Real time livestreaming - RPI FFmpeg and H5 Player - ffmpeg

Question

I work at a telehealth company and we are using connected medical devices in order to provide the doctor with real time information from these equipements, the equipements are used by a trained health Professional.
Those devices work with video and audio. Right now, we are using them with peerjs (so peer to peer connection) but we are trying to move away from that and have a RPI with his only job to stream data (so streaming audio and video).
Because the equipements are supposed to be used with instructions from a doctor we need the doctor to receive the data in real time.
But we also need the trained health professional to see what he is doing (so we need a local feed from the equipement)
How do we capture audio and video
We are using ffmpeg with a go client that is in charge of managing the ffmpeg clients and stream them to a SRS server.
This works but we are having a 2-3 sec delay when streaming the data. (rtmp from ffmpeg and flv on the front end)
ffmpeg settings :
("ffmpeg", "-f", "v4l2", `-i`, "*/video0", "-f", "flv", "-vcodec", "libx264", "-x264opts", "keyint=15", "-preset", "ultrafast", "-tune", "zerolatency", "-fflags", "nobuffer", "-b:a", "160k", "-threads", "0", "-g", "0", "rtmp://srs-url")
My questions
Is there a way for this set up to achieve low latency (<1 sec) (for the nurse and for the doctor) ?
Is the way I want to achieve this good ? Is there a batter way ?
Flow schema
Data exchange and use case flow:
Note: The nurse and doctor use HTTP-FLV to play the live stream, for low latency.

Answer 1

In your scenario, the latency is introduced by two parts:
The audio/video encoding by FFmpeg in RPI.
The player to consume and ingest the live stream.
FFmpeg in RPI
I noticed that you have already set some args, you could see full help by ffmpeg --help full to check these params.
The keyint equals to -g, so please remove keyint, and set the fps(-r). Please set -r 15 -g 15 which set the gop to 1s or 15fps:
-g <int> set the group of picture (GOP) size (from INT_MIN to INT_MAX) (default 12)
-r rate set frame rate (Hz value, fraction or abbreviation)
The x264 options preset and tune is useful for low latency, but also need to set another one profile to turn off bframe. Please set to -profile baseline -preset ultrafast -tune zerolatency for lower latency:
-preset <string> Set the encoding preset (cf. x264 --fullhelp) (default "medium")
-tune <string> Tune the encoding params (cf. x264 --fullhelp)
-profile <string> Set profile restrictions (cf. x264 --fullhelp)
You set a wrong -fflags nobuffer which is for decoder(player), instead you should use -fflags flush_packets for encoder:
-fflags <flags> (default autobsf)
flush_packets E.......... reduce the latency by flushing out packets immediately
nobuffer .D......... reduce the latency introduced by optional buffering
Note that the E means encoder while D means decoder/player.
The cli for FFmpeg, please covert to your params:
-vcodec libx264 \
-r 15 -g 15 \
-profile baseline -preset ultrafast -tune zerolatency \
-fflags flush_packets
However, I think these settings only works when you change your player settings, because the bottleneck is in the player now(latency 1~3s).
Player
For HTTP-FLV, please use conf/realtime.conf for SRS server, and please use ffplay to test the latency:
ffplay -fflags nobuffer -flags low_delay -i "http://your_server/live/stream.flv"
I think the latency should be <1s, better than H5 player, which uses MSE. You could compare the latency of them.
However, you couldn't let your users to use ffplay, it's test only for development. So we must use a low latency H5 player, that is WebRTC.
Please config SRS with conf/rtmp2rtc.conf which allows you to publish by FFmpeg by RTMP in low latency, and play the stream by WebRTC.
When your SRS is started, there is a WebRTC player, for example: http://localhost:8080/players/rtc_player.html and please read more about WebRTC from here
The url is very similar:
RTMP: rtmp://ip/live/livestream
FLV: http://ip/live/livestream.flv
HLS: http://ip/live/livestream.m3u8
WebRTC: webrtc://ip/live/livestream
If you use WebRTC player, the latency should be ~500ms and very stable.