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• sdp : Add an option for sending RTCP packets to the source of the last packets

  13 août 2013, par Martin Storsjö

  sdp : Add an option for sending RTCP packets to the source of the last packets
  An SDP description normally only contains the target IP address
  
  and port for the packets. This means that we don’t really have
  
  any clue where to send the RTCP RR packets - previously they’re
  
  sent to the destination IP written in the SDP (at the same port),
  
  which rarely is the actual peer. And if the source for the packets
  
  is on a different port than the destination, it’s never correct.
  With a new option, we can choose to send the packets to the
  
  address that the latest packet on each socket arrived from.
  
  — -
  
  Some may even argue that this should be the default - perhaps,
  
  but I’d rather keep it optional at first. Additionally, I’m not
  
  sure if sending RTCP RR directly back to the source is
  
  desireable for e.g. multicast.
  Signed-off-by : Martin Storsjö <martin@martin.st>
  

  • [DH] libavformat/rtsp.c
  • [DH] libavformat/rtsp.h

• Decoding H.264 UDP stream

  24 février 2019, par user2422659

  I have to work with ’FFMPEG’ in order to decode ’UDP’ streaming(Not ’RTP’ or ’RTSP’) of raw ’h264’ video.address.
  So far i have compiled and make ’ffplay’ decode successfully a local ’h264’ file.
  Now i have to add support of that ’UDP’ listener who will get chunks of data.
  As far as i understand ’AVFormatContext.fileName’ gets a file name or ’RTSP’
  Any help ?
  what is the proper way to do this ?

  Thanks in advanced.

• Android receive RTP/UDP audio stream from VLC/ffmpeg

  23 mai 2016, par Dom4S

  I was searching for a good answer for half a day, but I am a beginner at this stuff and I would appreciate any help.

  What I would like to achieve is to stream audio (mp3 files) within ffmpeg or vlc and receive it on an Android device by udp/rtp.

  This is what I was able to figure out myself sofar :

  1) There is Android class AudioStream and RTPStream. What I don’t know is how to use it. For example I create a stream via ffmpeg with : ffmpeg -re -i mymp3.mp3 -ar 8000 -acodec copy -f rtp rtp://192.168.0.100:5533, where 192.168.0.100 is the address of my Android device. Now I would like to receive it and play it.

  I found something like this on Stack :

  AudioStream audioStream;
  
  AudioGroup audioGroup;
  
  AudioCodec codec = AudioCodec.PCMU;
  
  StrictMode.ThreadPolicy policy = new StrictMode.ThreadPolicy.Builder().permitNetwork().build();
  
  StrictMode.setThreadPolicy(policy);
  
  AudioManager audio = (AudioManager)getSystemService(AUDIO_SERVICE);
  
  audio.setMode(AudioManager.MODE_IN_COMMUNICATION);
  
  audioGroup = new AudioGroup();
  
  audioGroup.setMode(AudioGroup.MODE_NORMAL);
  
  InetAddress inetAddress;
  
  try {
  
      inetAddress = InetAddress.getByName("163.11.62.208");
  
      audioStream = new AudioStream(inetAddress);
  
      audioStream.setMode(RtpStream.MODE_RECEIVE_ONLY);
  
      audioStream.setCodec(codec);
  
      InetAddress inetAddressRemote = InetAddress.getByName("163.11.169.206");
  
      audioStream.associate(inetAddressRemote, 5004);
  
      audioStream.join(audioGroup);
  
  }

  What is the first inetAddress 163.11.62.208 and what is the second one 163.11.169.206 ? Shoudln’t I just give an address of a stream ?

  2) Can I submit only streams in PCMU format ? Can I stream mp3 files ?

  3) Is it even possible ?