Application developers
Add real-time audio transport to apps without building the network audio layer yourself.
Roc Streaming is an open audio backbone for real-time streaming over IP networks, built for controlled latency and resilience across wired, wireless, and wide-area networks.

Sending audio over Wi-Fi, LAN, or the Internet is easy. Making it feel like a direct cable between audio devices is a challenging task.
A cable gives you predictable latency, stable quality, and no audible dropouts. A network adds jitter, packet loss, and unpredictable latency.
Roc Streaming is an implementation of real-time audio transport designed to close that gap: controlled latency, loss recovery, clock handling, and synchronization over standard IP networks.
Roc Streaming is for people who need to move audio between software, devices, rooms, or locations over ordinary IP networks, without relying on a proprietary transport stack.
Add real-time audio transport to apps without building the network audio layer yourself.
Build speakers, receivers, streamers, soundbars, and audio appliances on top of an open transport core.
Deploy zoned audio across offices, retail, hospitality, schools, gyms, venues, and other shared spaces.
Send audio between rooms, sites, or locations for monitoring, IFB, rehearsal, collaboration, and inter-site workflows.
Build custom multi-room setups or connect Roc to Linux, macOS, Android, and community-built audio systems.
Roc Streaming lets you choose the right level of control: use Roc Cast as a ready-to-use application, RocD as a deployable audio service, or Roc Toolkit as the foundation for fully custom real-time audio systems.
Fully open source, vendor-neutral, and licensed under commercial-friendly MPL-2.0 license.
Choose the layer that matches your needs: application, service, or library.
Use Roc from C, C++, Rust, Go, Java, or through RocD’s HTTP API.
Connect Roc to existing audio systems through OS audio stacks or virtual devices.
At the core of Roc Streaming is Roc Toolkit, the specialized transport foundation behind its real-time audio capabilities.
Streaming of CD- and HD-quality audio, from uncompressed PCM to lossless or lossy codecs.
End-to-end latency down to 10 milliseconds, with strict latency bounds and adaptive tuning for changing networks.
Recovering of lost packets without quality drop using Forward Erasure Correction codes or masking losses with Packet Loss Concealment algorithms.
Uses sensible defaults and adaptive algorithms for typical networks, with low-level controls available when your use case needs precise tuning.
Efficient lightweight core designed for real-time. Portable across OSes and CPUs, including embedded, desktop, and mobile platforms.
Built on the foundation of open Internet standards, proven over time and validated by numerous applications.
Learn how the ecosystem fits together, join the open-source community, or work with the Roc Streaming team on professional services.
Learn how Roc Toolkit, RocD, Roc Cast, bindings, and OS integrations fit together, and choose the right entry point for your use case.
Read the docs, explore the code, ask questions, report issues, and contribute to the open-source projects.
Work with the Roc Streaming team on custom solutions, product integration, consulting, and feature development on request.