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• Setting up RTP on Nginx

  2 février 2021, par Swap

  I'm trying to use Janus Media Server to relay WebRTC streams to a particular RTP host/port, from where ffmpeg can pick it up as an input and convert it further to an rtmp stream, which can then be used to broadcast to various social media platforms (such as, YouTube, Twitch, Facebook, etc.)

  


  My inspiration for this has been the following blog - https://www.meetecho.com/blog/firefox-webrtc-youtube-kinda/

  


  Specifically, I'm trying to replicate the following architecture -

  


  

  


  And Janus, as per their documentation, has a very neat API for doing it -

  


  {&#xA;    "request" : "rtp_forward",&#xA;    "room" : <unique numeric="numeric" of="of" the="the" room="room" publisher="publisher" is="is" in="in">,&#xA;    "publisher_id" : <unique numeric="numeric" of="of" the="the" publisher="publisher" to="to" relay="relay" externally="externally">,&#xA;    "host" : "<host address="address" to="to" forward="forward" the="the" rtp="rtp" and="and" packets="packets">",&#xA;    "host_family" : "",&#xA;    "audio_port" : <port to="to" forward="forward" the="the" audio="audio" rtp="rtp" packets="packets">,&#xA;    "audio_ssrc" : <audio ssrc="ssrc" to="to" use="use" when="when" optional="optional">,&#xA;    "audio_pt" : <audio payload="payload" type="type" to="to" use="use" when="when" optional="optional">,&#xA;    "audio_rtcp_port" : <port to="to" contact="contact" receive="receive" audio="audio" rtcp="rtcp" feedback="feedback" from="from" the="the" and="and" currently="currently" unused="unused" for="for">,&#xA;    "video_port" : <port to="to" forward="forward" the="the" video="video" rtp="rtp" packets="packets">,&#xA;    "video_ssrc" : <video ssrc="ssrc" to="to" use="use" when="when" optional="optional">,&#xA;    "video_pt" : <video payload="payload" type="type" to="to" use="use" when="when" optional="optional">,&#xA;    "video_rtcp_port" : <port to="to" contact="contact" receive="receive" video="video" rtcp="rtcp" feedback="feedback" from="from" the="the" optional="optional">,&#xA;    "simulcast" : ,&#xA;    "video_port_2" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" port="port" to="to" forward="forward" the="the" video="video" rtp="rtp" packets="packets" from="from" second="second" substream="substream"></if>layer to>,&#xA;    "video_ssrc_2" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" ssrc="ssrc" to="to" use="use" the="the" second="second" substream="substream"></if>layer; optional>,&#xA;    "video_pt_2" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" payload="payload" type="type" to="to" use="use" the="the" second="second" substream="substream"></if>layer; optional>,&#xA;    "video_port_3" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" port="port" to="to" forward="forward" the="the" video="video" rtp="rtp" packets="packets" from="from" third="third" substream="substream"></if>layer to>,&#xA;    "video_ssrc_3" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" ssrc="ssrc" to="to" use="use" the="the" third="third" substream="substream"></if>layer; optional>,&#xA;    "video_pt_3" : <if simulcasting="simulcasting" and="and" forwarding="forwarding" each="each" video="video" payload="payload" type="type" to="to" use="use" the="the" third="third" substream="substream"></if>layer; optional>,&#xA;    "data_port" : <port to="to" forward="forward" the="the" messages="messages">,&#xA;    "srtp_suite" : <length of="of" authentication="authentication" tag="tag" or="or" optional="optional">,&#xA;    "srtp_crypto" : "<key to="to" use="use" as="as" crypto="crypto" encoded="encoded" key="key" in="in" optional="optional">"&#xA;}&#xA;</key></length></port></port></video></video></port></port></audio></audio></port></host></unique></unique>

  


  For this, I've setup a Nginx server, where I've also installed Janus and everything's been running smoothly so far. But I'm quite clueless as to how to setup my Nginx server so that it accepts RTP connections (which will be forwarded as RTMP using ffmpeg).

  


  Please guide me to any relevant resources that would help me achieve this. Thanks in advance !

