = Router::builder(endpoint.clone())
.accept(AUDIO_ALPN, audio_router)
.spawn();
Node {
endpoint,
transport,
_router: router,
lookup,
}
}
/// One real, decodable Opus packet for a 20ms mono frame, prefixed with the
/// 4-byte little-endian sequence header the transport/jitter buffer expect.
fn packet(enc: &mut OpusEncoder, seq: u32) -> Bytes {
let pcm: Vec<i16> = (0..FRAME_SAMPLES)
.map(|i| if i % 2 == 0 { 2000 } else { -2000 })
.collect();
let encoded = enc.encode(&pcm).unwrap();
let mut buf = Vec::with_capacity(4 + encoded.len());
buf.extend_from_slice(&seq.to_le_bytes());
buf.extend_from_slice(&encoded);
Bytes::from(buf)
}
/// Drain link-state events until a `Connected` arrives, returning whether a
/// `Connecting` was seen first (the down→up transition). Panics on timeout so a
/// stuck reconnect fails loudly rather than hanging the suite.
async fn await_reconnect(
rx: &mut mpsc::Receiver<ConnEvent>,
deadline: tokio::time::Instant,
) -> bool {
let mut saw_connecting = false;
loop {
match tokio::time::timeout_at(deadline, rx.recv()).await {
Ok(Some(ConnEvent::Connecting(_))) => saw_connecting = true,
Ok(Some(ConnEvent::Connected(_))) => return saw_connecting,
Ok(Some(ConnEvent::Left(_))) => panic!("unexpected graceful Left during reconnect"),
Ok(None) => panic!("conn events channel closed unexpectedly"),
Err(_) => panic!("timed out waiting for Connected"),
}
}
}
#[tokio::test]
async fn loopback_sequenced_audio_reaches_peer_and_decodes() {
let a = spawn_node().await;
let b = spawn_node().await;
// Seed each side with the other's full address so direct dialing works.
a.lookup.add_endpoint_info(b.endpoint.addr());
b.lookup.add_endpoint_info(a.endpoint.addr());
let a_id = a.endpoint.id();
let b_id = b.endpoint.id();
a.transport.admit_audio_sender(b_id);
b.transport.admit_audio_sender(a_id);
// Subscribe to incoming datagrams on B before any are sent.
let mut b_rx = b.transport.receive_datagrams().await.expect("subscribe B");
// Reactive connection setup, exactly as core does on peer-join. Calling on
// both sides is fine: the lower EndpointId dials, the higher accepts, and a
// single shared connection forms.
a.transport.connect_peer(b.endpoint.addr()).await;
b.transport.connect_peer(a.endpoint.addr()).await;
// Let the dial + accept registration settle.
tokio::time::sleep(Duration::from_millis(500)).await;
// A sends 50 sequenced frames.
const N: u32 = 50;
let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
for seq in 0..N {
a.transport.broadcast(packet(&mut enc, seq));
tokio::time::sleep(Duration::from_millis(5)).await;
}
// Collect what B receives and feed it through a real jitter buffer,
// popping a frame per arrival to mirror the mixer's steady 20ms cadence
// (so the buffer stays shallow rather than overflowing its cap).
let mut jitter = JitterBuffer::new().unwrap();
let mut received = 0u32;
let mut decoded_frames = 0u32;
let deadline = tokio::time::Instant::now() + Duration::from_secs(2);
// Loop ends when the channel closes or the deadline is hit (pattern stops matching).
while let Ok(Some((from, bytes))) = tokio::time::timeout_at(deadline, b_rx.recv()).await {
assert_eq!(from, a_id, "datagram should be attributed to sender A");
assert!(bytes.len() >= 4, "packet carries a sequence header");
let seq = u32::from_le_bytes([bytes[0], bytes[1], bytes[2], bytes[3]]);
jitter.insert(seq, bytes[4..].to_vec());
received += 1;
if let Some(frame) = jitter.pop_frame() {
assert_eq!(
frame.len(),
FRAME_SAMPLES,
"decoded frame is one 20ms frame"
);
decoded_frames += 1;
}
if received >= N {
break;
}
}
// Drain whatever remains buffered behind the playout delay.
