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• Difference between MPEG-TS and RTP for H.264 (.mp4) video streaming ?

  6 avril 2022, par Candy

  So I am trying to stream a H.264 (.mp4) video over ETHERNET using the ffmpeg tool. I have read a little about transport of H.264 video over ethernet and have learnt that there are two methods ; mpeg-ts and RTP (both over UDP). I have been able to stream the .mp4 video through both methods in ffmpeg (over localhost) and haven't noticed any difference in quality or latency as such. What is the difference in concept and efficiency between the two protocols for transportation of video ? Or am I mixing two different concepts ? Any help is appreciated !

  


• Compress / Reduce file size of VIDEO

  2 avril 2014, par rishiJasapara

  I currently have a system in place where the user can upload a MP4 file and the same is available for download on mobile devices. But sometimes, the videos are more than 5MB in size and back here in my country, majority of the population uses 2G. So it typically takes 15-20 minutes to download large videos.

  Is there any way in which I can compress MP4 files while they are being uploaded and then save them in the folder so that instead of a 5MB video, I get a 2MB video which takes relatively less time to download. File format will always be MP4 only.

  I know of FFMPEG-PHP, but as far as I read, it only supports extracting information of the video and for video conversion. I could not find any reference for VIDEO COMPRESSION. I would be grateful if you could guide me on this.

• AppRTC : Google’s WebRTC test app and its parameters

  23 juillet 2014, par silvia

  If you’ve been interested in WebRTC and haven’t lived under a rock, you will know about Google’s open source testing application for WebRTC : AppRTC.

  When you go to the site, a new video conferencing room is automatically created for you and you can share the provided URL with somebody else and thus connect (make sure you’re using Google Chrome, Opera or Mozilla Firefox).

  We’ve been using this application forever to check whether any issues with our own WebRTC applications are due to network connectivity issues, firewall issues, or browser bugs, in which case AppRTC breaks down, too. Otherwise we’re pretty sure to have to dig deeper into our own code.

  Now, AppRTC creates a pretty poor quality video conference, because the browsers use a 640×480 resolution by default. However, there are many query parameters that can be added to the AppRTC URL through which the connection can be manipulated.

  Here are my favourite parameters :
  

  • hd=true : turns on high definition, ie. minWidth=1280,minHeight=720
  • stereo=true : turns on stereo audio
  • debug=loopback : connect to yourself (great to check your own firewalls)
  • tt=60 : by default, the channel is closed after 30min – this gives you 60 (max 1440)

  For example, here’s how a stereo, HD loopback test would look like : https://apprtc.appspot.com/?r=82313387&hd=true&stereo=true&debug=loopback .

  This is not the limit of the available parameter, though. Here are some others that you may find interesting for some more in-depth geekery :

  • ss=[stunserver] : in case you want to test a different STUN server to the default Google ones
  • ts=[turnserver] : in case you want to test a different TURN server to the default Google ones
  • tp=[password] : password for the TURN server
  • audio=true&video=false : audio-only call
  • audio=false : video-only call
  • audio=googEchoCancellation=false,googAutoGainControl=true : disable echo cancellation and enable gain control
  • audio=googNoiseReduction=true : enable noise reduction (more Google-specific parameters)
  • asc=ISAC/16000 : preferred audio send codec is ISAC at 16kHz (use on Android)
  • arc=opus/48000 : preferred audio receive codec is opus at 48kHz
  • dtls=false : disable datagram transport layer security
  • dscp=true : enable DSCP
  • ipv6=true : enable IPv6

  AppRTC’s source code is available here. And here is the file with the parameters (in case you want to check if they have changed).

  Have fun playing with the main and always up-to-date WebRTC application : AppRTC.

  UPDATE 12 May 2014

  AppRTC now also supports the following bitrate controls :

  • arbr=[bitrate] : set audio receive bitrate
  • asbr=[bitrate] : set audio send bitrate
  • vsbr=[bitrate] : set video receive bitrate
  • vrbr=[bitrate] : set video send bitrate

  Example usage : https://apprtc.appspot.com/?r=&asbr=128&vsbr=4096&hd=true