  


• Q&A : An interview with Matomo founder, Matthieu Aubry

  20 novembre 2018, par Joselyn Khor — About, Community

  Hey everyone ! Joselyn here. As always the views of our community remain top of mind. So to make sure you guys know the thinking behind these new projects, we reached out to Matomo’s founder, Matthieu, to ask questions you might want answered. Please check it out below !

  Hi guys, it’s Matthieu ! Here to answer some questions about the rebrand and the future of Matomo and Innocraft.

  What’s upcoming ?

  We’ve been busy implementing our rebrand into all aspects of Matomo and there’s also our new website, which is launching today ! The new website will help people better understand what Matomo is and how they can benefit from using modern web analytics.

  Why was Matomo and Innocraft brought onto one website ?

  In the past the separation caused a bit of confusion so we’re taking this as a chance to unite both the business brand, Innocraft and community brand, Matomo, on one website. Putting our focus on one brand, Matomo, makes it easier for people to see us with fresh eyes. We have a community side as well as a business side and while the community is still incredibly important to us, we find we have a powerful analytics tool that is capable of helping businesses too.

  Is Matomo becoming commercial or turning corporate ?

  No. nothing is changing. Matomo is still an open-source project and community. Although we’ll have a pricing page and “start free trial” on the new website brought over from Innocraft.cloud, the Matomo community will still play the biggest part on the Matomo website. We have dedicated sections focused on Community and On-Premise.

  The rebrand exercise helped us gain a refreshed perspective. After reflecting on how far we’ve come, we can feel more confident about Matomo Analytics itself as a platform. We believe it’s a great chance to bring that confidence into the brand and vision. We are proud that it’s an awesome open-source platform and at the same time it’s also powerful as a tool for businesses.

  Why is there no ‘download for free’ button on the homepage ?

   

  
  We feel many users coming to the site will get confused about our hosting options (Cloud and On-Premise) which is something you don’t usually consider when choosing an analytics tool.

  The reason for us to not have that button is when people see a “download for free” button on the homepage next to a “try it for free” button, it creates confusion. For those who do choose to download Matomo often become confused when they are left with a .zip file unaware how to install it and the technical requirements of self-hosting. We feel presenting our users with the simplest installation option first will give them the best chance possible to try Matomo to its full potential, without cost.

  And you can still find the link to Download Matomo in the footer of each page.
  

   

  Is Matomo still free to download and have forever ?

  Absolutely. The free open-source download can be found on the On-Premise section of the website, or download Matomo here.

  Why is it important to have a business behind the project ?

  There’s the reality that we have to make money in order for the Matomo project to survive … and thrive. The reason we still need a business side (Innocraft) is to fund and sustain the Matomo project. Whenever people purchase premium features, this helps finance the development of Matomo for our community.

  Because of the business we’re able to continually maintain and develop Matomo for you guys as well as future users. For example, the next release Matomo 3.8.0 is already mostly developed and will bring lots of interesting features too, like the two-factor authentication, Brute Force Protection, failed tracking requests reporting, lots of JavaScript tracker improvements, a new total summary row below reports, and many more security fixes, bug fixes, and other new features.

  So we see a business being very helpful in supporting our open-source community. Without a business side, our free, open-source project would not be able to survive.

  How will you protect the Matomo project ?

  We’ve ensured the Matomo project will be protected for the future as we wish to turn it into a not-for-profit foundation.

  We’ve also got a safeguard where the open-source code will stay under a GPL license forever. This is so we can guarantee, that no matter what happens, the Matomo project itself will stay completely free software.

  Is there a way for people to help ?

  There are heaps of ways to help ! You can help other Matomo users in the forums, contribute to fixes on GitHub, leave a great review (e.g. alternativeTo), help look for bugs with our Security Bounty Programme or participate and spread the word about Matomo in our community social media pages – Mastodon, Facebook, Twitter. Telling your friends about us would be very helpful too !

  What’s planned for the future ?
  We’ve worked hard to become the #1 open-source analytics platform (1.4 million websites use Matomo today), but now we need to empower even more individuals and businesses to take back control of their own data.

