[Live-devel] Possible bug in RTPInterface::sendDataOverTCP

[Hardik]

Hi

I am trying to stream rtp over tcp and i am facing the issue like stream 
is stopping after 3-4 mins(I am using VLC as a client), i applied 
following patch but didn't work for me.

>>
>>-      sendResult = send(socketNum, (char const*)(&data[numBytesSentSoFar]),
>>numBytesRemainingToSend, 0/*flags*/);
>>+
>>+      do {
>>+           sendResult = send(socketNum, (char
>>const*)(&data[numBytesSentSoFar]), numBytesRemainingToSend, 0/*flags*/);
>>+      } while(sendResult == -1 && envir().getErrno() == EAGAIN);
>>

I tried to incerease send buffer size to 100*1024 that is also not working for me, I am using 512000 bitrate.


---/Begin Message/  ---Thank you very much for detailed explanation.
Probably increasing of the send buffer (SO_SNDBUF) can help, it's value depends
on estimatedBitrate from RTPSink:if (rtpSink != NULL && rtpSink->estimatedBitrate() > 0) streamBitrate =
rtpSink->estimatedBitrate();

if (rtpGroupsock != NULL) {
     // Try to use a big send buffer for RTP -  at least 0.1 second of
     // specified bandwidth and at least 50 KB
     unsigned rtpBufSize = streamBitrate * 25 / 2; // 1 kbps * 0.1 s = 12.5 bytes
     if (rtpBufSize < 50 * 1024) rtpBufSize = 50 * 1024;
     increaseSendBufferTo(envir(), rtpGroupsock->socketNum(), rtpBufSize);
}

I was looking for this bunch of code, but i couldn't fine it, can you please help me? where i should change..

I am running the latest Live555 code 1-3-2015.

Thank you,

Regards,
-Hardik

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