From 5259283cf7f0a70688e438e40de8a786eeb3c368 Mon Sep 17 00:00:00 2001
From: ‘liusuyi’ <1951119284@qq.com>
Date: 星期六, 02 九月 2023 14:59:46 +0800
Subject: [PATCH] 优化
---
lib/mediamtx/mediamtx.yml | 49 ++++++++++++++++++++++++++++++++-----------------
1 files changed, 32 insertions(+), 17 deletions(-)
diff --git a/lib/mediamtx/mediamtx.yml b/lib/mediamtx/mediamtx.yml
index c26403a..9598640 100644
--- a/lib/mediamtx/mediamtx.yml
+++ b/lib/mediamtx/mediamtx.yml
@@ -64,8 +64,8 @@
###############################################
# RTSP parameters
-# Disable support for the RTSP protocol.
-rtspDisable: no
+# Enable support for the RTSP protocol.
+rtsp: yes
# List of enabled RTSP transport protocols.
# UDP is the most performant, but doesn't work when there's a NAT/firewall between
# server and clients, and doesn't support encryption.
@@ -104,8 +104,8 @@
###############################################
# RTMP parameters
-# Disable support for the RTMP protocol.
-rtmpDisable: no
+# Enable support for the RTMP protocol.
+rtmp: yes
# Address of the RTMP listener. This is needed only when encryption is "no" or "optional".
rtmpAddress: :1935
# Encrypt connections with TLS (RTMPS).
@@ -124,8 +124,8 @@
###############################################
# HLS parameters
-# Disable support for the HLS protocol.
-hlsDisable: yes
+# Enable support for the HLS protocol.
+hls: no
# Address of the HLS listener.
hlsAddress: :8888
# Enable TLS/HTTPS on the HLS server.
@@ -180,8 +180,8 @@
###############################################
# WebRTC parameters
-# Disable support for the WebRTC protocol.
-webrtcDisable: no
+# Enable support for the WebRTC protocol.
+webrtc: yes
# Address of the WebRTC listener.
webrtcAddress: :8889
# Enable TLS/HTTPS on the WebRTC server.
@@ -206,7 +206,7 @@
# STUN servers are used to obtain the public IP of server and clients. They are
# needed when server and clients are on different LANs.
# TURN/TURNS servers are needed when a direct connection between server and
- # clients is not possible. All traffic is routed through them.
+ # clients is not possible. All traffic is routed through them. stun:stun.l.google.com:19302
- url: stun:stun.l.google.com:19302
# if user is "AUTH_SECRET", then authentication is secret based.
# the secret must be inserted into the password field.
@@ -227,25 +227,36 @@
webrtcICETCPMuxAddress:
###############################################
+# SRT parameters
+
+# Enables support for the SRT protocol.
+srt: yes
+# Address of the SRT listener.
+srtAddress: :8890
+
+###############################################
# Path parameters
# These settings are path-dependent, and the map key is the name of the path.
-# It's possible to use regular expressions by using a tilde as prefix.
-# For example, "~^(test1|test2)$" will match both "test1" and "test2".
-# For example, "~^prefix" will match all paths that start with "prefix".
-# The settings under the path "all" are applied to all paths that do not match
+# It's possible to use regular expressions by using a tilde as prefix,
+# for example "~^(test1|test2)$" will match both "test1" and "test2",
+# for example "~^prefix" will match all paths that start with "prefix".
+# Settings under the path "all" are applied to all paths that do not match
# another entry.
paths:
all:
# Source of the stream. This can be:
- # * publisher -> the stream is published by a RTSP or RTMP client
+ # * publisher -> the stream is published by a RTSP, RTMP, WebRTC or SRT client
# * rtsp://existing-url -> the stream is pulled from another RTSP server / camera
# * rtsps://existing-url -> the stream is pulled from another RTSP server / camera with RTSPS
# * rtmp://existing-url -> the stream is pulled from another RTMP server / camera
# * rtmps://existing-url -> the stream is pulled from another RTMP server / camera with RTMPS
# * http://existing-url/stream.m3u8 -> the stream is pulled from another HLS server
# * https://existing-url/stream.m3u8 -> the stream is pulled from another HLS server with HTTPS
- # * udp://ip:port -> the stream is pulled from UDP, by listening on the specified IP and port
+ # * udp://ip:port -> the stream is pulled with UDP, by listening on the specified IP and port
+ # * srt://existing-url -> the stream is pulled from another SRT server
+ # * whep://existing-url -> the stream is pulled from another WebRTC server
+ # * wheps://existing-url -> the stream is pulled from another WebRTC server with HTTPS
# * redirect -> the stream is provided by another path or server
# * rpiCamera -> the stream is provided by a Raspberry Pi Camera
source: publisher
@@ -268,6 +279,8 @@
# If sourceOnDemand is "yes", the source will be closed when there are no
# readers connected and this amount of time has passed.
sourceOnDemandCloseAfter: 10s
+ # Maximum number of readers. Zero means no limit.
+ maxReaders: 0
###############################################
# Authentication path parameters
@@ -293,8 +306,8 @@
###############################################
# Publisher path parameters (when source is "publisher")
- # do not allow another client to disconnect the current publisher and publish in its place.
- disablePublisherOverride: no
+ # allow another client to disconnect the current publisher and publish in its place.
+ overridePublisher: yes
# if no one is publishing, redirect readers to this path.
# It can be can be a relative path (i.e. /otherstream) or an absolute RTSP URL.
fallback:
@@ -366,6 +379,8 @@
rpiCameraEV: 0
# Region of interest, in format x,y,width,height
rpiCameraROI:
+ # whether to enable HDR on Raspberry Camera 3.
+ rpiCameraHDR: false
# tuning file
rpiCameraTuningFile:
# sensor mode, in format [width]:[height]:[bit-depth]:[packing]
--
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