  Showing our community that we have a powerful platform is crucial, but alongside that our values are what define us. User privacy is still of utmost importance and we’re here to make it known that power needs to rest in the hands of people and not large corporations.

  You can rest easy knowing you’re doing your part in using trustworthy and dependable tools. By joining many other companies who are growing this movement to decentralise the Internet, we can build a safer, online world together.

  Join this analytics revolution and let us know what you think about Matomo !

• ffmpeg issue when using it from kurento to rtmp stream

  15 juillet 2020, par Maxi

  I successfully managed to connect ffmpeg with an RTP endpoint of my kurento server. On many tries before I got a "Connection timed out" due my docker configuration. In my docker logs I see following now after RTP endpoint creation and starting ffmpeg :

  


  streamy-server_1  | 2020-07-15 08:30:20.397  INFO 49 --- [nio-8080-exec-1] net.bramp.ffmpeg.RunProcessFunction      : ffmpeg -y -v error -protocol_whitelist file,http,https,tcp,tls,udp,rtp -rtbufsize 1500M -re -i /tmp/test.sdp -f flv -vcodec libx264 -pix_fmt yuv420p -s 640x480 -r 20/1 -b:v 1000000 -acodec libmp3lame -ar 44100 -b:a 1000000 -bufsize 4000k -maxrate 1000k -profile:v baseline -deinterlace -preset medium -g 60 -r 30 rtmps://live-api-s.facebook.com:443/rtmp/3163232097002611?xyz
streamy-server_1  | [h264 @ 0x56538aeae0a0] non-existing PPS 0 referenced
streamy-server_1  |     Last message repeated 1 times
streamy-server_1  | [h264 @ 0x56538aeae0a0] decode_slice_header error
streamy-server_1  | [h264 @ 0x56538aeae0a0] no frame!
streamy-server_1  | [h264 @ 0x56538aeae0a0] non-existing PPS 0 referenced
streamy-server_1  |     Last message repeated 1 times
streamy-server_1  | [h264 @ 0x56538aeae0a0] decode_slice_header error
streamy-server_1  | [h264 @ 0x56538aeae0a0] no frame!
streamy-server_1  | [h264 @ 0x56538aeae0a0] non-existing PPS 0 referenced
streamy-server_1  |     Last message repeated 1 times
streamy-server_1  | [h264 @ 0x56538aeae0a0] decode_slice_header error
streamy-server_1  | [h264 @ 0x56538aeae0a0] no frame!
streamy-server_1  | [h264 @ 0x56538aeae0a0] non-existing PPS 0 referenced
streamy-server_1  |     Last message repeated 1 times
streamy-server_1  | [h264 @ 0x56538aeae0a0] decode_slice_header error
streamy-server_1  | [h264 @ 0x56538aeae0a0] no frame!
streamy-server_1  | [h264 @ 0x56538aeae0a0] non-existing PPS 0 referenced
streamy-server_1  |     Last message repeated 1 times
[...] (this is repeated for the next ~20-30 seconds, then continues with:)
kurento_1         | 0:03:00.152967182     1 0x7faa58093b30 INFO    KurentoWebSocketTransport WebSocketTransport.cpp:296:keepAliveSessions: Keep alive 998a9271-615e-490c-acce-6bc22d9592f7
streamy-server_1  | Too many packets buffered for output stream 0:0.
streamy-server_1  | 2020-07-15 08:30:38.361 ERROR 49 --- [nio-8080-exec-1] c.maximummgt.streamy.WebsocketsHandler   : Unknown error while websockets session
streamy-server_1  | 
streamy-server_1  | java.lang.RuntimeException: java.io.IOException: ffmpeg returned non-zero exit status. Check stdout.
streamy-server_1  |     at net.bramp.ffmpeg.job.SinglePassFFmpegJob.run(SinglePassFFmpegJob.java:46) ~[ffmpeg-0.6.2.jar:0.6.2]


  


  At facebook live it does not receive the video stream. In log I just see this message :

  


  Facebook has not received video signal from the video source for some time. Check that the connectivity between the video source and Facebook is sufficient for the source resolution and bitrate. Check your video encoder logs for details. If problems persist, consider improving connection quality or reducing the bitrate of your video source.

