Hi ,
Thanks for your instant reply,
I am not saying there is bug in code. I am really sorry if you felt so. we
are using live555 code as RTSP server, we started with doing UDP streaming
which works very well, now our customer wants to change it to TCP. so we
are trying to debug.
we are using wis
> 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)
Oh please! Why do you think that this means that there is a bug in our code at
all, let alone a bug in one specific function?
I assume (although you didn’t say ex
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/*flag
> Probably increasing of the send buffer (SO_SNDBUF) can help
Yes, you can call "increaseSendBufferTo()" in your application. However,
you'll still get data loss if your stream's bitrate exceeds the capacity of
your TCP connection (which is *not* the same as the nominal bitrate of your
network
--- 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 (rtpGroup
> I tried to fix the problem by this way:
>
> - sendResult = send(socketNum, (char const*)(&data[numBytesSentSoFar]),
> numBytesRemainingToSend, 0/*flags*/);
> +
> + do {
> + sendResult = send(socketNum, (char
> const*)(&data[numBytesSentSoFar]), numBytesRemainingToSend
