Sergey Krashevich
117 lines
5.1 KiB
Markdown
117 lines
5.1 KiB
Markdown
What you should know about WebRTC:
|
|
|
|
- It's almost always a **direct [peer-to-peer](https://en.wikipedia.org/wiki/Peer-to-peer) connection** from your browser to the go2rtc app
|
|
- When you use Home Assistant, Frigate, Nginx, Nabu Casa, Cloudflare, and other software, they are only **involved in establishing** the connection; they are **not involved in transferring** media data
|
|
- WebRTC media cannot be transferred inside an HTTP connection
|
|
- Usually, WebRTC uses random UDP ports on the client and server to establish a connection
|
|
- Usually, WebRTC uses public [STUN](https://en.wikipedia.org/wiki/STUN) servers to establish a connection outside the LAN; these servers are only needed to establish a connection and are not involved in data transfer
|
|
- Usually, WebRTC will automatically discover all of your local and public addresses and try to establish a connection
|
|
|
|
If an external connection via STUN is used:
|
|
|
|
- Uses [UDP hole punching](https://en.wikipedia.org/wiki/UDP_hole_punching) technology to bypass NAT even if you haven't opened your server to the world
|
|
- For about 20% of users, the technology will not work because of the [Symmetric NAT](https://tomchen.github.io/symmetric-nat-test/)
|
|
- UDP is not suitable for transmitting 2K and 4K high bit rate video over open networks because of the high loss rate:
|
|
- https://habr.com/ru/companies/flashphoner/articles/480006/
|
|
- https://www.youtube.com/watch?v=FXVg2ckuKfs
|
|
|
|
## Default config
|
|
|
|
```yaml
|
|
webrtc:
|
|
listen: ":8555"
|
|
ice_servers:
|
|
- urls: [ "stun:stun.l.google.com:19302" ]
|
|
```
|
|
|
|
## Config
|
|
|
|
**Important!** This example is not for copy/pasting!
|
|
|
|
```yaml
|
|
webrtc:
|
|
# fix local TCP or UDP or both ports for WebRTC media
|
|
listen: ":8555" # address of your local server
|
|
|
|
# add additional host candidates manually
|
|
# order is important, the first will have a higher priority
|
|
candidates:
|
|
- 216.58.210.174:8555 # if you have static public IP-address
|
|
- stun:8555 # if you have dynamic public IP-address
|
|
- home.duckdns.org:8555 # if you have domain
|
|
|
|
# add custom STUN and TURN servers
|
|
# use `ice_servers: []` to remove defaults and leave it empty
|
|
ice_servers:
|
|
- urls: [ stun:stun1.l.google.com:19302 ]
|
|
- urls: [ turn:123.123.123.123:3478 ]
|
|
username: your_user
|
|
credential: your_pass
|
|
|
|
# optional filter list for auto-discovery logic
|
|
# some settings only make sense if you don't specify a fixed UDP port
|
|
filters:
|
|
# list of host candidates from auto-discovery to be sent
|
|
# includes candidates from the `listen` option
|
|
# use `candidates: []` to remove all auto-discovery candidates
|
|
candidates: [ 192.168.1.123 ]
|
|
|
|
# enable localhost candidates
|
|
loopback: true
|
|
|
|
# list of network types to be used for the connection
|
|
# includes candidates from the `listen` option
|
|
networks: [ udp4, udp6, tcp4, tcp6 ]
|
|
|
|
# list of interfaces to be used for the connection
|
|
# includes interfaces from unspecified `listen` option (empty host)
|
|
interfaces: [ eno1 ]
|
|
|
|
# list of host IP addresses to be used for the connection
|
|
# includes IPs from unspecified `listen` option (empty host)
|
|
ips: [ 192.168.1.123 ]
|
|
|
|
# range for random UDP ports [min, max] to be used for connection
|
|
# not related to the `listen` option
|
|
udp_ports: [ 50000, 50100 ]
|
|
```
|
|
|
|
By default, go2rtc uses a **fixed TCP** port and **fixed UDP** ports for each **direct** WebRTC connection: `listen: ":8555"`.
|
|
|
|
You can set a **fixed TCP** port and a **random UDP** port for all connections: `listen: ":8555/tcp"`.
|
|
|
|
You can also disable the TCP port and leave only random UDP ports: `listen: ""`.
|
|
|
|
## Config filters
|
|
|
|
**Important!** By default, go2rtc excludes all Docker-like candidates (`172.16.0.0/12`). This cannot be disabled.
|
|
|
|
Filters allow you to exclude unnecessary candidates. Extra candidates don't make your connection worse or better. But the wrong filter settings can break everything. Skip this setting if you don't understand it.
|
|
|
|
For example, go2rtc is installed on the host system. And there are unnecessary interfaces. You can keep only the relevant via `interfaces` or `ips` options. You can also exclude IPv6 candidates if your server supports them but your home network does not.
|
|
|
|
```yaml
|
|
webrtc:
|
|
listen: ":8555/tcp" # use fixed TCP port and random UDP ports
|
|
filters:
|
|
ips: [ 192.168.1.2 ] # IP-address of your server
|
|
networks: [ udp4, tcp4 ] # skip IPv6, if it's not supported for you
|
|
```
|
|
|
|
For example, go2rtc is inside a closed Docker container (e.g. [Frigate](https://frigate.video/)). You shouldn't filter Docker interfaces; otherwise, go2rtc won't be able to connect anywhere. But you can filter the Docker candidates because no one can connect to them.
|
|
|
|
```yaml
|
|
webrtc:
|
|
listen: ":8555" # use fixed TCP and UDP ports
|
|
candidates: [ 192.168.1.2:8555 ] # add manual host candidate (use docker port forwarding)
|
|
```
|
|
|
|
## Useful links
|
|
|
|
- https://www.ietf.org/archive/id/draft-ietf-wish-whip-01.html
|
|
- https://www.ietf.org/id/draft-murillo-whep-01.html
|
|
- https://github.com/Glimesh/broadcast-box/
|
|
- https://github.com/obsproject/obs-studio/pull/7926
|
|
- https://misi.github.io/webrtc-c0d3l4b/
|
|
- https://github.com/webtorrent/webtorrent/blob/master/docs/faq.md
|