<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class=""><div><blockquote type="cite" class=""><div class=""><div bgcolor="#FFFFFF" text="#000000" class="">Good guess, I am indeed running the vlc players at the same
computer. However, if I use multiple proxy servers and have one
player per proxy server it is running fine. In that case there are
still multiple copies of vlc on the same computer. What could be the
explanation for this behavior?<br class=""></div></div></blockquote><div><br class=""></div>I’m not sure. However, earlier, you wrote:</div><div><br class=""></div><div><blockquote type="cite" class=""><div bgcolor="#FFFFFF" text="#000000" class="">I already tried to increase the buffer size in the RTSPServer (increaseSendBufferTo) to 2 MB, but it does not result in a better performance. By the way, I also increased the buffer size at the OS (Linux) with sysctl net.core.rmem_max.<br class=""></div></blockquote><div><br class=""></div></div>Make sure that you’re increasing the OS buffer size for the socket that matters: *Not* the RTSP server’s main socket (because that’s used only to handle connections from new clients), nor the “RTSPClientConnection” socket (because that’s used only to handle incoming RTSP commands, unless you’re streaming everything RTP-over-TCP), but instead the socket that’s used by the “RTPSink” objects (to transmit RTP/UDP packets).<div class=""><br class=""></div><div class="">Unfortunately right now I don’t know of a way to access this socket without modifying the “ProxyServerMediaSession.cpp” code, but you can access this socket in the “ProxyServerMediaSubsession:createNewRTPSink()” implementation (starting at line 554 of “ProxyServerMediaSession.cpp”). The socket in question is</div><div class=""><span class="Apple-tab-span" style="white-space:pre"> </span>rtpGroupsock->socketNum()</div><div class=""><br class=""></div><div class="">Note also that the “increaseSendBufferTo()” function returns the resulting size of the buffer, so you can look at the result of this call, to see what the resulting buffer size actually becomes.</div><div class=""><br class=""></div><div class="">I’d be interested to see if increasing the RTP socket’s OS buffer size alleviates your problem at all.</div><br class=""><div apple-content-edited="true" class="">
<span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; "><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: Helvetica; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px; -webkit-border-horizontal-spacing: 0px; -webkit-border-vertical-spacing: 0px; -webkit-text-decorations-in-effect: none; -webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; ">Ross Finlayson<br class="">Live Networks, Inc.<br class=""><a href="http://www.live555.com/" class="">http://www.live555.com/</a></span></span>
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