ONsip has some tips for handling re-INVITEs with rtpproxy:<br><br><a href="http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html#rtp_loose_route">http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html#rtp_loose_route
</a><br><br>Advises to use force_rtp_proxy(l) on reinvites.<br><br><div><span class="gmail_quote">On 11/29/06, <b class="gmail_sendername">John Peters</b> <<a href="mailto:petersprc@gmail.com">petersprc@gmail.com</a>> wrote:
</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Not sure why that's happening. Probably setting canreinvite=no on the asterisk side will eliminate the re-INVITEs as a temporary solution, but still would like to know what is happening...
<br><br><span class="gmail_quote">
wrote:</span><br>> Sometimes, a calls b and b hears a, and a hears b for a second but a second<br>> INVITE comes to phone B that causes it to redirect rtp to be point to point.<br>> Sometimes there is no audio.<br>
> Sometimes, everything works fine.<br><br>> At one point, rtp from a was going to asterisk, but asterisk was not sending<br>> the rtp on to b, and b was trying to send traffic point to point.
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