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目前研究大致有幾種做法:
協定 | 延遲 | 客戶端軟體 | 伺服器軟體 | 伺服器廣播流量需求 |
---|---|---|---|---|
HTTP | 7~20 sec | 所有瀏覽器 | Icecast | client數 × bitrate |
WebRTC | 即時(?) | 主要瀏覽器 | NodeJS/Janus (見下方) | client數×bitrate |
RTP/RTSP unicast | 即時(?) | VLC, Gstreamer, ffmpeg | VLC, GStreamer, ffmpeg | client數×bitrate |
RTP/RTSP multicast | 2 sec | VLC, Gstreamer, ffmpeg | VLC, GStreamer, ffmpeg | 1×bitrate |
On MacOS Server
ffmpeg -f avfoundation -i ":1" -acodec libmp3lame -ab 32k -ac 1 -f rtp rtp://239.0.0.10:12345
Make sure that the multicast packets are sent to the interface that is connected to the same subnet with the clients.
sudo route add -host 239.0.0.10 -interface en0
Client
/Applications/VLC.app/Contents/MacOS/VLC -vvv rtp://@239.0.0.10:12345
RTMP is a half propriety protocol, previously used by Flash http://letzgro.net/blog/the-popularity-of-rtmp-protocol-keeps-growing/
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