changed 7 years ago
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Conference Interpretation (Audio broadcasting)

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

WebRTC

References - readings

Reference - relevant projects

RTSP/RTP

multicast

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

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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