What is SRT?

SRT operates over UDP, which does not guarantee packet delivery order or integrity. To handle this, SRT implements Application Layer Forward Error Correction (ARQ). When a packet is lost, the receiver requests a retransmission from the sender. This process happens quickly, keeping latency low compared to TCP, which waits for full acknowledgment before proceeding.

The protocol also handles network jitter by buffering data briefly. This allows the receiver to reassemble out-of-order packets without freezing the stream. SRT supports AES-128 encryption, securing the data in transit without adding significant overhead.

Key features include:

• Latency Control: Operators can set a target latency, and SRT adjusts buffering to meet it.
• Congestion Control: It adapts the data rate to match available bandwidth, preventing buffer bloat.
• Interoperability: SRT works with standard UDP ports, making it easy to configure through firewalls that block other protocols.

Contribution links often run over cellular, satellite, or public internet connections. These links are prone to packet loss and variable latency. If your ingest protocol fails when the network wobbles, you lose the live event. SRT is designed for these conditions. It prioritizes stream continuity over perfect packet delivery, using its recovery mechanisms to fill gaps.

For operators, this means fewer interruptions during live broadcasts. You can send high-bitrate streams from remote locations without dropping frames. The built-in encryption also addresses security concerns. You do not need to wrap the stream in an additional secure layer, which simplifies your workflow and reduces processing load on encoders.

Using SRT for contribution allows you to scale your live operations. Whether you are ingesting from a studio or a field reporter, the protocol provides a consistent, reliable transport layer that sits between your encoder and your streaming platform.

RTMP is a long-standing standard for live streaming, but it relies on TCP. TCP keeps data integrity but adds latency because it waits for acknowledgments. SRT uses UDP with its own error correction, resulting in lower latency and better performance on unstable networks.

• Ignoring Latency Settings: Leaving latency at default values may not suit your encoder or network. Test different latency modes (sender, receiver, variable) to find the sweet spot.
• Firewall Misconfiguration: SRT uses UDP. If your firewall blocks UDP packets, the connection will fail. Make sure the correct ports are open for both inbound and outbound traffic.
• Overlooking Bandwidth: SRT adapts to bandwidth, but if the link is too weak, quality will drop. Monitor bandwidth usage to make sure it matches your stream requirements.
• Mixing Protocols Incorrectly: Some encoders support both SRT and RTMP. Make sure you are configuring the correct protocol type in your encoder settings to avoid connection errors.