<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></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 dir="ltr" class=""><div class="">Thank you for your reply, I should not express myself well, I think your method provided is the normal way in live555 which add live aac source audio in to current Media session, but I want to remain the aac audio RTP streamed by Gstreamer, and the H264 video is finished by live555, then is that possible that add the audio which RTP streamed out by Gstreamer to current live555 video RTSP media session.</div></div></div></blockquote><div><br class=""></div>Ugh.</div><div><br class=""></div><div>Because your audio RTP stream is multicast, I think that you *may* be able to do what you want as follows:</div><div><span class="Apple-tab-span" style="white-space:pre"> </span>1/ Create a “Groupsock” object with address 244.1.1.1 and port number 10004, then</div><div><span class="Apple-tab-span" style="white-space:pre"> </span>2/ Create a “MPEG4GenericRTPSink” object, as I described in my previous email, using this “Groupsock” object, then</div><div><span class="Apple-tab-span" style="white-space:pre"> </span>3/ Create a “PassiveServerMediaSubsession” object, by calling “PassiveServerMediaSubsession::createNew(your-rtpSink-object, NULL)”, then</div><div><span class="Apple-tab-span" style="white-space:pre"> </span>4/ Add this “PassiveServerMediaSubsession” object - along with your existing “OnDemandServerMediaSubsession” (subclass) object (for H.264) to your RTSP server.</div><div><br class=""></div><div>But I’m not sure how well RTSP clients will be able to handle a stream that consists of two substreams - one unicast; the other multicast. I think this will work for VLC (which uses our RTSP client code), but I can’t say for sure.</div><div><br class=""></div><div>In any case, this is very weird (and it’s unlikely that you’ll ever be able to get proper audio/video synchronization this way). It’d be much better to feed your audio source directly into our RTSP server (as you already do with H.264), without using ‘Gstreamer’ crap. (We do RTP/RTSP much better than they do.)</div><br class=""><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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