Ready-to-use SRT / WebRTC / RTSP / RTMP / LL-HLS media server and media proxy that allows to read, publish, proxy, record and playback video and audio streams.
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# supported stream protocols (the handshake is always performed with TCP)
protocols: [udp, tcp]
# port of the TCP RTSP listener
rtspPort: 8554
# port of the UDP RTP listener
rtpPort: 8000
# port of the UDP RTCP listener
rtcpPort: 8001
# script to run when a client connects
preScript:
# script to run when a client disconnects
postScript:
# timeout of read operations
readTimeout: 5s
# timeout of write operations
writeTimeout: 5s
# time after which a stream is considered dead
streamDeadAfter: 15s
# supported authentication methods
authMethods: [basic, digest]
# enable pprof on port 9999 to monitor performance
pprof: false
# these settings are path-dependent. The settings under the path 'all' are
# applied to all paths that do not match a specific entry.
paths:
all:
# source of the stream - this can be:
# * record -> the stream is provided by a client through the RECORD command (like ffmpeg)
# * rtsp://original-url -> the stream is pulled from another RTSP server
source: record
# if the source is an RTSP url, this is the protocol that will be used to pull the stream
sourceProtocol: udp
# username required to publish
publishUser:
# password required to publish
publishPass:
# IPs or networks (x.x.x.x/24) allowed to publish
publishIps: []
# username required to read
readUser:
# password required to read
readPass:
# IPs or networks (x.x.x.x/24) allowed to read
readIps: []