Whats the difference with RT and SRT?
SRT provides secure, reliable, tunable low-latency transport for live video, while traditional real-time transport methods (RTP-based streams) prioritize minimal latency but lack built-in reliability and encryption. This article explains their core differences in reliability, latency, security, and network behavior.
What RT refers to in streaming
In broadcast and streaming discussions, RT often means Real-Time Transport Protocol (RTP) used with RTCP for control. It is the backbone for carrying audio and video over IP networks and is designed for low latency, but it does not include guaranteed delivery or default encryption.
RTP basics
RTP handles timing, payload identification, sequence numbering, and synchronization. It typically runs over UDP to keep delay low, while RTCP provides periodic quality feedback. Encryption is not part of RTP itself; when security is needed, SRTP (Secure Real-Time Transport Protocol) is used as an add-on.
The following list highlights core RTP characteristics:
- Operates on top of UDP for low latency.
- Provides timing information, sequence numbers, and payload IDs for synchronization.
- Offers no built-in reliability or retransmission guarantees.
- RTCP gives quality feedback but does not ensure delivery.
- Encryption is optional via SRTP, not part of the core RTP spec.
In practice, RTP is fast and standardized but relies on additional mechanisms to address packet loss and security, which can introduce complexity in live workflows.
What SRT is and how it works
Secure Reliable Transport (SRT) is a transport protocol purpose-built for live video over the Internet. It combines low latency with reliability, security, and NAT traversal to make streaming over imperfect networks feasible.
Key features of SRT
- End-to-end security with built-in encryption (AES-128 by default).
- Reliability over lossy networks through optional ARQ (automatic retransmission) and FEC (forward error correction).
- Low-latency operation with configurable latency settings to balance speed and resilience.
- NAT traversal and firewall friendliness via a handshake-based connection setup.
- UDP-based transport with additional reliability and security features baked in.
These features make SRT well-suited for contribution links, remote productions, and scenarios where packet loss and network variance are common.
Where SRT sits in the workflow
In many modern live workflows, SRT handles the transport between the origin and the receiving edge or CDN, while standard streaming formats (like HLS or DASH) or protocols (like RTMP/RTP) may still be used for ingestion or playback. SRT’s strengths—reliability, encryption, and NAT traversal—address gaps often encountered with traditional RTP-based paths.
Where RTP and SRT differ in practice
The practical differences between RTP-based transport and SRT can be summarized along several axes that matter for live video workflows:
- Reliability: RTP is best-effort; SRT provides ARQ/FEC options to recover lost packets.
- Security: RTP itself is not encrypted by default; SRTP adds encryption; SRT includes encryption as a core feature.
- Latency control: RTP latency depends on endpoint jitter buffers; SRT exposes explicit latency settings for balance between speed and robustness.
- NAT/firewall traversal: RTP relies on separate NAT traversal tools or VPNs; SRT includes built-in NAT traversal capabilities.
- Typical use cases: RTP is common in traditional, tightly controlled networks and CDN ingest pipelines; SRT is favored for internet-based contributions, remote production, and scenarios with unreliable networks.
Choosing between these approaches depends on network conditions, security requirements, and the desired trade-off between latency and reliability. Many broadcasters today combine SRT for transport over the internet with standard streaming delivery (HLS/DASH) to audiences.
Summary
In essence, SRT is a modern, secure, and reliability-oriented transport designed for variable networks, offering encryption and adjustable latency, while RTP-based transport prioritizes ultra-low latency with standardization but relies on separate mechanisms for reliability and security. Understanding your network environment and workflow helps determine which approach best fits your live-streaming pipeline.
