# mediamtx v1.20.0

- Product: mediamtx (https://whatsnew.fyi/product/mediamtx)
- Vendor: bluenviron
- Date: 2026-08-05
- Version: v1.20.0
- Original notes: https://github.com/bluenviron/mediamtx/releases/tag/v1.20.0
- Permalink: https://whatsnew.fyi/product/mediamtx/releases/v1.20.0

What's New is an index, not a publisher: every entry below links to the vendor's own release notes, which are the authoritative source. Entries are labelled where they are hand-curated sample data, pre-releases, or drawn from a secondary source such as a developer blog.

Reuse: the summaries, labels and curation here are © What's New. Quote freely with attribution and a link back; wholesale republication of the corpus is not permitted — terms: https://whatsnew.fyi/terms. The vendors' own release notes remain their publishers'.

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- **added** — Support forwarding streams natively with destinations for each path configuration using RTSP, RTMP, and SRT protocols
- **added** — Support publishing and reading through native QUIC
- **added** — Support draft-17 for Media-Over-QUIC
- **added** — Support draft-16 for Media-Over-QUIC
- **added** — Add OpenAPI definition of the Playback server
- **fixed** — Support for regexp groups greater than 10
- **changed** — Increase max namespace field count to 32 for Media-Over-QUIC
- **fixed** — Fix inability to read some AV1 streams with RTSP by restoring stripping of temporal unit delimiters
- **changed** — Log write errors in RTSP
- **fixed** — Make multicast errors on single interfaces non-fatal to allow writing to other interfaces
- **fixed** — Fix parsing URL from multitrack OBS in RTMP server
- **changed** — Change mapping between URL and tcURL, app, streamKey in RTMP client
- **fixed** — Fix compatibility with YouTube in RTMP client
- **fixed** — Fix goroutine leak during part rotation in HLS muxer
- **changed** — Remove lower limit on AAC access unit count in MPEG-TS segments for HLS muxer
- **changed** — Recompute PTS of MPEG-TS AAC in HLS muxer
- **changed** — Sort WebRTC tracks deterministically by MID, RID, trackID and streamID
- **fixed** — Fix packet corruption when reading G722 in WebRTC
- **changed** — Reset recomputed audio PTS if it drifts too much in WebRTC
- **fixed** — Fix packet lost error when routing streams from WebRTC by recomputing sequence numbers after empty packets

##### New major features

General

* support forwarding streams natively (https://github.com/bluenviron/mediamtx/issues/5558) It is now possible to define forward destinations for each path configuration. For each destination, the server will create a client that will forward the stream to the intended destination. Supported protocols are RTSP, RTMP, SRT. API and metrics have also been improved to allow monitoring the new forwarding system. Documentation: https://mediamtx.org/docs/features/forward

Media-Over-QUIC

* support publishing and reading through native QUIC (https://github.com/bluenviron/mediamtx/issues/6039)
* support draft-17 (https://github.com/bluenviron/mediamtx/issues/6040)
* support draft-16 (https://github.com/bluenviron/mediamtx/issues/6045)

##### Fixes and improvements

General

* docs: add missing MoQ ports (https://github.com/bluenviron/mediamtx/issues/5983)
* Add OpenAPI definition of the Playback server (https://github.com/bluenviron/mediamtx/issues/5546)
* fix support for regexp groups greater than 10 (https://github.com/bluenviron/mediamtx/issues/6033)

Media-Over-QUIC

* increase max namespace field count to 32 (https://github.com/bluenviron/mediamtx/issues/6043)
* link primitives to draft-17 sections (https://github.com/bluenviron/mediamtx/issues/6044)

RTSP

* fix inability to read some AV1 streams with RTSP (https://github.com/bluenviron/mediamtx/issues/6001) (https://github.com/bluenviron/mediamtx/issues/6006) Since v1.16.0, temporal unit delimiters were not stripped from AV1 streams anymore. This has been restored, healing AV1 streams read with RTSP.
* log write errors (https://github.com/bluenviron/mediamtx/issues/6023)
* make multicast errors on single interfaces non-fatal (https://github.com/bluenviron/gortsplib/issues/1115) (https://github.com/bluenviron/mediamtx/issues/5574) When writing multicast packets to several interfaces at one, a write error to a single interface was fatal and prevented writing to the other ones. Fix this.

RTMP

* server: fix parsing URL from multitrack OBS (https://github.com/bluenviron/mediamtx/issues/6007) (https://github.com/bluenviron/gortmplib/issues/93)
* client: change mapping between URL and tcURL, app, streamKey (https://github.com/bluenviron/gortmplib/issues/94) (https://github.com/bluenviron/mediamtx/issues/4676) URLs passed to clients are now mapped into RTMP-native fields (tcURL, app, streamKey) in this way: tcURL contains URL without credentials and without fragment, app contains path and query of tcURL, streamKey contains the fragment.
* client: fix compatibility with YouTube (https://github.com/bluenviron/mediamtx/issues/5558) (https://github.com/bluenviron/gortmplib/issues/95)

HLS

* muxer: fix goroutine leak during part rotation (https://github.com/bluenviron/gohlslib/issues/372) (https://github.com/bluenviron/gohlslib/issues/373)
* muxer: remove lower limit on AAC access unit count in MPEG-TS segments (https://github.com/bluenviron/gohlslib/issues/329) (https://github.com/bluenviron/gohlslib/issues/377)
* muxer: recompute PTS of MPEG-TS AAC (https://github.com/bluenviron/gohlslib/issues/379) iOS requires a precise timestamp that is often not available in AAC streams. Recompute timestamp from scratch.

WebRTC

* sort tracks in a deterministic way (https://github.com/bluenviron/mediamtx/issues/5988) (https://github.com/bluenviron/mediamtx/issues/5989) When ingesting tracks with WebRTC, track order was randomized, preventing multi-track always-available streams from working reliably, since they require tracks to be ordered in a precise way. WebRTC tracks are not ordered by MID, RID, trackID and streamID respectively.
* fix packet corruption when reading G722 (https://github.com/bluenviron/mediamtx/issues/6000)
* reset recomputed audio PTS if it drifts too much (https://github.com/bluenviron/mediamtx/issues/6021)
* fix "packet lost" error when routing streams from WebRTC (https://github.com/bluenviron

_[Truncated at 4000 characters — full notes: https://github.com/bluenviron/mediamtx/releases/tag/v1.20.0]_