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cplpearce/ATAK-UAS-RTSP-README.md

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Created August 17, 2023 19:03
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rtsp-simple-server ATAK configuration (Linux)
Raw
ATAK-UAS-RTSP-README.md

Some users report having to comment out the following, your mileage may vary:

readBufferSize, runOnPublish, runOnPublishRestart, runOnRead, runOnReadRestart

ATAK users change the observer URL in UAS Tool to point to 554 (or whatever port you decide to use)

Download the latest rtsp-simple-server (0.17.13)

wget https://github.com/aler9/rtsp-simple-server/releases/download/v0.17.13/rtsp-simple-server_v0.17.13_linux_amd64.tar.gz

Extract the binary and a yaml config file

tar -zxvf rtsp-simple-server_v0.17.13_linux_amd64.tar.gz

Copy the binary to /usr/local/bin/

sudo cp rtsp-simple-server /usr/local/bin/rtsp-simple-server

Copy the configuration file to use with ATAK to /usr/local/etc

sudo curl https://gist.githubusercontent.com/curtishall/77b9dd3660511b7e173fbc4647ccfcb3/raw/be64945dcecad80f02e48c7bd763a1a7a3b77920/rtsp-simple-server.yml -o usr/local/etc/rtsp-simple-config.yml

Create a server file

sudo tee /etc/systemd/system/rtsp-simple-server.service >/dev/null << EOF
[Unit]
After=network.target
[Service]
ExecStart=/usr/local/bin/rtsp-simple-server /usr/local/etc/rtsp-simple-config.yml
[Install]
WantedBy=multi-user.target
EOF

Enable the newly created rtsp-simple-server service

sudo systemctl enable rtsp-simple-server

Start rtsp-simple-server and tail syslog to see how things look

sudo systemctl start rtsp-simple-server && tail -f /var/log/syslog

In ATAK UAS Tool use the following Network Preferences:

  • Video Broadcast Type: Wowza Video
  • Destination IP Address: i.e. - 192.168.86.232
  • Video Destination Port: 554
  • Use SSL: No
  • Video Broadcast Identifier: live/ATAK

(The live/ATAK can be changed, but make sure to not put an ending slash)

Then try to broadcast video, you should see this in /var/log/syslog:

Jan 12 23:55:41 rtsp-atak rtsp-simple-server[17053]: 2022/01/12 23:55:41 INF [RTSP] [conn 192.168.86.60:52250] opened Jan 12 23:55:41 rtsp-atak rtsp-simple-server[17053]: 2022/01/12 23:55:41 INF [RTSP] [session 343057601] opened by 192.168.86.60:52250 Jan 12 23:55:41 rtsp-atak rtsp-simple-server[17053]: 2022/01/12 23:55:41 INF [RTSP] [session 343057601] is publishing to path 'live/ATAK', 1 track with TCP

Then test with VLC -> Open Network Stream, adjust path name according:

rtsp://ipaddress:554/live/ATAK

HLS instructions coming soon.

