Since WCS build 5.2.1504 a stream can be played from server via WebRTC in a number of video qualities like HLS ABR. A browser will play a first available quality track, then playing client can switch between qualities if necessary.
Since builds WCS 5.2.1663 and SFU SDK 2.0.231 player may switch between available ABR qualities automatically when playback channel state is changing. Automatic quality switching is based on WebRTC playback statistics data.
Chrome | Firefox | Safari | Chromium Edge | |
---|---|---|---|---|
Windows | + | + | + | |
Linux | + | + | + | |
Mac OS | + | + | + | + |
Android | + | + | + | |
iOS | + (iOS 14.4) | + (iOS 14.4) | + |
A streams published by all of the available ways can be played via WebRTC ABR:
Any stream will be transcoded to H264+Opus to be played as WebRTC ABR.
SFU functions with Simulcast are used to implement WebRTC ABR support. A stream published video track is transcoded to a number of H264 video tracks with a different parameters, and the tracks are sent to client as SFU qualities. Audio track is transcoded to Opus and is also sent to client as SFU track. An SFU room named as stream published is created on server because room is a main SFU object.
The following should be done to play a stream published on WCS using SFU Simulcast:
codecs_exclude_sfu=alaw,ulaw,g729,speex16,g722,mpeg4-generic,telephone-event,flv,mpv,vp8,h265 profiles=42e01f,640028 wcs_sfu_bridge_enabled=true |
WCS must be restarted to apply the settings.
WebRTC ABR quality profiles are set in /usr/local/FlashphonerWebCallServer/conf/wcs_sfu_bridge_profiles.yml
file. By default, the following profiles are used:
profiles: s : width : 320 height : 240 bitrate : 500 gop : 60 fps : 30 m : width : 640 height : 480 bitrate : 800 gop : 60 fps : 30 h : width : 960 height : 720 bitrate : 1300 gop : 60 fps : 30 |
The following profile parameters are supported:
height
- picture height (mandatory)width
- picture widthbitrate
- encoding bitrate in kbpsgop
- group of frames sizefps
- frames number per secondThe gop
parameter actually sets key frames encoding period. In the example above, for 30 frames per second a key frame will be formed every 2 seconds.
Quality profiles may be arbitrary named, they are used to choose a quality at client side. In the example above, the profiles may be named as 240p
, 480p
и 720p
respectively.
WCS must be restarted to apply the quality profiles settings.
Since build 5.2.1665, if video quality profiles are equal for HLS ABR and WebRTC ABR configurations, the same encoders will be used. For example, with WebRTC ABR setup
profiles: 240p: height: 240 bitrate: 500 codec: h264 gop: 60 fps: 30 480p: height: 480 bitrate: 1000 codec: h264 gop: 60 fps: 30 720p: height: 720 bitrate: 1500 codec: h264 gop: 60 fps: 30 |
and HLS ABR setup
profiles: -240p: audio: codec: mpeg4-generic rate: 48000 channels: 2 groupId: audio video: height: 240 bitrate: 500 codec: h264 gop: 60 fps: 30 audioGroupId: audio -480p: audio: codec: mpeg4-generic rate: 48000 channels: 2 groupId: audio video: width: 0 height: 480 bitrate: 1000 codec: h264 gop: 60 fps: 30 audioGroupId: audio -720p: audio: codec: mpeg4-generic rate: 48000 channels: 2 groupId: audio video: width: 0 height: 720 bitrate: 1500 codec: h264 gop: 60 fps: 30 audioGroupId: audio |
only 3 video encoders will be created
The following video profile parameters should be equal if they are set:
Since build 5.2.1663, video quality profiles will be sorted in the order set in /usr/local/FlashphonerWebCallServer/conf/wcs_sfu_bridge_profiles.yml
file. For example, if the profiles described like this
profiles: 240: height: 240 bitrate: 500 codec: h264 gop: 60 fps: 30 480: height: 480 bitrate: 1000 codec: h264 gop: 60 fps: 30 720: height: 720 bitrate: 1500 codec: h264 gop: 60 fps: 30 1080: height: 1080 bitrate: 3000 codec: h264 gop: 60 fps: 30 |
a client will receive a profiles list ordered like
240, 480, 720, 1080 |
If there are two profiles with the same name in the setup, an undefined behavior occurs. To resolve it, server will use only the last profile with the same name.
To reduce a server load while video encoding, since WCS build 5.2.1840 it is possible to transcode a maximum ABR quality (which is usually the original stream resolution and bitrate) only if there are B-frames in a source stream. The feature may be enabled by the following parameter
h264_b_frames_force_transcoding=true |
In this case the server will detect B-frames in a stream analizing a certain frames count (10 by default)
frame_cnt_to_determine_their_type=10 |
If there are B-frames in the stream, the maximum ABR quality will be transcoded and will be available for playback.
If there are no B-frames in the stream, the maximum ABR quality will not be transcoded. The original quality should be requested separately from a playing client.
1. For test we use:
https://test1.flashphoner.com:8444/client2/sfu/webrtc-abr-player/player.html
to play the stream2. Publish RTMP stream test
720p 30 fps with 2000 kbps bitrate
3. Open WebRTC ABR Player example page, set Stream name
to test
, and click Play
. Stream playback will start
1. Not all the qualities may be available to a client depending on channel bandwidth. In this case, unavailable quality buttons in WebRTC ABR Player will be displayed in red.
2. WebRTC ABR increases server CPU load because a number of video encoders are created per every published stream depending on quality profiles count.
3. In iOS Safari audio is muted by default in WebRTC ABR Player example, and unmute button is dispalyed on audio tag because iOS Safari requires a user action to enable sound