while let Some(frame) = jitter.pop_frame() {
assert_eq!(frame.len(), FRAME_SAMPLES);
decoded_frames += 1;
}
// On localhost essentially nothing should be lost over a real QUIC datagram path.
assert!(
received >= N - 2,
"expected to receive ~{N} datagrams, got {received}"
);
// And nearly all received packets should decode to PCM (a few absorbed by
// the initial priming delay).
assert!(
decoded_frames >= N - 5,
"expected to decode ~{N} frames, got {decoded_frames} (received {received})"
);
}
/// Connection transparency: over a real loopback link, `connection_stats()`
/// must report the peer's selected path as direct (relay disabled here), with
/// an IP remote address and counters that advance while audio flows — and the
/// `connstats::derive` seam must turn two such snapshots into badge info with
/// live rates.
#[tokio::test]
async fn connection_stats_report_a_direct_path_with_live_counters() {
let a = spawn_node().await;
let b = spawn_node().await;
a.lookup.add_endpoint_info(b.endpoint.addr());
b.lookup.add_endpoint_info(a.endpoint.addr());
let a_id = a.endpoint.id();
let b_id = b.endpoint.id();
a.transport.admit_audio_sender(b_id);
b.transport.admit_audio_sender(a_id);
// Keep B's receive path subscribed like production (drained implicitly).
let _b_rx = b.transport.receive_datagrams().await.expect("subscribe B");
a.transport.connect_peer(b.endpoint.addr()).await;
b.transport.connect_peer(a.endpoint.addr()).await;
tokio::time::sleep(Duration::from_millis(500)).await;
let snap = |stats: Vec<(iroh::EndpointId, peerspeak::network::PathSnapshot)>| {
stats
.into_iter()
.find(|(id, _)| *id == b_id)
.map(|(_, s)| s)
.expect("peer B should appear in A's connection stats")
};
let s1 = snap(a.transport.connection_stats());
assert!(!s1.is_relay, "loopback with relay disabled must be direct");
assert!(
s1.remote_addr.parse::<std::net::SocketAddr>().is_ok(),
"direct path address should be ip:port, got {}",
s1.remote_addr
);
// Stream real audio so the path counters move.
let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
for seq in 0..25u32 {
a.transport.broadcast(packet(&mut enc, seq));
tokio::time::sleep(Duration::from_millis(5)).await;
}
let s2 = snap(a.transport.connection_stats());
assert!(s2.tx_bytes > s1.tx_bytes, "sent bytes should advance");
assert!(
s2.tx_datagrams > s1.tx_datagrams,
"sent datagrams should advance"
);
// The derivation seam turns the two snapshots into live badge info.
let info = peerspeak::core::connstats::derive(Some(&s1), &s2, Duration::from_millis(200));
assert!(!info.relay);
assert_eq!(info.remote_addr, s2.remote_addr);
assert!(info.rtt_ms < 1000, "localhost RTT should be sane");
assert!(
info.up_kbps
.expect("same path + positive window has a rate")
> 0.0,
"audio was flowing, so the upstream rate must be non-zero"
);
}
/// Read datagrams off a raw connection until `target` arrive or the deadline
/// passes, asserting each carries the 4-byte sequence header.
async fn count_audio(conn: &Connection, target: u32, deadline: tokio::time::Instant) -> u32 {
let mut received = 0;
while let Ok(Ok(bytes)) = tokio::time::timeout_at(deadline, conn.read_datagram()).await {
assert!(bytes.len() >= 4, "packet carries a sequence header");
received += 1;
if received >= target {
break;
}
}
received
}
#[tokio::test]
async fn dialer_reconnects_after_link_drops() {
// The unit under test is the IrohTransport dialer. The peer is a hand-rolled
// endpoint we fully control, so we can explicitly close the live connection
// (which the dialer observes promptly, unlike a silent handle drop) and then
// accept the dialer's re-dial — all on one stable address it can re-reach.
