WebRTC
Provides WebRTC output
Dependency on other protocol(s)
| Dependency | Description |
|---|---|
| HTTP(S) | Provides WebRTC output |
Playing WebRTC from MistServer
| Method | Type | URL |
|---|---|---|
| WebRTC with WebSocket signalling | webrtc | ws(s)://HOST:PORT/webrtc/STREAMNAME |
| WebRTC with WHEP signalling | whep | http(s)://HOST:PORT/webrtc/STREAMNAME |
WebRTC Optional configurations
| Option | Description | Type | Default | API | Commandline |
|---|---|---|---|---|---|
| UDP bind address (internal) | Interface address or hostname to bind SRTP UDP socket to. Defaults to originating interface address. | String | bindhost | --bindhost | |
| Certificate | (Root) certificate(s) file(s) to append to chain | String | cert | --cert | |
| debug | The debug level at which messages need to be printed. | debug | Inherited from parent process | debug | --debug |
| Default track sorting | What tracks are selected first when no specific track selector is used for playback. | = Default (last added for live, first added for VoD) bps_lth = Bit rate, low to high bps_htl = Bit rate, high to low id_lth = Track ID, low to high id_htl = Track ID, high to low res_lth = Resolution, low to high res_htl = Resolution, high to low | default_track_sorting | --default_track_sorting | |
| STUN/TURN config | An array of RTCIceServer objects, each describing one server which may be used by the ICE agent; these are typically STUN and/or TURN servers. These will be passed verbatim to the RTCPeerConnection constructor as the 'iceServers' property. | json | iceservers | --iceservers | |
| Write jitter log | Writes log of frame transmit jitter to /tmp/ for each outgoing connection | Flag | 0 | jitterlog | --jitterlog |
| Key | Private key for SSL | String | key | --key | |
| RTP lost timeout | Amount of packets any track will wait for a packet to arrive before considering it lost | Number (unsigned integer) | 30 | losttimeout | --losttimeout |
| RTP lost timeout (mobile) | Amount of packets any track will wait for a packet to arrive before considering it lost, on mobile connections | Number (unsigned integer) | 90 | losttimeoutmobile | --losttimeoutmobile |
| merge sessions | if enabled, merges together all views from a single user into a single combined session. if disabled, each view (reconnection of the signalling websocket) is a separate session. | Flag | 0 | mergesessions | --mergesessions |
| Disallow NACKs for viewers | Disallows viewers to send NACKs for lost packets | Flag | 0 | nackdisable | --nackdisable |
| RTP NACK timeout | Amount of packets any track will wait for a packet to arrive before NACKing it | Number (unsigned integer) | 5 | nacktimeout | --nacktimeout |
| RTP NACK timeout (mobile) | Amount of packets any track will wait for a packet to arrive before NACKing it, on mobile connections | Number (unsigned integer) | 15 | nacktimeoutmobile | --nacktimeoutmobile |
| Write packet log | Writes log of full packet contents to /tmp/ for each connection | Flag | 0 | packetlog | --packetlog |
| Preferred audio codecs | Comma separated list of audio codecs you want to support in preferred order. e.g. opus,ALAW,ULAW | String | opus,ALAW,ULAW | preferredaudiocodec | --webrtc-audio-codecs |
| Preferred video codecs | Comma separated list of video codecs you want to support in preferred order. e.g. H264,VP8 | String | H264,VP9,VP8 | preferredvideocodec | --webrtc-video-codecs |
| UDP bind address (public) | Interface address or hostname for clients to connect to. Defaults to internal address. | String | pubhost | --pubhost | |
| Username | Username to drop privileges to - default if unprovided means do not drop privileges | String | root | username | --username |