2007/12/20, Kristian Kielhofner <<a href="mailto:kristian.kielhofner@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">kristian.kielhofner@gmail.com</a>>:<div><span class="gmail_quote">
</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hello,<br><br> I'm very interested in a multicast RTP implementation for Asterisk.<br>I'm having some paging problems with app_page on smaller systems.<br>Having to setup a meetme, a new SIP channel for each participant, and
<br>handle all of the RTP for EVERY RTP seems a little unnecessary.</blockquote><div><br>I am currently working on a dialplan app that does just about that.<br><br>If you could provide me specs/info about how your phones
<br>expect the data (codec, packetization time) etc, I'll try to keep the <br>app as generic as possible.<br><br>The linksys phones do expect a start/stop signal to accept this<br>paging feature as far as I found out by capturing traffic. Unfortunately
<br>Linksys didn't bother letting me have the data/information on how<br>these packets have to be formated.<br><br>The Cisco 79xx series phones do have a similar feature. You do have<br>to authenticate against each phone's internal webserver though to
<br>have them listen to your multicast traffic.<br><br>If anyone has information about similar devices, be it phones or<br>automation systems like the <a href="http://barix.com">barix.com</a> series devices, which can<br>
handle rtp streams, please let me know.<br><br>-Andreas<br><br></div></div>