//
// The dialer must hold the lower id so the transport's deterministic rule
// makes it dial, so order the two identities up front.
let s1 = iroh::SecretKey::generate();
let s2 = iroh::SecretKey::generate();
let (dialer_secret, peer_secret) = if s1.public().to_string() < s2.public().to_string() {
(s1, s2)
} else {
(s2, s1)
};
let dialer = spawn_node_with_key(dialer_secret).await;
let mut peer = spawn_capture_peer(peer_secret).await;
let dialer_id = dialer.endpoint.id();
let peer_id = peer.endpoint.id();
assert!(dialer_id.to_string() < peer_id.to_string());
// Seed addresses both ways and form the initial link.
dialer.lookup.add_endpoint_info(peer.endpoint.addr());
peer.lookup.add_endpoint_info(dialer.endpoint.addr());
let mut conn_events = dialer
.transport
.subscribe_conn_events()
.await
.expect("subscribe dialer conn events");
dialer.transport.connect_peer(peer.endpoint.addr()).await;
// The peer accepts the dialer's initial connection.
let conn1 = tokio::time::timeout(Duration::from_secs(10), peer.conns_rx.recv())
.await
.expect("timed out awaiting initial connection")
.expect("connection channel closed");
assert!(
await_reconnect(
&mut conn_events,
tokio::time::Instant::now() + Duration::from_secs(10)
)
.await,
"initial link should report Connecting then Connected"
);
// Drop the live link by closing it from the peer side. An explicit close
// sends a CONNECTION_CLOSE the dialer sees right away.
conn1.close(0u32.into(), b"transient drop");
drop(conn1);
// The dialer's supervisor should re-dial the (unchanged) peer address, and
// the peer accepts the fresh connection.
let conn2 = tokio::time::timeout(Duration::from_secs(15), peer.conns_rx.recv())
.await
.expect("timed out awaiting reconnect")
.expect("connection channel closed");
assert!(
await_reconnect(
&mut conn_events,
tokio::time::Instant::now() + Duration::from_secs(15)
)
.await,
"dropped link should report Connecting (down) then Connected (recovered)"
);
// Audio must flow over the rebuilt link. Give the freshly-registered send
// path a beat, then stream frames and confirm they reach the peer on conn2.
tokio::time::sleep(Duration::from_millis(200)).await;
let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
for seq in 0..30u32 {
dialer.transport.broadcast(packet(&mut enc, seq));
tokio::time::sleep(Duration::from_millis(5)).await;
}
let received = count_audio(
&conn2,
25,
tokio::time::Instant::now() + Duration::from_secs(3),
)
.await;
assert!(
received >= 20,
"audio should resume after reconnect; got {received} frames"
);
}
/// A graceful close (our goodbye code `1`) must report `Left` and NOT reconnect —
/// the opposite of `dialer_reconnects_after_link_drops`, which closes with code 0
/// (a transient drop). This is what lets a clicked "disconnect" evict the peer
/// promptly instead of leaving it stuck "reconnecting". Mirrors the reconnect
/// test's harness but asserts the supervisor stops rather than re-dials.
