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# Enable publishing and reading streams with the WebRTC protocol.
webrtc: yes
# Address of the WebRTC HTTP listener.
webrtcAddress: :8889
# Enable TLS/HTTPS on the WebRTC server.
webrtcEncryption: no
# Path to the server key.
# This can be generated with:
# openssl genrsa -out server.key 2048
# openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
webrtcServerKey: server.key
# Path to the server certificate.
webrtcServerCert: server.crt
# Value of the Access-Control-Allow-Origin header provided in every HTTP response.
# This allows to play the WebRTC stream from an external website.
webrtcAllowOrigin: '*'
# List of IPs or CIDRs of proxies placed before the WebRTC server.
# If the server receives a request from one of these entries, IP in logs
# will be taken from the X-Forwarded-For header.
webrtcTrustedProxies: []
# Address of a local UDP listener that will receive connections.
# Use a blank string to disable.
webrtcLocalUDPAddress: :8189
# Address of a local TCP listener that will receive connections.
# This is disabled by default since TCP is less efficient than UDP and
# introduces a progressive delay when network is congested.
webrtcLocalTCPAddress: ''
# WebRTC clients need to know the IP of the server.
# Gather IPs from interfaces and send them to clients.
webrtcIPsFromInterfaces: yes
# List of interfaces whose IPs will be sent to clients.
# An empty value means to use all available interfaces.
webrtcIPsFromInterfacesList: []
# List of additional hosts or IPs to send to clients.
webrtcAdditionalHosts: ["buro.example", "mediamtx.buro.example"]
# ICE servers. Needed only when local listeners can't be reached by clients.
# STUN servers allows to obtain and share the public IP of the server.
# TURN/TURNS servers forces all traffic through them.
webrtcICEServers2: []
# - 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.
# username: ''
# password: ''
# clientOnly: false
# Time to wait for the WebRTC handshake to complete.
webrtcHandshakeTimeout: 10s
# Maximum time to gather video tracks.
webrtcTrackGatherTimeout: 2s
# The maximum time to gather STUN candidates.
webrtcSTUNGatherTimeout: 5s
Server logs
No response
Network dump
No response
The text was updated successfully, but these errors were encountered:
Which version are you using?
Which operating system are you using?
Linux amd64 Docker
Describe how to replicate the issue
i am trying to have
js browser client (vue app) WHIP -> caddy (reverse proxy) -> mediamtx
but i continously get cors errors
has anybody succeded in publishing whip to mediamtx behind a reverse proxy?
caddyfile:
mediamtx config
Server logs
No response
Network dump
No response
The text was updated successfully, but these errors were encountered: