webrtc-core-0.5.35 has been yanked.

webrtc-core

Production-ready, low-latency core for realtime audio pipelines: slab-based packet buffers, jitter handling, SRTP (AES‑GCM) protection, and RTCP feedback (NACK / TWCC). Designed for high-throughput, low-footprint deployments.

[1] Why this crate

Purpose-built API for high-throughput, low-latency production systems (VoIP, conferencing, and realtime media services).
Highly concurrent, low-allocation primitives: slab allocator, index/byte rings, jitter buffer and RTCP queue — engineered to minimize pause and allocation jitter.
In-place SRTP (AES‑GCM) protection to avoid extra copies on the hot path.

[2] Enterprise readiness

Deterministic memory usage via preallocated slabs and ring buffers.
Thread-safe, cache-padded atomics and minimal locking for scalable multi-threaded ingestion.
Small runtime and dependency surface to ease embeddability and auditability.
Designed for easy horizontal scaling: stateless ingest + small per-worker footprint.

Design philosophy: optimize the real-time audio fast path while keeping memory bounded and latency predictable.

[3] Quickstart

cargo add webrtc-core

Minimal usage

use webrtc_core::EngineHandle;

let handle = EngineHandle::builder().build();
let payload = vec![0u8; 160];
handle.feed_packet(&payload, 1u16, 0x1234).ok();

Provide SRTP key (in-place protection for future packets)

let key = [0u8; 32];
handle.provide_keying_material(key);

RTCP sender (async example)

use std::net::SocketAddr;
use std::sync::Arc;
use tokio::net::UdpSocket;
use webrtc_core::EngineHandle;

#[tokio::main]
async fn main() -> anyhow::Result<()> {
    let addr: SocketAddr = "127.0.0.1:9000".parse()?;
    let peer: SocketAddr = "127.0.0.1:9001".parse()?;
    let socket = Arc::new(UdpSocket::bind(addr).await?);
    let handle = EngineHandle::builder().build();
    let rtcp_task = handle.start_rtcp_sender(socket.clone(), peer);

    let payload = vec![0u8; 160];
    let _ = handle.feed_packet(&payload, 42u16, 0x1234);

    let _ = rtcp_task.await;
    Ok(())
}

[4] Core API

EngineHandle — ergonomic entry point: builder, feed_packet, provide_keying_material, start_rtcp_sender, shutdown.
MediaEngine — internal processing loop: jitter handling, packet processing, RTCP emission.
SrtpContext — AES‑GCM SRTP helpers (in-place protect/unprotect).
SlabAllocator, IndexRing, AudioJitterBuffer, RtcpSendQueue — low-level primitives for high-throughput audio.

[5] Notes

Emphasizes predictable latency over generality; uses preallocated slabs and ring buffers to avoid allocations in the hot path.
RTCP generation is conservative and designed to be driven by the jitter buffer's gap detection.

[6] Performance

Measured snapshot (this environment)

Command run: cargo run --release --example bench_micro
Observed (release build on this Windows dev machine):
- avg protect_inplace: 31 ns
- implied theoretical protects/sec: ~32M (1_000_000_000 / 31)

[7] Profiling tips

Build with --release and use perf (Linux) or Windows Performance Analyzer for hotspots.
For host-optimized results set RUSTFLAGS='-C target-cpu=native' when building.

[8] Example behaviors

examples/bench_micro.rs prints micro-benchmark metrics (see above).
examples/discord_clone.rs is a simple UDP+RTCP demo and starts a long-running RTCP sender loop - it will run until interrupted.

[9] Credits

xbl1e - creator and maintainer

webrtc-core 0.5.35

webrtc-core