#[tokio::test]
async fn dialer_reports_left_on_graceful_close() {
let s1 = iroh::SecretKey::generate();
let s2 = iroh::SecretKey::generate();
let (dialer_secret, peer_secret) = if s1.public().to_string() < s2.public().to_string() {
(s1, s2)
} else {
(s2, s1)
};
let dialer = spawn_node_with_key(dialer_secret).await;
let mut peer = spawn_capture_peer(peer_secret).await;
dialer.lookup.add_endpoint_info(peer.endpoint.addr());
peer.lookup.add_endpoint_info(dialer.endpoint.addr());
let mut conn_events = dialer
.transport
.subscribe_conn_events()
.await
.expect("subscribe dialer conn events");
dialer.transport.connect_peer(peer.endpoint.addr()).await;
let conn1 = tokio::time::timeout(Duration::from_secs(10), peer.conns_rx.recv())
.await
.expect("timed out awaiting initial connection")
.expect("connection channel closed");
assert!(
await_reconnect(
&mut conn_events,
tokio::time::Instant::now() + Duration::from_secs(10)
)
.await,
"initial link should report Connecting then Connected"
);
// Close with the goodbye code (1) — an intentional leave, not a blip.
conn1.close(1u32.into(), b"leave");
drop(conn1);
// The supervisor must surface `Left` (and must NOT try to reconnect: any
// Connecting/Connected here would be the bug). The peer also must not see a
// re-dial.
let deadline = tokio::time::Instant::now() + Duration::from_secs(10);
match tokio::time::timeout_at(deadline, conn_events.recv()).await {
Ok(Some(ConnEvent::Left(id))) => {
assert_eq!(id, peer.endpoint.id(), "Left should name the peer");
}
Ok(Some(ConnEvent::Connecting(_))) | Ok(Some(ConnEvent::Connected(_))) => {
panic!("a graceful close must not trigger a reconnect");
}
Ok(None) => panic!("conn events channel closed unexpectedly"),
Err(_) => panic!("timed out waiting for Left after graceful close"),
}
// And no re-dial reaches the peer within a short window.
let redial = tokio::time::timeout(Duration::from_secs(2), peer.conns_rx.recv()).await;
assert!(
redial.is_err(),
"supervisor must not re-dial after a graceful leave"
);
}
#[tokio::test]
async fn dialer_connects_and_reconnects_without_an_address_lookup() {
// Regression for the redial-loop bug: a transient drop that also triggers a
// gossip Leave/NeighborDown purges the peer from the address lookup, so a
// dialer that resolves by bare id (the old behavior) has nothing to dial and
// loops forever. The fix retains the peer's full address in the transport and
// dials it directly. To prove the retained address — not the lookup — is what
// carries the dial, we never seed the dialer's lookup with the peer at all
// (and disable the relay), so a bare-id dial could not resolve an address for
// either the initial connect or the reconnect; only `connect_peer`'s retained
// EndpointAddr makes them possible.
let s1 = iroh::SecretKey::generate();
let s2 = iroh::SecretKey::generate();
let (dialer_secret, peer_secret) = if s1.public().to_string() < s2.public().to_string() {
(s1, s2)
} else {
(s2, s1)
};
let dialer = spawn_node_with_key(dialer_secret).await;
let mut peer = spawn_capture_peer(peer_secret).await;
let dialer_id = dialer.endpoint.id();
let peer_id = peer.endpoint.id();
assert!(dialer_id.to_string() < peer_id.to_string());
// Deliberately leave the dialer's lookup empty for the peer. (The peer side
// only ever accepts, so it needs no addressing for the dialer.)
let mut conn_events = dialer
.transport
.subscribe_conn_events()
.await
.expect("subscribe dialer conn events");
// The full address handed here is the dialer's only way to reach the peer.
dialer.transport.connect_peer(peer.endpoint.addr()).await;
let conn1 = tokio::time::timeout(Duration::from_secs(10), peer.conns_rx.recv())
.await
.expect("timed out awaiting initial connection (retained address path)")
.expect("connection channel closed");
assert!(
await_reconnect(
&mut conn_events,
tokio::time::Instant::now() + Duration::from_secs(10)
)
.await,
"initial link should report Connecting then Connected"
);
// Drop the live link from the peer side so the dialer must reconnect — again
// with no lookup to fall back on.
