Latency start value
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A planning value instead of false certainty
The calculator combines RTT, jitter, packet loss, upload, and protocol headroom. The result is a sensible starting value, but no substitute for a real test stream on the same cellular route.
Calculate latency and headroom
Safe target bitrate
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Upload headroom
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Rating
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What RTT, jitter, and packet loss mean
RTT: there and back
RTT describes how long a data packet takes to reach the destination and for the reply to come back. It is not the same as the total delay of the live stream, but it is an important basis for the SRT buffer you need.
Jitter: varying transit time
Jitter happens when packets do not arrive at even intervals. Cellular connections are especially sensitive to it. A larger buffer can absorb such swings but adds delay.
Packet loss: missing data
With packet loss, data has to be retransmitted or covered up by the decoder. If the loss stays high, extra latency and headroom only help so much. Then you should improve the connection first.
Upload headroom: room to spare
The video bitrate should not use up the available upload completely. You need headroom for audio, protocol data, retransmissions, and short swings. What counts is the performance you can rely on over time, not the highest reading from a speed test.
What SRTLA bonding changes
SRTLA can bond several internet connections and spread data across different paths. That makes the stream more resilient to dropouts if the connections really complement each other, for example SIM cards from different networks together with Wi-Fi or Ethernet. Bonding does not create guaranteed bandwidth, though, and does not remove dead zones that hit all the cellular paths you use at the same time.
More connections therefore do not automatically mean a correspondingly multiplied upload rate. For planning, the stable total performance under real conditions matters more than the sum of individual best values.
A real-world example for a mobile IRL stream
Say two connections together deliver about 12 Mbit/s of upload on a lasting basis. The stream is meant to go out at 6 Mbit/s of video, while the measured RTT is 80 milliseconds, jitter is 25 milliseconds, and packet loss is one percent. From that, the calculator suggests a starting value for latency and target bitrate.
This value is deliberately not a firm guarantee. Use it to start a test stream on the same route you will use later. If the picture stays stable, you can keep the setting. If artifacts, dropouts, or big swings show up, raise the latency step by step first and then check whether bitrate or connection selection needs adjusting.
Using the result properly
- Start with the calculated value and test for at least ten minutes under realistic conditions.
- If you see picture errors or big swings, raise the latency step by step.
- More SIM cards help as redundancy above all when they do not all use the same cellular network.
- The sum of short speed tests is not a guaranteed sustained performance.
Privacy and security
This tool runs locally in your browser. IRL4YOU neither transmits nor stores what you enter. Credentials still do not belong in screenshots or public support messages.