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</o:shapelayout></xml><![endif]--></head><body lang=FR link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Hi Ross,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Thanks for your support and your patience.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>But as you said reducing SR period just reduce the risk.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Perhaps a trade-off between the standard and what could do a live555 user could be to give a way to RTSPClient user to give the fSyncTime initialization.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Through a non standard way to acquire the time reference (perhaps additional RTSP ), I guess it will then be possible through MediaSubSession -> RTPSource -> RTPReceptionStats to set the initial value.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Do you think this could be an acceptable way ?<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Best Regards,<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> Michel.<o:p></o:p></span></p><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div><div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>[@@ THALES GROUP INTERNAL @@]<o:p></o:p></span></p></div></div></div><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>De :</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> live-devel-bounces@ns.live555.com [mailto:live-devel-bounces@ns.live555.com] <b>De la part de</b> Ross Finlayson<br><b>Envoyé :</b> lundi 18 mars 2013 21:57<br><b>À :</b> LIVE555 Streaming Media - development & use<br><b>Objet :</b> Re: [Live-devel] Synchronize the "RTCP" time using RTSP Range ?<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><div><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>What I was suggesting is this “missing” NTP information (before RTCP “SR”) could perhaps be computed from RTSP PLAY answer.</span><o:p></o:p></p></div><div><p class=MsoNormal><span lang=EN-US style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>It is quite annoying to drop data waiting for RTCP “SR” because we cannot send the timestamp through RTSP/RTP/RTCP mechanism.</span><o:p></o:p></p></div></div></blockquote><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal>What I can probably do is upgrade our server code so that it sends a RTCP "SR" packet immediately upon handling a "PLAY" command (before the first RTP packet is sent). (I think this is actually prescribed in one of the IETF RFCs.) That won't be 100% reliable (because this first RTCP packet may get lost, but most of the time it will eliminate the delay before RTCP synchronization occurs.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Yes, it would be nice if the "RTP-Info:" data included a NTP-format time (the same as in RTCP "SR"s), but unfortunately that's not in the standard.<o:p></o:p></p></div><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p><div><p class=MsoNormal><span class=apple-style-span><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif";color:black'>Ross Finlayson</span></span><span style='font-size:13.5pt;font-family:"Helvetica","sans-serif";color:black'><br><span class=apple-style-span>Live Networks, Inc.</span><br><span class=apple-style-span><a href="http://www.live555.com/">http://www.live555.com/</a></span></span> <o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></body></html>