How to set SRT latency and buffer correctly
Why extra buffer can recover lost packets, how RTT affects the setting, and sensible values for stable and mobile networks.
| Property | SRT | SRTLA | RTMP / RTMPS |
|---|---|---|---|
| Typical use | Resilient single link | Mobile IRL with bonding | OBS, studio, broad compatibility |
| Multiple internet links | No | Yes | No |
| Packet-loss recovery | Yes, retransmission | Yes, plus link aggregation | Not designed like SRT |
| Configurable recovery latency | Yes | Yes, through SRT transport | Not an SRT-style recovery window |
| Compatibility | Good, but not universal | Mainly IRL apps/tools | Very broad |
SRT is a UDP-based transport designed for reliable live delivery. Its latency window gives lost packets time to be retransmitted. That helps with jitter and moderate packet loss, while deliberately adding some delay.
Choose SRT when you have one internet path but want more resilience than a simple RTMP connection. See the SRT latency and buffer guide for tuning the recovery window.
SRTLA adds link aggregation in front of SRT. Multiple network paths – for example 5G, LTE and Wi‑Fi – can be used at the same time. Its implementation dynamically balances traffic across available links for capacity aggregation and redundancy.
Choose SRTLA when you stream on the move and have at least two independent internet connections. That is where a single mobile link often becomes the weakest part of an IRL stream.
RTMP is still widely supported by OBS, vMix, hardware encoders and streaming platforms. RTMPS carries the same RTMP stream over TLS, protecting the connection to the ingest from passive eavesdropping and tampering.
Choose RTMP/RTMPS when your encoder does not support SRT, you have a stable fixed connection, or broad compatibility matters more than bonding and SRT recovery.