Crate rtp_engine 

rtp-engine

A pure Rust RTP media engine for VoIP applications.

This crate provides the building blocks for real-time audio communication:

• Codecs: G.711 (μ-law/A-law) and Opus encoding/decoding
• RTP/RTCP: Packet construction, parsing, and statistics
• SRTP: Secure RTP with AES-CM-128-HMAC-SHA1-80
• Audio devices: Cross-platform capture and playback via cpal
• Resampling: Sample rate conversion between codecs and devices

Quick Start

use rtp_engine::{MediaSession, CodecType};
use std::net::SocketAddr;

#[tokio::main]
async fn main() -> Result<(), Box<dyn std::error::Error>> {
    let remote: SocketAddr = "192.168.1.100:5004".parse()?;
     
    // Start a media session with G.711 μ-law
    let session = MediaSession::start(10000, remote, CodecType::Pcmu).await?;
     
    // The session captures from mic and plays to speaker automatically
    // Send DTMF
    session.send_dtmf("1");
     
    // Mute/unmute
    session.set_mute(true);
     
    // Get statistics
    let stats = session.stats();
    println!("Packets sent: {}", stats.packets_sent);
     
    // Stop when done
    session.stop();
    Ok(())
}

Feature Flags

• g711 (default): G.711 μ-law and A-law codecs
• opus: Opus codec support (requires libopus)
• srtp: SRTP/SRTCP encryption
• device: Audio device capture/playback via cpal

Re-exports

pub use codec::CodecType;

pub use error::Error;

pub use error::Result;

pub use jitter::JitterBuffer;

pub use jitter::JitterConfig;

pub use jitter::JitterMode;

pub use jitter::JitterStats;

pub use resample::f32_to_i16;

pub use resample::i16_to_f32;

pub use resample::resample_linear;

pub use resample::resample_linear_i16;

pub use rtp::RtpStats;

pub use srtp::SrtpContext;

Modules

codec

Audio codec implementations for RTP media.

device

Audio device abstraction for capture and playback.

error

Error types for rtp-engine.

jitter

Jitter buffer implementations for RTP media.

resample

Audio resampling utilities.

rtp

RTP (Real-time Transport Protocol) implementation.

srtp

SRTP (Secure RTP) implementation per RFC 3711.

Structs

MediaSession

A complete RTP media session with bidirectional audio.