Receiving RTP stream - AudioStream, AudioGroup

Question

I would like to listen an RTP audio stream, however the voice has little gaps in it - not continues. What may be the solution? Am I missing something on Receiver(android) side or Streamer(ffmpeg) side?

I'm using ffmpeg to stream RTP audio,

ffmpeg -f lavfi -i aevalsrc="sin(400*2*PI*t)" -ar 8000 -vcodec pcm_u8 -f rtp rtp://192.168.0.15:41954 (port changes.)

And here is my related android code:

AudioStream audioStream;
AudioGroup audioGroup;
@Override
public void onStart() {
    super.onStart();
    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_ECHO_SUPPRESSION);
    InetAddress inetAddress;
    try {
        inetAddress = InetAddress.getByName("192.168.0.15");
        audioStream = new AudioStream(inetAddress);
        audioStream.setCodec(AudioCodec.PCMU);
        audioStream.setMode(RtpStream.MODE_NORMAL);
        InetAddress inetAddressRemote = InetAddress.getByName("192.168.0.14");
        audioStream.associate(inetAddressRemote, 6000);
        ((TextView)findViewById(R.id.tv_port)).setText("Port : " + String.valueOf(audioStream.getLocalPort()));
        audioStream.join(audioGroup);
    }
    catch ( UnknownHostException e ) {
        e.printStackTrace();
    }
    catch ( SocketException e ) {
        e.printStackTrace();
    }
}

Answer 1

Answering my own question, the problem was with android rtp packet management.

Android said that ... assume packet interval is 50ms or less. in the AudioGroup source file.

However, RTP packets are sending with interval 60ms.

That means 50ms is not enough and this leads the problem as described below.

Incoming: X X X X X X Y Y Y Y Y Y X X X X X X Y Y Y Y Y Y X X X X X X
Reading : X X X X X Y Y Y Y Y X X X X X Y Y Y Y Y X X X X X Y Y Y Y Y
          ^ ^ ^ ^ ^ - - - - - - - - - - - - - - - - - - - - ^ ^ ^ ^ ^ 
          ^                                                 ^
          |                                                 |
          |---- just these overlapping packets is valid ----|
          |---- and other packets discarding due to --------|
          |---- invalid RTP headers. -----------------------|

X, Y < packets

I have just one packet in every 300ms interval. That results jittery sound.

I'll send a bug report for this, hope it helps someone.

For ones who are really want to listen raw RTP stream, I suggest them to read packets manually and decode it to PCM 16bit (which is the only audio format that android soundcard supports) and write it on AudioTrack.