conn1.close(0u32.into(), b"transient drop");
drop(conn1);
let conn2 = tokio::time::timeout(Duration::from_secs(15), peer.conns_rx.recv())
.await
.expect("timed out awaiting reconnect (retained address path)")
.expect("connection channel closed");
assert!(
await_reconnect(
&mut conn_events,
tokio::time::Instant::now() + Duration::from_secs(15)
)
.await,
"reconnect should report Connecting then Connected with no lookup at all"
);
// Confirm the rebuilt link actually carries audio.
tokio::time::sleep(Duration::from_millis(200)).await;
let mut enc = OpusEncoder::new(48000, Channels::Mono, Application::Voip).unwrap();
for seq in 0..30u32 {
dialer.transport.broadcast(packet(&mut enc, seq));
tokio::time::sleep(Duration::from_millis(5)).await;
}
let received = count_audio(
&conn2,
25,
tokio::time::Instant::now() + Duration::from_secs(3),
)
.await;
assert!(
received >= 20,
"audio should resume after reconnecting via the retained address; got {received} frames"
);
}
/// A node that also serves the file plane (`FILES_ALPN`), mirroring how core
/// registers the `FileRouter` for a session.
async fn spawn_file_server() -> Node {
let lookup = MemoryLookup::new();
let endpoint = Endpoint::builder(presets::Minimal)
.secret_key(iroh::SecretKey::generate())
.relay_mode(RelayMode::Disabled)
.address_lookup(lookup.clone())
.bind()
.await
.expect("bind endpoint");
let transport = Arc::new(IrohTransport::new(endpoint.clone()));
let audio_router = AudioRouter::new();
audio_router.bind(&transport);
let file_router = peerspeak::network::iroh_impl::FileRouter::new();
file_router.bind(&transport);
let router = Router::builder(endpoint.clone())
.accept(AUDIO_ALPN, audio_router)
.accept(peerspeak::protocol::FILES_ALPN, file_router)
.spawn();
Node {
endpoint,
transport,
_router: router,
lookup,
}
}
/// Phase 3C: a file fetch must deliver EXACTLY the declared size — short,
/// overlong, and unknown-id transfers are all rejected with local errors, and
/// an exact transfer round-trips byte-identically.
#[tokio::test]
async fn file_plane_requires_exact_declared_size() {
let fetcher = spawn_node().await;
let server = spawn_file_server().await;
fetcher.lookup.add_endpoint_info(server.endpoint.addr());
server.lookup.add_endpoint_info(fetcher.endpoint.addr());
let server_id = server.endpoint.id();
// Member gating: the server only serves current room members.
server.transport.admit_audio_sender(fetcher.endpoint.id());
let blob = vec![42u8; 1000];
let id = [7u8; 32];
server
.transport
.serve_attachment(id, Arc::new(blob.clone()));
// Exact declared size: byte-identical round trip.
let got = fetcher
.transport
.fetch_blob(server_id, id, 1000)
.await
.expect("exact-size fetch succeeds");
assert_eq!(got, blob);
// Declared larger than served (short transfer): rejected, not cached as-is.
let err = fetcher
.transport
.fetch_blob(server_id, id, 2000)
.await
.expect_err("short transfer must fail");
assert!(
err.to_string().contains("incomplete transfer"),
"unexpected error: {err}"
);
// Declared smaller than served (overlong transfer): the bounded read
// rejects the stream rather than truncating it into a "valid" result.
let err = fetcher
.transport
.fetch_blob(server_id, id, 500)
.await
.expect_err("overlong transfer must fail");
assert!(
err.to_string().contains("read failed"),
"unexpected error: {err}"
);
// Unknown id: the empty body reads as the sender no longer having it.
let err = fetcher
.transport
.fetch_blob(server_id, [9u8; 32], 1000)
.await
.expect_err("unknown id must fail");
assert!(
err.to_string().contains("no longer has the file"),
"unexpected error: {err}"
);
}