  


  On Kurento Java code I am doing following :

  


  


  1. Create an RTP endpoint

     


  


  2. Connect it with the video source from the user

     


  


  3. Create an SDP offer and save it to a file (currently /tmp/test.sdp)

     


  


  4. Process SDP offer with the RTP endpoint

     


  


  5. Start the ffmpeg process with (net.bramp.ffmpeg.builder.FFmpegBuilder) :

     


     FFmpegBuilder builder = new FFmpegBuilder()
.addExtraArgs("-protocol_whitelist", "file,http,https,tcp,tls,udp,rtp")
.addExtraArgs("-rtbufsize", "1500M")
.addExtraArgs("-re")
.setInput("/tmp/test.sdp")
.addOutput(rtmpURL)
.setFormat("flv")
.addExtraArgs("-bufsize", "4000k")
.addExtraArgs("-maxrate", "1000k")
.setAudioCodec("libmp3lame")
.setAudioSampleRate(FFmpeg.AUDIO_SAMPLE_44100)
.setAudioBitRate(1_000_000)
.addExtraArgs("-profile:v", "baseline")
.setVideoCodec("libx264")
.setVideoPixelFormat("yuv420p")
.setVideoResolution(width, height)
.setVideoBitRate(1_000_000)
.setVideoFrameRate(20)
.addExtraArgs("-deinterlace")
.addExtraArgs("-preset", "medium")
.addExtraArgs("-g", "60")
.addExtraArgs("-r", "30")
.done();

FFmpegExecutor executor = new FFmpegExecutor(ffmpeg, ffprobe);
executor.createJob(builder).run();



     


  


  


  Can somebody guide here on this issue ? Thanks in advance

  


  EDIT 01 : I disabled now

  


  // FFmpegExecutor executor = new FFmpegExecutor(ffmpeg, ffprobe);
// executor.createJob(builder).run();


  


  so that ffmpeg does not start automatically. After java created the test.sdp, I ran ffmpeg by myself in the console not to stream to facebook but to a mp4 :

  


  ffmpeg -loglevel debug -protocol_whitelist file,crypto,udp,rtp -re -vcodec libvpx -acodec opus -i /tmp/test.sdp -vcodec libx264 -acodec aac -y output.mp4

  


  The output looks as following - I interrupted it after 25s with CTRL + C

  


