Support reading with WebRTC

[aler9]

Fixes #566

Usage

1. download a nightly release or build from source:

   rtsp-simple-server_v0.20.2-9-g9399f6a_darwin_amd64.tar.gz rtsp-simple-server_v0.20.2-9-g9399f6a_linux_amd64.tar.gz rtsp-simple-server_v0.20.2-9-g9399f6a_linux_arm64v8.tar.gz rtsp-simple-server_v0.20.2-9-g9399f6a_linux_armv6.tar.gz rtsp-simple-server_v0.20.2-9-g9399f6a_linux_armv7.tar.gz rtsp-simple-server_v0.20.2-9-g9399f6a_windows_amd64.zip

2. get or generate a TLS certificate:

   openssl genrsa -out server.key 2048
   openssl req -new -x509 -sha256 -key server.key -out server.crt -days 3650
   

3. start the server and publish something

4. visit `https://server-host:8889/path-of-the-stream

Configuration

# Disable support for the WebRTC protocol.
webrtcDisable: no
# Address of the WebRTC listener.
webrtcAddress: :8889
# Path to the server key. This is mandatory since HTTPS is mandatory in order to use WebRTC.
# 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: []
# List of STUN servers. These are used in WebRTC to get the public IP of both server and clients.
webrtcStunServers: [stun.l.google.com:19302]

Roadmap

• [x] read H264 tracks
• [ ] read Opus tracks
• [ ] read PCMA tracks
• [ ] read PCMU tracks
• [ ] read VP8 tracks
• [ ] read VP9 tracks
• [ ] authentication
• [ ] API
• [ ] metrics
• [ ] TURN servers
• [ ] publish video/audio from a browser
• [ ] find a way to generate and store a TLS certificate if the default one is missing

[brunobaraobr]

Hello, first I would like to thank you for the beautiful work you have done, it is simply incredible. I would like to know in webrtc is it possible to add support to H265 in WS

[aler9]

@brunobaraobr streaming H265 to the browser will be possible through Low-Latency HLS, please follow #469

Unfortunately WebRTC doesn't support H265 except when it's used inside macOS Safari, therefore at the moment it wouldn't be much useful to implement it.

[codecov-commenter]

Codecov Report

Merging #1242 (814ff45) into main (478607a) will decrease coverage by 4.12%. The diff coverage is 17.25%.

@@            Coverage Diff             @@
##             main    #1242      +/-   ##
==========================================
- Coverage   65.10%   60.98%   -4.13%     
==========================================
  Files         110      113       +3     
  Lines       11033    11805     +772     
==========================================
+ Hits         7183     7199      +16     
- Misses       3279     4033     +754     
- Partials      571      573       +2     

Impacted Files	Coverage Δ
internal/conf/authmethod.go	0.00% <ø> (ø)
internal/conf/credential.go	0.00% <ø> (ø)
internal/conf/encryption.go	0.00% <ø> (ø)
internal/conf/hlsvariant.go	0.00% <ø> (ø)
internal/conf/ipsorcidrs.go	0.00% <ø> (ø)
internal/conf/logdestination.go	0.00% <ø> (ø)
internal/conf/loglevel.go	18.75% <ø> (ø)
internal/conf/protocol.go	30.00% <ø> (ø)
internal/conf/sourceprotocol.go	0.00% <ø> (ø)
internal/conf/stringduration.go	50.00% <ø> (ø)
... and 24 more

:mega: We’re building smart automated test selection to slash your CI/CD build times. Learn more