Raw
rtsp-simple-server.yml
###############################################
# General parameters
# sets the verbosity of the program; available values are "error", "warn", "info", "debug".
logLevel: info
# destinations of log messages; available values are "stdout", "file" and "syslog".
logDestinations: [stdout]
# if "file" is in logDestinations, this is the file which will receive the logs.
logFile: /tmp/rtsp-simple-server.log
# timeout of read operations.
readTimeout: 10s
# timeout of write operations.
writeTimeout: 10s
# number of read buffers.
# a higher number allows a higher throughput,
# a lower number allows to save RAM.
readBufferCount: 512
# enable the HTTP API.
api: yes
# address of the API listener.
apiAddress: 0.0.0.0:9997
# enable Prometheus-compatible metrics.
metrics: no
# address of the metrics listener.
metricsAddress: 127.0.0.1:9998
# enable pprof-compatible endpoint to monitor performances.
pprof: no
# address of the pprof listener.
pprofAddress: 127.0.0.1:9999
# command to run when a client connects to the server.
# this is terminated with SIGINT when a client disconnects from the server.
# the server port is available in the RTSP_PORT variable.
runOnConnect:
# the restart parameter allows to restart the command if it exits suddenly.
runOnConnectRestart: no
###############################################
# RTSP parameters
# disable support for the RTSP protocol.
rtspDisable: no
# supported 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.
# UDP-multicast allows to save bandwidth when clients are all in the same LAN.
# TCP is the most versatile, and does support encryption.
# The handshake is always performed with TCP.
protocols: [tcp, udp]
# encrypt handshake and TCP streams with TLS (RTSPS).
# available values are "no", "strict", "optional".
encryption: "no"
# address of the TCP/RTSP listener. This is needed only when encryption is "no" or "optional".
rtspAddress: :554
# address of the TCP/TLS/RTSPS listener. This is needed only when encryption is "strict" or "optional".
rtspsAddress: :8555
# address of the UDP/RTP listener. This is needed only when "udp" is in protocols.
rtpAddress: :8000
# address of the UDP/RTCP listener. This is needed only when "udp" is in protocols.
rtcpAddress: :8001
# IP range of all UDP-multicast listeners. This is needed only when "multicast" is in protocols.
multicastIPRange: 224.1.0.0/16
# port of all UDP-multicast/RTP listeners. This is needed only when "multicast" is in protocols.
multicastRTPPort: 8002
# port of all UDP-multicast/RTCP listeners. This is needed only when "multicast" is in protocols.
multicastRTCPPort: 8003
# path to the server key. This is needed only when encryption is "strict" or "optional".
# this can be generated with:
# openssl genrsa -out server.key 2048
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
serverKey: server.key
# path to the server certificate. This is needed only when encryption is "strict" or "optional".
serverCert: server.crt
# authentication methods.
authMethods: [basic, digest]
# read buffer size.
# this doesn't influence throughput and shouldn't be touched unless the server
# reports errors about the buffer size.
readBufferSize: 2048
###############################################
# RTMP parameters
# disable support for the RTMP protocol.
rtmpDisable: no
# address of the RTMP listener.
rtmpAddress: :1935
###############################################
# HLS parameters
# disable support for the HLS protocol.
hlsDisable: no
# address of the HLS listener.
hlsAddress: :8888
# by default, HLS is generated only when requested by a user;
# this option allows to generate it always, avoiding an initial delay.
hlsAlwaysRemux: no
# number of HLS segments to generate.
# increasing segments allows more buffering,
# decreasing segments decreases latency.
hlsSegmentCount: 3
# minimum duration of each segment.
# the final segment duration is also influenced by the interval between IDR frames,
# since the server changes the segment duration to include at least a IDR frame in each one.
hlsSegmentDuration: 1s
# value of the Access-Control-Allow-Origin header provided in every HTTP response.
# This allows to play the HLS stream from an external website.
hlsAllowOrigin: '*'
###############################################
# Path parameters
# these settings are path-dependent.
# 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
# another entry.
paths:
all:
# source of the stream - this can be:
# * publisher -> the stream is published by a RTSP or RTMP client
# * rtsp://existing-url -> the stream is pulled from another RTSP server
# * rtsps://existing-url -> the stream is pulled from another RTSP server with RTSPS
# * rtmp://existing-url -> the stream is pulled from another RTMP server
# * 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
# * redirect -> the stream is provided by another path or server
source: publisher
# if the source is an RTSP or RTSPS URL, this is the protocol that will be used to
# pull the stream. available values are "automatic", "udp", "multicast", "tcp".
# the TCP protocol can help to overcome the error "no UDP packets received recently".
sourceProtocol: automatic
# if the source is an RTSP or RTSPS URL, this allows to support sources that
# don't provide server ports or use random server ports. This is a security issue
# and must be used only when interacting with sources that require it.
sourceAnyPortEnable: no
# if the source is a RTSPS or HTTPS URL, and the source certificate is self-signed
# or invalid, you can provide the fingerprint of the certificate in order to
# validate it anyway.
# the fingerprint can be obtained by running:
# openssl s_client -connect source_ip:source_port </dev/null 2>/dev/null | sed -n '/BEGIN/,/END/p' > server.crt
# openssl x509 -in server.crt -noout -fingerprint -sha256 | cut -d "=" -f2 | tr -d ':'
sourceFingerprint:
# if the source is an RTSP or RTMP URL, it will be pulled only when at least
# one reader is connected, saving bandwidth.
sourceOnDemand: no
# if sourceOnDemand is "yes", readers will be put on hold until the source is
# ready or until this amount of time has passed.
sourceOnDemandStartTimeout: 10s
# if sourceOnDemand is "yes", the source will be closed when there are no
# readers connected and this amount of time has passed.
sourceOnDemandCloseAfter: 10s
# if the source is "redirect", this is the RTSP URL which clients will be
# redirected to.
sourceRedirect:
# if the source is "publisher" and a client is publishing, do not allow another
# client to disconnect the former and publish in its place.
disablePublisherOverride: no
# if the source is "publisher" and 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:
# username required to publish.
# sha256-hashed values can be inserted with the "sha256:" prefix.
publishUser:
# password required to publish.
# sha256-hashed values can be inserted with the "sha256:" prefix.
publishPass:
# ips or networks (x.x.x.x/24) allowed to publish.
publishIPs: []
# username required to read.
# sha256-hashed values can be inserted with the "sha256:" prefix.
readUser:
# password required to read.
# sha256-hashed values can be inserted with the "sha256:" prefix.
readPass:
# ips or networks (x.x.x.x/24) allowed to read.
readIPs: []
# command to run when this path is initialized.
# this can be used to publish a stream and keep it always opened.
# this is terminated with SIGINT when the program closes.
# the path name is available in the RTSP_PATH variable.
# the server port is available in the RTSP_PORT variable.
runOnInit:
# the restart parameter allows to restart the command if it exits suddenly.
runOnInitRestart: no
# command to run when this path is requested.
# this can be used to publish a stream on demand.
# this is terminated with SIGINT when the path is not requested anymore.
# the path name is available in the RTSP_PATH variable.
# the server port is available in the RTSP_PORT variable.
runOnDemand:
# the restart parameter allows to restart the command if it exits suddenly.
runOnDemandRestart: no
# readers will be put on hold until the runOnDemand command starts publishing
# or until this amount of time has passed.
runOnDemandStartTimeout: 10s
# the runOnDemand command will be closed when there are no
# readers connected and this amount of time has passed.
runOnDemandCloseAfter: 10s
# command to run when a client starts publishing.
# this is terminated with SIGINT when a client stops publishing.
# the path name is available in the RTSP_PATH variable.
# the server port is available in the RTSP_PORT variable.
runOnPublish:
# the restart parameter allows to restart the command if it exits suddenly.
runOnPublishRestart: no
# command to run when a clients starts reading.
# this is terminated with SIGINT when a client stops reading.
# the path name is available in the RTSP_PATH variable.
# the server port is available in the RTSP_PORT variable.
runOnRead:
# the restart parameter allows to restart the command if it exits suddenly.
runOnReadRestart: no
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