  root@app:/var/www# ffmpeg -loglevel debug -protocol_whitelist file,crypto,udp,rtp -re -vcodec libvpx -acodec opus -i /tmp/test.sdp -vcodec libx264 -acodec aac -y output.mp4
ffmpeg version 3.4.6-0ubuntu0.18.04.1 Copyright (c) 2000-2019 the FFmpeg developers
  built with gcc 7 (Ubuntu 7.3.0-16ubuntu3)
  configuration: --prefix=/usr --extra-version=0ubuntu0.18.04.1 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librubberband --enable-librsvg --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-omx --enable-openal --enable-opengl --enable-sdl2 --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-libopencv --enable-libx264 --enable-shared
  libavutil      55. 78.100 / 55. 78.100
  libavcodec     57.107.100 / 57.107.100
  libavformat    57. 83.100 / 57. 83.100
  libavdevice    57. 10.100 / 57. 10.100
  libavfilter     6.107.100 /  6.107.100
  libavresample   3.  7.  0 /  3.  7.  0
  libswscale      4.  8.100 /  4.  8.100
  libswresample   2.  9.100 /  2.  9.100
  libpostproc    54.  7.100 / 54.  7.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-protocol_whitelist' ... matched as AVOption 'protocol_whitelist' with argument 'file,crypto,udp,rtp'.
Reading option '-re' ... matched as option 're' (read input at native frame rate) with argument '1'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libvpx'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'opus'.
Reading option '-i' ... matched as input url with argument '/tmp/test.sdp'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'libx264'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'aac'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option 'output.mp4' ... matched as output url.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url /tmp/test.sdp.
Applying option re (read input at native frame rate) with argument 1.
Applying option vcodec (force video codec ('copy' to copy stream)) with argument libvpx.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument opus.
Successfully parsed a group of options.
Opening an input file: /tmp/test.sdp.
[NULL @ 0x55c2344e6a00] Opening '/tmp/test.sdp' for reading
[sdp @ 0x55c2344e6a00] Format sdp probed with size=2048 and score=50
[sdp @ 0x55c2344e6a00] audio codec set to: pcm_mulaw
[sdp @ 0x55c2344e6a00] audio samplerate set to: 44000
[sdp @ 0x55c2344e6a00] audio channels set to: 1
[sdp @ 0x55c2344e6a00] video codec set to: h264
[sdp @ 0x55c2344e6a00] RTP Packetization Mode: 1
[udp @ 0x55c2344e90c0] end receive buffer size reported is 131072
[udp @ 0x55c2344e9320] end receive buffer size reported is 131072
[sdp @ 0x55c2344e6a00] setting jitter buffer size to 500
[udp @ 0x55c2344ea0a0] end receive buffer size reported is 131072
[udp @ 0x55c2344ea180] end receive buffer size reported is 131072
[sdp @ 0x55c2344e6a00] setting jitter buffer size to 500
[sdp @ 0x55c2344e6a00] Before avformat_find_stream_info() pos: 305 bytes read:305 seeks:0 nb_streams:2
[libvpx @ 0x55c2344eeb80] v1.7.0
[libvpx @ 0x55c2344eeb80] --prefix=/usr --enable-pic --enable-shared --disable-install-bins --disable-install-srcs --size-limit=16384x16384 --enable-postproc --enable-multi-res-encoding --enable-temporal-denoising --enable-vp9-temporal-denoising --enable-vp9-postproc --target=x86_64-linux-gcc
[libvpx @ 0x55c2344eeb80] Invalid sync code e06101.
[libvpx @ 0x55c2344eeb80] Failed to decode frame: Bitstream not supported by this decoder
[libvpx @ 0x55c2344eeb80] Invalid sync code e06101.
[libvpx @ 0x55c2344eeb80] Failed to decode frame: Bitstream not supported by this decoder
    Last message repeated 1 times
[libvpx @ 0x55c2344eeb80] Invalid sync code e06101.
[libvpx @ 0x55c2344eeb80] Failed to decode frame: Bitstream not supported by this decoder
[sdp @ 0x55c2344e6a00] Non-increasing DTS in stream 1: packet 3 with DTS 5940, packet 4 with DTS 5940
[...]
[sdp @ 0x55c2344e6a00] Non-increasing DTS in stream 1: packet 956 with DTS 2027726, packet 957 with DTS 2027726
[libvpx @ 0x55c2344eeb80] Failed to decode frame: Bitstream not supported by this decoder
[libvpx @ 0x55c2344eeb80] Invalid sync code e06101.
[libvpx @ 0x55c2344eeb80] Failed to decode frame: Bitstream not supported by this decoder
