Goodday everybody,
I have question about Nginx streaming software and the resteaming from other streams.
I wanne restream another stream (third party) with nginx because this way I wanne create so called thumbs from the (third party) stream and stream it on my own site .
The thumbs part is what I all know , the part of the restream is the unkown part for me .
Long story short now .
1. The stream link what I like restream is
http://154.57.145.83/flv/5285079c2c9e5/testat123.flv
As out going stream on my site I wanne have it like :
rtmp://145.44.194.308:1935/myapp/flv:test.flv
I have found this like on stackoverflow as well but it haven't help me out so far . (How to restream an udp live stream using nginx rtmp module?)
This is my code what I have used and dont seems to work .
exec_pull ffmpeg -i
http://154.57.145.83/flv/5285079c2c9e5/testat123.flv -c:v libx264 -c:a
libfaac -ar 44100 -ac 2 -f flv rtmp://145.44.194.308:1935/myapp/test;
So I hope someone here can help me out because I think other people will like to do same thing as well
Greatings and have yourself an great day
I have found the anwers for someone that wanne use it as well
exec_pull ffmpeg -re -i
http://154.57.145.83/flv/5285079c2c9e5/testat123.flv -c:a libmp3lame
-ar 44100 -ac 1 -f flv rtmp://415.493.196.188:1935/myapp/test 2>>/tmp/log/ffmpeg.log;
Related
We have an NGINX RMTP module installed and while testing the same we came to know that the bitrate for the output was a around 7Mbps irrespective of the input stream's bitrate and as we have a lot of people watching these streams I would like to know how to reduce the same to about 4Mbps for this module?
Also, does NGNIX's RTMP module support H.265 instead of the standard H.264 which can help set the bitrate to about 2Mbps.
You can transcode the incoming rtmp stream with ffmpeg with maxrate & b:v in this case you can control the maximum bitrate. Here is a simple example(for this example use another app show as well):
application live {
live on;
exec_push ffmpeg -i rtmp://localhost/$app/$name -async 1 -vsync -1
-c:a libfdk_aac -b:a 128k -c:v libx264 -b:v 2000k -maxrate 3000k -f flv -preset superfast -profile:v baseline rtmp://localhost:1935/show/$name_with_maxrate
}
I use nginx and ffmpeg to restream video from my provider. Previously I use ffmpeg with arguments where I reencoding video and reencoding audio, because my server is to slow I resigned from reencoding.
So now, I use that command :
ffmpeg -re -i http://link.somelink.com:6565/21d12d1/17233 -map 0 -c copy -bsf:a aac_adtstoasc -f flv -flvflags no_duration_filesize rtmp://test_ip/canal/stream
This works only when my provider streaming with aac audio codec, but sometimes my provider change audio codec to ac3. And then this doesn't work. I try something like this :
ffmpeg -thread_queue_size 32768 -re -i http://link.somelink.com:6565/21d12d1/17233 -c:v copy -c:a aac -f flv -flvflags no_duration_filesize rtmp://test_ip/canal/stream
And it all looks like it's all right in console with ffmpeg, but my restreaming video doesn't work. Ngnix throws 304 exception sometime.
Any suggestions?
Please help,
It's very important for me...
Ac3 is not in supported codecs list. You should encode your stream accordingly.
RTMP supports only a limited number of codecs. The most popular RTMP video codecs are H264, Sorenson-H263 (aka flv) and audio codecs AAC, MP3, Nellymoser, Speex. If your video is encoded with these codecs (the most common pair is H264/AAC) then you do not need any conversion. Otherwise you need to convert video to one of supported codecs.
https://github.com/arut/nginx-rtmp-module/wiki/Getting-started-with-nginx-rtmp
I want to take an HLS stream and transcode it to RTMP and simulcast it with the nginx RTMP module.
It's not working, however (I have it placed in the application section of the RTMP module).
exec ffmpeg -i -re http://<HLS>.m3u8 -acodec aac -vcodec libx264 -f flv rtmp://localhost/live/test;
When I try to view my RTMP stream in VLC, it is not loading. I have tried several variations of that ffmpeg directive, none have worked. Any advice? If you need to see more of my config file, I can provide that, but this server has been working previously perfectly when sending video over via a Teradek encoder. This new wrinkle is just not working.
