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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