mediasoup 源码分析 （四）PlainTransport 分析

一、接收裸RTP流

PlainTransport 可以接收裸RTP流，也可以接收AES加密的RTP流。源码中提供了一个通过ffmpeg发送裸RTP流到mediasoup的脚本，具体地址为：mediasoup-demo/broadcasters/ffmpeg.sh

脚本就是通过HTTP Post发送创建PlainTranport请求，然后通过ffmpeg向指定地址+端口，发送RTP流

res=$(${HTTPIE_COMMAND} \
	POST ${SERVER_URL}/rooms/${ROOM_ID}/broadcasters/${BROADCASTER_ID}/transports \
	type="plain" \
	comedia:=true \
	rtcpMux:=false \
	2> /dev/null)

ffmpeg发送RTP流

#
# NOTES:
# - We can add ?pkt_size=1200 to each rtp:// URI to limit the max packet size
#   to 1200 bytes.
#
ffmpeg \
	-re \
	-v info \
	-stream_loop -1 \
	-i ${MEDIA_FILE} \
	-map 0:a:0 \
	-acodec libopus -ab 128k -ac 2 -ar 48000 \
	-map 0:v:0 \
	-pix_fmt yuv420p -c:v libvpx -b:v 1000k -deadline realtime -cpu-used 4 \
	-f tee \
	"[select=a:f=rtp:ssrc=${AUDIO_SSRC}:payload_type=${AUDIO_PT}]rtp://${audioTransport