<html><head><meta http-equiv="Content-Type" content="text/html charset=iso-8859-1"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;">In your previous message, you mentioned the possibility - in the "RTPInterface::sendDataOverTCP()" function - of the "send()" call succeeding in delivering partial (but not complete) data. I have had no other reports of this ever happening, but - on looking at the code - I realized that the code was not allowing for this possibility.<div><br></div><div>So, I have now released a new version of the code (2013.12.04) that should properly handle this possibility. Please upgrade to this new version (but also stop streaming over slow networks; you shouldn't be putting yourself in this situation to begin with!).</div><div><br></div><div>BTW, you also said:<br><div><blockquote type="cite"><div text="#000000" bgcolor="#FFFFFF">Most clients seem to handle this but if your client is based off of
the reference source (osrtspserver) it will fail.</div></blockquote><div><br></div>What is "osrtspserver"? I have never heard of this. In any case, the RTSP (and RTP/RTCP) protocol is defined by an IETF RFC, and does not have a 'reference source'. Whatever "osrtspserver" is, it is no more legitimate than any other implementation.</div></div><br><br><div>
<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>Live Networks, Inc.<br><a href="http://www.live555.com/">http://www.live555.com/</a></span>
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