<html><head><base href="x-msg://125/"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; "><div><blockquote type="cite"><span class="Apple-style-span" style="border-collapse: separate; 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; font-size: medium; "><div lang="RU" link="blue" vlink="purple"><div class="WordSection1" style="page: WordSection1; "><div style="margin-top: 0cm; margin-right: 0cm; margin-left: 0cm; margin-bottom: 0.0001pt; font-size: 11pt; font-family: Calibri, sans-serif; "><span lang="EN-US">I am working on rtsp client software. Everything works fine until the transport on the server side was changed to tcp. I got the 461 error in “after-setup” handler. Ok, this is the easiest part – I just make some modification to code and handle this case (another try to setup session is done with the call to sendSetupCommand() and streamUsingTcp set to “true”, hope this is enough?).</span></div></div></div></span></blockquote><div><br></div>I think you are misunderstanding what is happening here.</div><div><br></div><div>When you (the client) request RTP-over-TCP streaming (by adding the "-t" option to "openRTSP"), the server decides whether or not it wants to support this. Because it returns a 461 error (meaning "Unsupported Transport"), this suggests that the server *does not* support RTP-over-TCP streaming. You cannot 'fix' this, and you should not modify the existing client code; the existing code is working OK. (Note also that if you modify the supplied code, you cannot expect any support on this mailing list.)</div><div><br></div><div>Because this server apparently does not support RTP-over-TCP streaming, the "-t" option to "openRTSP" will not work, and you should not use it. (If you don't receive any data when you don't use "-t", then that probably means that you are behind a firewall that is blocking UDP packets. If so, you will need to fix (or remove) your firewall.)</div><br><br><div apple-content-edited="true">
<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; font-size: medium; "><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; font-size: medium; ">Ross Finlayson<br>Live Networks, Inc.<br><a href="http://www.live555.com/">http://www.live555.com/</a></span></span>
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