[libvpx @ 0x55c2344eeb80] Invalid sync code e06101.
[libvpx @ 0x55c2344eeb80] Failed to decode frame: Bitstream not supported by this decoder
[libvpx @ 0x55c2344eeb80] Invalid sync code e06101.
[libvpx @ 0x55c2344eeb80] Failed to decode frame: Bitstream not supported by this decoder
[libvpx @ 0x55c2344eeb80] Invalid sync code 4a3bd8.
[sdp @ 0x55c2344e6a00] Non-increasing DTS in stream 1: packet 960 with DTS 2036690, packet 961 with DTS 2036690
[libvpx @ 0x55c2344eeb80] Failed to decode frame: Bitstream not supported by this decoder
[sdp @ 0x55c2344e6a00] interrupted
[sdp @ 0x55c2344e6a00] decoding for stream 1 failed
[sdp @ 0x55c2344e6a00] rfps: 30.000000 0.000926
[sdp @ 0x55c2344e6a00] rfps: 60.000000 0.003706
[sdp @ 0x55c2344e6a00] rfps: 120.000000 0.014824
[sdp @ 0x55c2344e6a00] Setting avg frame rate based on r frame rate
[sdp @ 0x55c2344e6a00] Could not find codec parameters for stream 1 (Video: vp8 (libvpx), 1 reference frame, none(progressive)): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
[sdp @ 0x55c2344e6a00] After avformat_find_stream_info() pos: 305 bytes read:305 seeks:0 frames:962
Input #0, sdp, from '/tmp/test.sdp':
  Metadata:
    title           : KMS
  Duration: N/A, start: 0.033000, bitrate: N/A
    Stream #0:0, 0, 1/44000: Audio: opus, 48000 Hz, mono, fltp
    Stream #0:1, 962, 1/90000: Video: vp8, 1 reference frame, none(progressive), 30 fps, 30 tbr, 90k tbn, 90k tbc
Successfully opened the file.
Parsing a group of options: output url output.mp4.
Applying option vcodec (force video codec ('copy' to copy stream)) with argument libx264.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument aac.
Successfully parsed a group of options.
Opening an output file: output.mp4.
[file @ 0x55c234563320] Setting default whitelist 'file,crypto'
Successfully opened the file.
[libvpx @ 0x55c2344eb0e0] v1.7.0
[libvpx @ 0x55c2344eb0e0] --prefix=/usr --enable-pic --enable-shared --disable-install-bins --disable-install-srcs --size-limit=16384x16384 --enable-postproc --enable-multi-res-encoding --enable-temporal-denoising --enable-vp9-temporal-denoising --enable-vp9-postproc --target=x86_64-linux-gcc
Stream mapping:
  Stream #0:1 -> #0:0 (vp8 (libvpx) -> h264 (libx264))
  Stream #0:0 -> #0:1 (opus (native) -> aac (native))
Press [q] to stop, [?] for help
Finishing stream 0:0 without any data written to it.
Finishing stream 0:1 without any data written to it.
detected 2 logical cores
[graph_1_in_0_0 @ 0x55c234561360] Setting 'time_base' to value '1/48000'
[graph_1_in_0_0 @ 0x55c234561360] Setting 'sample_rate' to value '48000'
[graph_1_in_0_0 @ 0x55c234561360] Setting 'sample_fmt' to value 'fltp'
[graph_1_in_0_0 @ 0x55c234561360] Setting 'channel_layout' to value '0x4'
[graph_1_in_0_0 @ 0x55c234561360] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x4
[format_out_0_1 @ 0x55c2345611e0] Setting 'sample_fmts' to value 'fltp'
[format_out_0_1 @ 0x55c2345611e0] Setting 'sample_rates' to value '96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
[AVFilterGraph @ 0x55c234560620] query_formats: 4 queried, 9 merged, 0 already done, 0 delayed
Nothing was written into output file 0 (output.mp4), because at least one of its streams received no packets.
frame=    0 fps=0.0 q=0.0 Lsize=       0kB time=-577014:32:22.77 bitrate=  -0.0kbits/s speed=N/A
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Input file #0 (/tmp/test.sdp):
  Input stream #0:0 (audio): 0 packets read (0 bytes); 0 frames decoded (0 samples);
  Input stream #0:1 (video): 0 packets read (0 bytes); 0 frames decoded;
  Total: 0 packets (0 bytes) demuxed
Output file #0 (output.mp4):
  Output stream #0:0 (video): 0 frames encoded; 0 packets muxed (0 bytes);
  Output stream #0:1 (audio): 0 frames encoded (0 samples); 0 packets muxed (0 bytes);
  Total: 0 packets (0 bytes) muxed
0 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x55c234563420] Statistics: 0 seeks, 0 writeouts
[aac @ 0x55c23458bea0] Qavg: -nan
[AVIOContext @ 0x55c2344ef6e0] Statistics: 305 bytes read, 0 seeks
Exiting normally, received signal 2.
root@app:/var/www# ls -lt
total 72
-rw-r--r-- 1 root root     0 Jul 15 09:12 output.mp4