DATA / STREAMING

SRT Latency & Buffer Calculator

Calculate the SRT latency buffer required to survive packet loss and jitter on lossy public internet contribution links.

Essential for determining the correct buffer depth for remote production. To accurately dimension your buffer, first determine your video bitrate requirements. Explore more broadcast engineering tools for IP video and media networking.

Recommended SRT Latency 180 ms
Buffer Capacity Required 225 KB
Est. Packets in Flight 171 pkt
Diagnostics: Math Result: 180 ms.

Common SRT Use Cases

REMI / Remote Production

Sending ISO camera feeds from a stadium directly to a centralized control room via vMix or hardware encoders over public internet lines, minimizing satellite truck costs.

Cloud Vision Mixing

Pushing live venue feeds to AWS/Azure instances running OBS or vMix in the cloud, requiring stable ingestion buffering to handle fluctuating cloud routing.

Bonded Cellular (LiveU / Aviwest)

Using SRT over 4G/5G modems where packet loss and jitter can spike rapidly. High SRT latency values (500ms+) provide the necessary safety net for clean transmission.

Global Syndication

Distributing a master broadcast feed across continents. RTT can easily exceed 200ms, meaning SRT Latency must safely exceed 800ms–1000ms for continuous playback without buffer underrun.

Design Stable Links

Save this calculator for your next remote production or cloud broadcast.


What is the SRT Latency & Buffer Calculator?

Optimizes the Secure Reliable Transport (SRT) protocol for public internet contribution. It calculates the necessary buffer depth required to survive packet loss across lossy Wide Area Networks (WAN).

This utility was engineered to eliminate the guesswork from high-pressure broadcast environments. Whether you are managing multi-camera sync or complex IP networking, precision measurements ensure that the technical infrastructure stays stable throughout the duration of the event.

Real-World Use Cases

Used by MCR engineers to tune contribution links from remote venues, ensuring that even with 5-10% packet drop on a cellular bond, the video remains jitter-free and stable.

Understanding these variables is critical for minimizing system-wide latency and avoiding technical debt in live productions. This tool provides instant validation of your technical specifications before you even power on a single piece of hardware.

Technical Methodology

Follows the standard formula where SRT latency is calculated based on Round Trip Time (RTT) and a multiplier (typically 4x) to allow for sufficient retransmission windows or 'ARQ' attempts. To accurately dimension your buffer, you must first determine your stream's Video Bitrate in Mbps for 1080p or 4K content.

Our formulas are based on established SMPTE and EBU standards. By calculating these values theoretically, engineering teams can design predictable workflows and set clear expectations for clients regarding production quality and protocol limitations.