EDIT: Just had a thought. It’d probably help to have the codec information of the incoming HLS stream. Here it is:
Video Codec: H264 - MPEG-4 AVC
Resolution: 640x360
Frame rate: 24
Decoded format: Planar 4:2:0 YUV
Audio Codec: MPEG AAC Audio (mp4a)
Channels: Stereo
Sample rate:48000Hz
If you run in terminal
ffmpeg -i -re http://<HLS>.m3u8 -acodec aac -vcodec libx264 -f flv rtmp://localhost/live/test;
are you able to play the stream in VLC?
I've been attempting to transcode a stream produced by obs studio to my nginx server and send it off to youtube. Now I've made it work with twitch and I know these settings are actually transcoding it mostly correctly and is viewable. The problem being that youtube live picks it up as Bad video settings and tells me to change the current video container format. The other side effect that is probly unrelated is the stream looks really poorly on youtube. Looks like it was streamed at a poor bitrate and stuff but the real problem is the bad video settings error.
The ffmpeg command being used is as follows
ffmpeg -i rtmp://localhost/Private/Private1 -vb 6000k -minrate 6000k -maxrate 6000k -bufsize 6000k -s 1280x720 -c:v libx264 -preset faster -r 50 -g 100 -keyint_min 50 -x264opts nal-hrd=cbr:force-cfr=1 -sws_flags lanczos -tune film -pix_fmt yuv420p -c:a copy -f flv -threads 6 -strict normal rtmp://a.rtmp.youtube.com/live2/{key}
I've tried with different framerates and been googling for awhile and found nothing or interpreted everything wrongly. Either way I would be very happy for some help here.
System info.
OS: Ubuntu Server 16.04 LTS
Ram: 10gb
Processor: AMD Phenom(tm) II X6 1090T
GPU: Geforce GT 520
Internet.
Upload 15mbit
Download 150mbit
If you need any more info I will gladly send it. Thanks for reading.
Edit 1
After some googling about what I'm doing wrong I decided to try and change stuff slightly and came up with this command
ffmpeg -re -i rtmp://localhost/(app)/(key) -c:v libx264 -r 50 -g 100 -keyint_min 100 -x264opts "keyint=100:min-keyint=100:no-scenecut" -sws_flags lanczos -profile:v baseline -preset veryfast -vb 6000K -minrate 6000k -maxrate 6000k -bufsize 6000k -s 1280x720 -tune film,zerolatency -pix_fmt yuv420p -f flv -c:a copy -ac 1 -strict normal rtmp://(output site)/(output app)/(output key)
which as of my current testing seems to at least have a healthy stream for longer than 2 minutes if i only output to youtube live directly. Ive found output to my nginx server then youtube live breaks things.
my nginx rtmp settings are on this link https://pastebin.com/siE99Tv8
Edit 2
If I push the stream to a site like restream to stream it to youtube then it seems to be working. tested for 25 minutes with no change of them saying bad video container or anything. So I'm going to say nginx is partly to blame in how its distributing the files? Unsure what I'm doing wrong. I am pretty sure ffmpeg isn't to blame here at least
Seems YouTube does not like nginx. I found two solutions for this.
Solution 1
Add "meta copy;" to you nginx config as follow:
rtmp {
server {
listen 1935;
application youtube{
live on;
meta copy;
push rtmp://a.rtmp.youtube.com/live2/(key);
}
}
}
Solution 2
Modify nginx-rtmp-module/ngx_rtmp_codec_module.c and replace the line:
ngx_string("Server"),
with
ngx_string("xtradata"),
then recompile nginx.
I'm trying to use Nginx to do live stream to combine two stream into one, so I need to spawn FFMpeg, like so
ffmpeg -i "rtmp://in/1" -i "rtmp://in/2" -filter_complex "overlay=70:50" -vcodec libx264 -preset ultrafast -f flv rtmp://out
However, is there a way to detect if one of the incoming stream drops, and I can continue the stream? From what I'm reading, it is not possible, and ffmpeg task will be killed.