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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 Ralf, 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 the link to the RCF6501 that bring a solution to know the timestamp of the first RTP packet.<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'>I follow times to times the ONVIF specification <a href="http://www.onvif.org/specs/stream/ONVIF-Streaming-Spec-v210.pdf">http://www.onvif.org/specs/stream/ONVIF-Streaming-Spec-v210.pdf</a> that also propose to insert a NTP timestamp (and other things to allow streaming backward) in the RTP header extension but using a different profile.<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'>In previous mail Ross told me that supporting RTP header extension in live555 is not in the roadmap.<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'>I will continue to try to find a 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'>Thanks,<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><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><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><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> Ralf Globisch<br><b>Envoyé :</b> mercredi 20 mars 2013 09:17<br><b>À :</b> finlayson@live555.com; live-devel@ns.live555.com<br><b>Objet :</b> Re: [Live-devel] Synchronize the "RTCP" time using RTSP Range ?<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Tahoma","sans-serif"'>Michel, I think the probability of loosing the first SR is very minor, SR packets are relatively small in comparison to the video packets and are less likely to get dropped by routers. Or do you have any indication otherwise? Ross's fix should do the trick.<o:p></o:p></span></p></div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Tahoma","sans-serif"'><br>However if you want to decrease that probability further RFC6051( </span><a href="http://tools.ietf.org/html/rfc6051" target="_blank"><span style='font-size:9.5pt;font-family:"Tahoma","sans-serif"'>http://tools.ietf.org/html/rfc6051</span></a><span style='font-size:9.5pt;font-family:"Tahoma","sans-serif"'> ) might be of use.</span><o:p></o:p></p><div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Tahoma","sans-serif"'>I'm not sure if it applies here, but from what I recall it's basically an extension header for RTP that includes an NTP timestamp allowing one to do as the name says: rapid synchronisation. You could add this extension header to the first e.g. 5 RTP packets *if* you notice a problem with the first SR being lost. To be sure read the draft to make sure it applies to this use case. <o:p></o:p></span></p></div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Tahoma","sans-serif"'><br>> As I said before (and am, I hope, saying for the last time), I'm not going to be doing anything non-standard here.</span><o:p></o:p></p><div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Tahoma","sans-serif"'>@Ross I would say this is a standard-conformant way? Standard conformant RTP applications *should* ignore the header in any case if the don't understand it, so it shouldn't break other clients...<o:p></o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Tahoma","sans-serif"'>@Michel If you do write the extension header and rapid sync code, you could consider contributing it back to the live555 codebase.<br><br>Regards,<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:9.5pt;font-family:"Tahoma","sans-serif"'>Ralf<o:p></o:p></span></p></div></div></div></body></html>