ffmpeg推流到mediasoup测试

环境:

推流端: ffmpeg

server端: media-soup

接收端: chrome

设置：

export SERVER_URL=https://x.x.x.x:4443 //注意端口号是4443

export SERVER_URL=https://118.178.125.107:4443

export ROOM_ID=1; // 需要先在浏览器里边加入房间号1；否则推流会失败;

export MEDIA_FILE=./test.webM // 待推流的webM文件（webM用于封装opus和vp8）

1. chrome 默认不支持h264视频流播放（当然可以通过设置支持）

https://www.infoq.cn/article/2011/01/chrome-h264

ffmpeg 推流到media-soup, 接收端是google浏览器;

当推h264流时, 提示"http: warning: HTTP 500 unsupported codec [mimeType:video/h264, payloadType:101]"，不支持编码, h264这么通用的编码格式居然不支持, 于是怀疑media-soup配置可能有问题, 后来google搜索了一下, 发现原来是chrome放弃了对h264的支持, 转到自家的vp8/vp9系列;

另外, 自带的ffmpeg居然不支持libfaac, 感觉世界开始抛弃非开源项目了;

2. 推流格式: audio: opus video: vp8

不管什么格式的音视频, ffmpeg都能给它转码到 vp8/opus, 所以, 只要rtp参数填写为正确的opus和vp8 即可，在chrome里边也能正常显示;

其它： google支持的格式 opus对标aac, vp8对标h264, vp9对标h265;

ffmpeg.sh 是边转码, 边推流, 转码耗时本来就大, 可能会影响推流; 所以我们先转码为chrome能播的音视频流

需要转码为音频: opus, 视频: vp8, 由于mp4暂不支持vp8，所以需要用webM封装格式;

ffmpeg转码参考:

https://wiki.fiveyellowmice.com/wiki/%E5%A6%82%E4%BD%95%E4%BD%BF%E7%94%A8_FFmpeg_%E8%BF%9B%E8%A1%8C%E8%A7%86%E9%A2%91%E8%BD%AC%E7%A0%81:%E8%BD%AC%E7%A0%81%E6%97%B6%E8%83%BD%E9%A1%BA%E4%BE%BF%E4%B8%80%E8%B5%B7%E5%81%9A%E7%9A%84%E4%BA%8B%E6%83%85

视频缩放和裁剪参考:

https://amoskong.wordpress.com/2014/09/29/%E8%BD%AC-%E4%BD%BF%E7%94%A8-ffmpeg-%E7%BC%A9%E6%94%BE%E3%80%81%E8%A3%81%E5%89%AA%E3%80%81%E5%89%AA%E8%BE%91%E8%A7%86%E9%A2%91/

由于需要适配浏览器播放窗口大小和最大码率大小, 故将原有的1080p视频缩放为200p, 然后再裁剪到适合浏览器窗口播放的大小; 视频码率降低到300kbps，码率太大容易卡顿. 因为推流端没有做码率自适应;

ffmpeg -i mj-test.mp4 -c:a libopus -ac 2 -ar 48000 -pix_fmt yuv420p -vf scale=355:200,crop=200:170:0:0 -c:v libvpx -b:v 300k -deadline realtime o.webM

ffmpeg推流脚本为:

ffmpeg \

-re \

-v info \

-stream_loop -1 \

-i ${MEDIA_FILE} \

-map 0:a:0 \

-acodec copy \

-map 0:v:0 \

-c:v copy\

-f tee \

"[select=a:f=rtp:ssrc=${AUDIO_SSRC}:payload_type=${AUDIO_PT}]rtp://${audioTransportIp}:${audioTransportPort}?rtcpport=${audioTransportRtcpPort}|[select=v:f=rtp:ssrc=${VIDEO_SSRC}:payload_type=${VIDEO_PT}]rtp://${videoTransportIp}:${videoTransportPort}?rtcpport=${videoTransportRtcpPort}"

此处的audio和video都不再转码.

最终效果为:

ffmpeg 推流与mediasoup server交互流程:

1. 校验roomid是否存在(也可以自己创建)

verifying that room '1' exists...

+ http --check-status --verify=no GET https://118.178.125.107:4443/rooms/1

2. 请求server端创建广播

>>> creating Broadcaster...

+ http --check-status --verify=no POST https://118.178.125.107:4443/rooms/1/broadcasters id=fL4KvJdi908zgfcI6UgP3ryyqXtVFfS0 displayName=Broadcaster 'device:={"name": "FFmpeg"}'

3. 请求server端创建audio 传输

>>> creating mediasoup PlainTransport for producing audio...

++ http --check-status --verify=no POST https://118.178.125.107:4443/rooms/1/broadcasters/fL4KvJdi908zgfcI6UgP3ryyqXtVFfS0/transports type=plain comedia:=true rtcpMux:=false

ack:

res='{"id":"c871abab-1f47-4ddb-b3a7-0074733276a4","ip":"118.178.125.107","port":40001,"rtcpPort":40006}'

++ audioTransportId=c871abab-1f47-4ddb-b3a7-0074733276a4

++ audioTransportIp=118.178.125.107

++ audioTransportPort=40001

++ audioTransportRtcpPort=40006

4. 请求server端创建video 传输

>>> creating mediasoup PlainTransport for producing video...

++ http --check-status --verify=no POST https://118.178.125.107:4443/rooms/1/broadcasters/fL4KvJdi908zgfcI6UgP3ryyqXtVFfS0/transports type=plain comedia:=true rtcpMux:=false

ack:

+ res='{"id":"b3069d51-7a4b-406e-862d-b1356801fe8a","ip":"118.178.125.107","port":40002,"rtcpPort":40018}'

++ videoTransportId=b3069d51-7a4b-406e-862d-b1356801fe8a

++ videoTransportIp=118.178.125.107

++ videoTransportPort=40002

++ videoTransportRtcpPort=40018

5. 请求server端创建audio Producer

>>> creating mediasoup audio Producer...

+ http --check-status --verify=no -v POST https://118.178.125.107:4443/rooms/1/broadcasters/fL4KvJdi908zgfcI6UgP3ryyqXtVFfS0/transports/c871abab-1f47-4ddb-b3a7-0074733276a4/producers kind=audio 'rtpParameters:={ "codecs": [{ "mimeType":"audio/opus", "payloadType":100, "clockRate":48000, "channels":2, "parameters":{ "sprop-stereo":1 } }], "encodings": [{ "ssrc":1111 }] }'

6. 请求server端创建video Producer

>>> creating mediasoup video Producer...

+ http --check-status --verify=no -v POST https://118.178.125.107:4443/rooms/1/broadcasters/fL4KvJdi908zgfcI6UgP3ryyqXtVFfS0/transports/b3069d51-7a4b-406e-862d-b1356801fe8a/producers kind=video 'rtpParameters:={ "codecs": [{ "mimeType":"video/vp8", "payloadType":101, "clockRate":90000 }], "encodings": [{ "ssrc":2222 }] }'

7. 运行ffmpeg向server端指定rtp端口发送rtp/rtsp数据

>>> running ffmpeg...

+ ffmpeg -re -v info -stream_loop -1 -i ./test.webM -map 0:a:0 -acodec copy -map 0:v:0 -c:v copy -f tee '[select=a:f=rtp:ssrc=1111:payload_type=100]rtp://118.178.125.107:40001?rtcpport=40006|[select=v:f=rtp:ssrc=2222:payload_type=101]rtp://118.178.125.107:40002?rtcpport=40018'

8. 程序退出的时候自动删除前边创建的broadcaster

+ http --check-status --verify=no DELETE https://118.178.125.107:4443/rooms/1/broadcasters/fL4KvJdi908zgfcI6UgP3ryyqXtVFfS0

遇到的问题：

./ffmpeg.sh

>>> verifying that room '1' exists...

http: error: SSLError: HTTPSConnectionPool(host='118.178.125.107', port=4443): Max retries exceeded with url: /rooms/1 (Caused by SSLError(SSLError(1, '[SSL: CERTIFICATE_VERIFY_FAILED] certificate verify failed (_ssl.c:852)'),)) while doing GET request to URL: https://118.178.125.107:4443/rooms/1

解决:

HTTPIE_COMMAND="http --check-status --verify=no" 后边加上--verify=no 不校验Https

其他问题:

chrome 可以通过设置支持h264, 但是webrtc支持h264需要修改源码, 还是比较麻烦, 目前先能看到效果就好, 至于什么编码, 并不是最关心的, 当然后续上产品会修改代码并支持h264;