Questions[rtsp](server)

I use ffmpeg + nginx/rtmp to restream video from ip cam to web page. The command I use is:

ffmpeg -rtsp_transport tcp -use_wallclock_as_timestamps 1 -i rtsp://admin:[email protected]/ -f flv -c:v copy -c:a copy rtmp://hls.mydomain.com/show/120

Then I use this URL to watch it in chrome:

http://hls.mydomain.com:8080/hls/120.m3u8

The problem is that the video in chrome is very choppy and can barely last only a few minutes before it stops.

On the re-streming server, I saw that ffmpeg quit automatically, without any error output:

ffmpeg version 4.1.6-1~deb10u1 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 8 (Debian 8.3.0-6)
  configuration: --prefix=/usr --extra-version='1~deb10u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libx264 --enable-shared
  libavutil      56. 22.100 / 56. 22.100
  libavcodec     58. 35.100 / 58. 35.100
  libavformat    58. 20.100 / 58. 20.100
  libavdevice    58.  5.100 / 58.  5.100
  libavfilter     7. 40.101 /  7. 40.101
  libavresample   4.  0.  0 /  4.  0.  0
  libswscale      5.  3.100 /  5.  3.100
  libswresample   3.  3.100 /  3.  3.100
  libpostproc    55.  3.100 / 55.  3.100
Input #0, rtsp, from 'rtsp://admin:[email protected]/':
  Metadata:
    title           : SDP Descrption
    comment         : SDP Description
  Duration: N/A, start: 1604569309.322911, bitrate: N/A
    Stream #0:0: Video: h264 (Baseline), yuvj420p(pc, bt709, progressive), 1920x1080, 25 fps, 25 tbr, 90k tbn, 50 tbc
    Stream #0:1: Audio: aac (LC), 48000 Hz, mono, fltp
Output #0, flv, to 'rtmp://hls.mydomain.com/show/120':
  Metadata:
    title           : SDP Descrption
    comment         : SDP Description
    encoder         : Lavf58.20.100
    Stream #0:0: Video: h264 (Baseline) ([7][0][0][0] / 0x0007), yuvj420p(pc, bt709, progressive), 1920x1080, q=2-31, 25 fps, 25 tbr, 1k tbn, 90k tbc
    Stream #0:1: Audio: aac (LC) ([10][0][0][0] / 0x000A), 48000 Hz, mono, fltp
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
[rtsp @ 0x5559c1d26940] CSeq 6 expected, 0 received.0:16.73 bitrate=2725.1kbits/s speed= 1.1x    
[rtsp @ 0x5559c1d26940] CSeq 6 expected, 0 received.0:21.50 bitrate=2447.1kbits/s speed=1.08x    
    Last message repeated 1 times
[rtsp @ 0x5559c1d26940] CSeq 6 expected, 0 received.0:35.10 bitrate=2191.8kbits/s speed=1.05x    
    Last message repeated 1 times
[rtsp @ 0x5559c1d26940] CSeq 6 expected, 0 received.0:39.04 bitrate=2180.5kbits/s speed=1.04x    
    Last message repeated 2 times
[flv @ 0x5559c1d30440] Failed to update header with correct duration.406.8kbits/s speed=1.03x    
[flv @ 0x5559c1d30440] Failed to update header with correct filesize.
frame= 1009 fps= 17 q=-1.0 Lsize=   17912kB time=00:01:00.28 bitrate=2434.0kbits/s speed=1.02x    
video:17401kB audio:459kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.288416%

I confirmed that the exit code of ffmpeg is 0. Streaming and serving are on two standalone Debian Linux server.

Any help is greatly appreciated. Also, I would like to know the command like arguments to re-stream H265 video. TIA.

I'm having an issue connecting my wowza server to an axis encoder via RTSP. It's an M7001 encoder with up to date firmware (5.20.1)

I'm using a connection of the format:

rtsp://root:password@IPADDRESS/media.amp

The exact same connection string works correctly from VLC media player. When I look at the stream codec information, it would seem there are two streams. Stream 0 - which is empty Stream 1 - which contains the actual video, and which plays properly with VLC

Wowza seems only to try and play stream 0, and therefore shows only a blank screen. Is there a parameter I can add to the URL force it to use stream 1?

I've attached a screenshot showing the stream infor from VLC.

How can I keep the flow (protocol rtsp, codec h264) in file (container mp4)? That is, on input an endless stream (with CCTV camera), and the output files in mp4 format size of 5-10 minutes of recording time.

OS: debian, ubuntu Software: vlc, ffmpeg (avconv)

Currently this scheme is used:

cvlc rtsp://admin:[email protected]:554/ch1-s1 --sout=file/ts:stream.ts
ffmpeg -i stream.ts -vcodec copy -f mp4 stream.mp4

But it can not record video continuously (between restarts vlc is a loss of about 10 seconds of live video)

So I've been bouncing between CDN's trying to optimize our serving solution. We're serving short video clips into our code that display either an html5 video element (MP4 or OGG) or a flash swf (FLV) as a fallback.

We're using Amazon S3 for the storage of these media files and CloudFront as our content delivery network. We've been experimenting with the flash fallback using CloudFront's Flash Media Server distribution and we're streaming the FLV into the Flash fallback. This works great :)

However, I'd like to accomplish the same setup (Origin > Streaming CDN > Client) with the MP4 and OGG files into the HTML5 placement. To do this, I need to stream them over RTP or RTSP, rather than CloudFront & FMS's RTMP.

So far I haven't had much luck finding a CDN that supports RTSP streams. Is there a way to use CloudFront's Streaming Server and somehow convert the stream protocol into RTSP using the mp4 and ogg as the source files?

(I should also note that price is a fairly big factor here. The project is sort of dependent on the price point on a per TB/GB/MB level.)

Is it possible to force the Apple Darwin Streaming Server to send streaming over port 80Is it possible to force the Apple Darwin Streaming Server to send streaming over port 80?

Background: We have a client that connects to our streaming server over a 3G/edge network. Which fails with a connection fail error. We can see the RTSP communitation between the terminal (handset) and the server when they use port 80 but it fails completely over all other ports (e.g. 554) The current hypothesis is that the 3G/edge operator blocks traffic on other ports then 80. Also the client reports that streaming from YouTubes Streaming Servers do work, I'm guessing YouTubes streaming server sends the streaming data over port 80.

So would it be possible to send streaming data from port 80 on the same (which probably would conflict with the RTSP server) or a "slave" server on port 80.