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improve documentation (#5586)

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Alessandro Ros
2026-03-18 09:47:46 +01:00
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docs/4-other/25-decrease-packet-loss.md
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MediaMTX is meant for routing live streams, and makes use of a series of protocols and techniques which try to preserve the real-time aspect of streams and minimize latency at cost of losing packets in transmit, in particular:
- most protocols are built on UDP, which is an "unreliable transport", specifically picked because it allows to drop late packets in case of network congestions.
- most protocols are built on UDP, which is an "unreliable transport", specifically picked because it allows dropping late packets in case of network congestion.
- there's a circular buffer that stores outgoing packets and drops packets if full.
Packet losses are usually detected and printed in _MediaMTX_ logs.
If you need to improve the stream reliability and decrease packet losses, the first thing to do is to check whether the physical network between the _MediaMTX_ instance and the intended publishers and readers has sufficient bandwidth for transmitting the media stream. Most of the times, packet losses are caused by a network which is not fit for this scope. This limitation can be overcome by either recompressing the stream with a lower bitrate, or by upgrading the network infrastructure (routers, cables, Wi-Fi, firewalls, topology, etc).
If you need to improve the stream reliability and decrease packet losses, the first thing to do is to check whether the physical network between the _MediaMTX_ instance and the intended publishers and readers has sufficient bandwidth for transmitting the media stream. Most of the time, packet losses are caused by a network which is not fit for this scope. This limitation can be overcome by either recompressing the stream with a lower bitrate, or by upgrading the network infrastructure (routers, cables, Wi-Fi, firewalls, topology, etc).
Nonetheless there are some parameters that can be tuned to improve the situation, at cost of increasing RAM consumption: