实时通讯，摄像头作为设备端推流， app 作为用户端拉流，推拉流协议对比

比如拿 SRS 开源搭一套，就简单支持 rtmp ，或者 webrtc ，也都可以支持，再加一些边缘服务器部署了当转码之类的，畅所欲言。

Live streams can be published to the server with:

|protocol|variants|video codecs|audio codecs|
|--------|--------|------------|------------|
|[SRT clients](#srt-clients)||H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
|[SRT cameras and servers](#srt-cameras-and-servers)||H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
|[WebRTC clients](#webrtc-clients)|Browser-based, WHIP|AV1, VP9, VP8, H264|Opus, G722, G711 (PCMA, PCMU)|
|[WebRTC servers](#webrtc-servers)|WHEP|AV1, VP9, VP8, H264|Opus, G722, G711 (PCMA, PCMU)|
|[RTSP clients](#rtsp-clients)|UDP, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec|
|[RTSP cameras and servers](#rtsp-cameras-and-servers)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec|
|[RTMP clients](#rtmp-clients)|RTMP, RTMPS, Enhanced RTMP|AV1, VP9, H265, H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), G711 (PCMA, PCMU), LPCM|
|[RTMP cameras and servers](#rtmp-cameras-and-servers)|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
|[HLS cameras and servers](#hls-cameras-and-servers)|Low-Latency HLS, MP4-based HLS, legacy HLS|AV1, VP9, H265, H264|Opus, MPEG-4 Audio (AAC)|
|[UDP/MPEG-TS](#udpmpeg-ts)|Unicast, broadcast, multicast|H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
|[Raspberry Pi Cameras](#raspberry-pi-cameras)||H264||

And can be read from the server with:

|protocol|variants|video codecs|audio codecs|
|--------|--------|------------|------------|
|[SRT](#srt)||H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|
|[WebRTC](#webrtc)|Browser-based, WHEP|AV1, VP9, VP8, H264|Opus, G722, G711 (PCMA, PCMU)|
|[RTSP](#rtsp)|UDP, UDP-Multicast, TCP, RTSPS|AV1, VP9, VP8, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG and any RTP-compatible codec|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G726, G722, G711 (PCMA, PCMU), LPCM and any RTP-compatible codec|
|[RTMP](#rtmp)|RTMP, RTMPS, Enhanced RTMP|H264|MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3)|
|[HLS](#hls)|Low-Latency HLS, MP4-based HLS, legacy HLS|AV1, VP9, H265, H264|Opus, MPEG-4 Audio (AAC)|

And can be recorded and played back with:

|format|video codecs|audio codecs|
|------|------------|------------|
|[fMP4](#record-streams-to-disk)|AV1, VP9, H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video, M-JPEG|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3, G711 (PCMA, PCMU), LPCM|
|[MPEG-TS](#record-streams-to-disk)|H265, H264, MPEG-4 Video (H263, Xvid), MPEG-1/2 Video|Opus, MPEG-4 Audio (AAC), MPEG-1/2 Audio (MP3), AC-3|

ref: https://github.com/bluenviron/mediamtx/blob/main/README.md

个人选择：rtsp/webrtc 推流。hls 看

@NessajCN #2 我看看。之前只看了 zlm 和 srs

webrtc 是为了双向通信，视频和音频双向 webrtc 适合的视频会议，远程桌面分享，而这个在直播里面没有必要，并且做并发更困难。

直播里面主要是一对多广播音视频信息，并没有多少需要双向的需求。

搞清楚服务的定位先， 如果是监控摄像头，那么首选是 rtsp, 个人电脑摄像头或手机推流的确是 rtmp 。

@mightybruce #4 嗯，确实 webrtc 相关的 cpu 占用会比其他协议更大。但是国内目前摄像头推流阿里、小米都还是 rtmp ，rtsp 还是少。