[Live-devel] 回复： Re: live555 under high bitrate streaming

[Sourav Pal]

2008/6/11 Xzou <xiangwust at yahoo.cn>:
>
> I wrote a simple program which just receive a stream using rtp and dump it based on live555.In this program,I increased the socket buffer to 2,000,000 bytes using the function increaseReceiveBufferTo() provided by live555.Then  compare  the dumped  file  and the original file,I found it's much smaller than the original file.When playing the dumped file using a common media player like VLC or mplayer, the video picture has broken.The original file's bitrate is about 20M bit per second.That seems many data has dropped.But my network card works good,it can work in 100M bit/s. Then I perceive there is something wrong in rtp processing in live555 under high bitrate environment.
>
> Sourav Pal <souravpal at gmail.com> 写道：
>
> 2008/6/10 Xzou <xiangwust at yahoo.cn>:
>>
>> Recently,I'm trying to construct a embedded rtsp/rtp client using live555 in order to receive very high bitrate media stream which could be nearly 20M bit per second.But the bad news is my client lost a lot of packets when the bitrate of media stream is higher than 4M bit per second.
>> My cpu frequency is 200M hz and have no more than 64M ram available for application softwares.
>> How could I fix it?
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>
> Where are your packets getting dropped ? How did you measure it ?
> You might want to increase the size of the udp buffers. That does help sometimes.
>
> Thanks,
> Sourav
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The problem might not be at your NIC. These sort of high bit rate cane
cause packet drops at both the streaming side ( most likely) as well
as on the receive side. It might be that UDP buffers are dropping the
packets. You might want to increase the udp buffer size\
at both the receive and transmit end. If you are in linux , use the following :

sysctl  net.core.rmem_max = < compute how much you need for 20mbps>
sysctl  net.core.wmem_max = < compute how much you need for 20mbps>

Thanks,
Sourav