[FFmpeg-trac] #7283(ffmpeg:new): Drops on the input when rtsp is relayed

FFmpeg trac at avcodec.org
Wed Jun 27 08:48:27 EEST 2018


#7283: Drops on the input when rtsp is relayed
-------------------------------------+-------------------------------------
             Reporter:  jidckii      |                     Type:  defect
               Status:  new          |                 Priority:  important
            Component:  ffmpeg       |                  Version:  git-
             Keywords:  rtsp rtp     |  master
  streaming                          |               Blocked By:
             Blocking:               |  Reproduced by developer:  0
Analyzed by developer:  0            |
-------------------------------------+-------------------------------------
 Summary of the bug:
 How to reproduce:
 {{{
 % ffmpeg -i input ... output
 ffmpeg version
 built on ...
 }}}
 Patches should be submitted to the ffmpeg-devel mailing list and not this
 bug tracker.

 Hello!
 I for a very long time I can not solve the problem of retransmission of
 live stream.
 According to my assumptions, this is the result of the input and output
 work in the same thread.
 Whatever buffers I set up in the end result, I'm getting early loss of
 packets when reordering RTP

 {{{RTP: misssed XX pacets}}}
 or errors like
 {{{jitter buffer is full}}}


 I know about the existence of the parameter async and cache
 https://ffmpeg.org/ffmpeg-all.html#async
 , but they can not be added to the rtsp input.

 I'm using the latest builds from git, assembled in the docker image from
 this repository:
 https://hub.docker.com/r/jrottenberg/ffmpeg/

 version at the time of this post:
 {{{
 ffmpeg -version
 ffmpeg version N-90807-g00099ef Copyright (c) 2000-2018 the FFmpeg
 developers
 built with gcc 5.4.0 (Ubuntu 5.4.0-6ubuntu1~16.04.9) 20160609
 configuration: --disable-debug --disable-doc --disable-ffplay --enable-
 shared --enable-avresample --enable-libopencore-amrnb --enable-
 libopencore-amrwb --enable-gpl --enable-libass --enable-libfreetype
 --enable-libvidstab --enable-libmp3lame --enable-libopenjpeg --enable-
 libopus --enable-libtheora --enable-libvorbis --enable-libvpx --enable-
 libx265 --enable-libxvid --enable-libx264 --enable-nonfree --enable-
 openssl --enable-libfdk_aac --enable-libkvazaar --enable-libaom --extra-
 libs=-lpthread --enable-postproc --enable-small --enable-version3 --extra-
 cflags=-I/opt/ffmpeg/include --extra-ldflags=-L/opt/ffmpeg/lib --extra-
 libs=-ldl --prefix=/opt/ffmpeg
 }}}
 Here are the startup parameters:

 {{{
 ffmpeg -v 48 -nostats -analyzeduration 20000000 -fflags igndts -fflags
 genpts -fflags latm
 -max_delay 500000 -reorder_queue_size 10000 -rtsp_transport udp
 -r 15 -i rtsp: admin: @ 192.168.86.169: 554/0? .sdp
 -map 0 -r 15 -c: v copy -an -f mpegts
 -muxdelay 1 -max_interleave_delta 0 -fflags + genpts
 udp: 239.0.0.1: 1234? ttl = 1? pkt_size = 1316
 }}}

 During the experiments, I changed the values ​​of the parameters,
 responsible for buffering, but whatever I put, the packets are still lost
 and the quality of the video stream spoils.

--
Ticket URL: <https://trac.ffmpeg.org/ticket/7283>
FFmpeg <https://ffmpeg.org>
FFmpeg issue tracker


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