v0.1-rc.9

• Add support environment vars for config #143
• Add support multiple configs
• Add support raw YAML config from cli
• Add auto build binaries on release by @skrashevich in #158
• Add config editor in Web UI by @skrashevich in #153
• Add version to Web UI
• Rewrite stream info in Web UI
• Rewrite WS+MP4 stream for iPhones
• Fix invalid tcpType for UDP candidate by @tsightler in #165 #166

v0.1-rc.8

Important change

WebRTC now uses single 8555 TCP and UDP port by default and for all connections. No need to define it in config. If you have public IP-address - its better to forward both TCP and UDP port.

Other changes

• Fix RTSP server memory leaks #163 @tsightler
• Fix Ivideon source memory leaks
• Fix WebSocket connections memory leaks
• Fix exec source memory leaks
• Fix MJPEG source memory leaks
• Fix RTSP tracks list in info json
• Fix RTSP session states handle
• Fix lots of active connections #156
• Fix empty streams list on stream lock
• Fix ffmpeg request to same stream (for audio transcoding)
• Fix multiple requests from different consumers

v0.1-rc.7

• Add support for simultaneous requests from different consumers
• Add producer url to logs
• Fix double start for RTSP source (possible fix for #47 #144)
• Fix STUN candidate in IPv6 format #148
• Fix wrong RTSP H264 profile for some cameras #155
• Fix GetMedias for producer in reconnect state #111
• Speedup container building using Golang cross-building #150 @skrashevich

v0.1-rc.6

• Add support hardware acceleration for transcoding (read more)
• Add support width and height params for FFmpeg
• Update JS player for integration with other soft
• Disable JS stream background by default
• Fix Firefox WebRTC support #120
• Fix RTSP H264 with two SEI in packet
• Fix RTSP MJPEG processing in some cases

v0.1-rc.5

Main changes

• New JS video player will automatically select best technology according on: codecs inside your stream, current browser capabilities and current network configuration
• New stream page with support multiple stream on page with automatic or manual technology selection #126
• Add built-in support HTTP-MJPEG and HTTP-JPEG (snapshots) sources

Other changes

• Add support auth for RTSP server #114
• Add feature to SETUP new RTSP tracks after PLAY #112
• Add support MP4 over WebSocket (for Apple browsers without MSE support)
• Add support MJPEG over WebSocket
• Update to go 1.19 for binaries
• Add binary for win arm64
• Fix binary for mac arm64
• Fix H265 for WebRTC in Safari
• Fix H265 for MSE in Safari on M1 CPU #121
• Fix memory leaks with MSE
• Fix WebSocket origin check with wrong port #118 #28
• Remove log warning on WebSocket disconnect

v0.1-rc.4

Many improvements with MSE and H265 codec support in different browsers

• Totally rewritten MSE player as separate JS component
• Add support codecs negotiation for MSE
• Add support autopause for MSE while browser or page not active
• Add support super modern MSE in Workers for Chrome 108+
• Auto enable audio for MSE if browser allows it
• Add support CORS for API
• Update about MSE H265 support in readme (adds Safari and Android Chrome)
• Fix H265 support for MSE in Safari
• Fix empty SPS for MSE H265 #99
• Fix WebRTC async connection (less delay on start from Web UI)
• Fix race (concurency) for Track processing
• Fix multiple transcoding when track not exists

v0.1-rc.3

• Add support AAC-ELD audio for HomeKit source
• Add ffmpeg async option (useful for HomeKit audio)
• Add support stream name as ffmpeg input
• Add support multiple transcoding for ffmpeg
• Add rotate option for ffmpeg source
• Add go2rtc version info #58
• Add User-Agent to go2rtc RTSP requests #106
• Add exit on error handler for exec source
• Timeout for exec source increased to 60 seconds
• Set video track for WebRTC always first #65
• Errors from exec will show with exec debug log
• Fix processing wrong data for RTSP source #47
• Fix support external RTSP producers
• Fix stop ffmpeg without matching tracks
• Fix and rewrite HomeKit source

Example. Stream name, multiple transcoding and async option for HomeKit:

sources:
  aqara_g3:
    - hass:Camera-Hub-G3-AB12
    - ffmpeg:aqara_g3#audio=aac#audio=opus#async

v0.1-rc.2

• Add support AAC audio for RTSP, RTMP, MSE, MP4
• Add duration API for MP4 file
• Update MSE stream on client and server side
• Update RTSP server filters #66

v0.1-rc.1

• Add reconnection feature for stream #39 #87
• Add support for a stream to link to itself #65
• Add API /api/frame.jpeg for MJPEG stream #62
• Add cache for public IP for 5 min
• Add 5 sec timeout to RTSP and FFmpeg RTSP
• Fix timeouts for receiving public IP (without Internet fix) #78
• Fix WriteRTP concurrent map iteration and map write #74
• Fix RemoveConsumer invalid memory address #31 #60
• Fix slice bounds out of range #44

v0.1-beta.10

• Add support HTTP-FLV
• Fix RTSP processing for Amcrest IP4M-1051 #35
• Fix backchannel reconnection issue #60
• Fix RemoveConsumer panic #31 #45