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<DIV><SPAN class=268205318-23022005><FONT face=Arial size=2>While trying to
deploy a bunch of Polycom IP 500 phones, I ran in to the following. I
limited the RTP ports from 8000-8050 to limit holes in firewall. Pretty
soon Asterisk ran out of RTP ports. Traced the problem back to how * is
handling SUBSCRIBE. A sip structure is allocated as soon as a request is
received, which also allocates RTP ports. Normally, this is not a problem
as the structure is released as soon as the request is answered, which is pretty
quick. However, SUBSCRIBE has an expiry header, till which time * keeps
around the call structure and hence hanging on to those RTP ports.
(SUBSCRIBE doesn't even need an RTP port). This problem is amplified by
Polycom "Buddy Lists" as each phone subscribes to changes in extension status of
all the "buddy" up to 7 extensions.</FONT></SPAN></DIV>
<DIV><SPAN class=268205318-23022005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=268205318-23022005><FONT face=Arial size=2>I just added few
lines in handle_request functions to release RTP ports if it is a
subscribe. Seem to work fine.. Should I send this to Dev list or
somewhere?</FONT></SPAN></DIV>
<DIV><SPAN class=268205318-23022005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=268205318-23022005><FONT face=Arial size=2>Hope this helps some
one running in to similar problems.</FONT></SPAN></DIV>
<DIV><SPAN class=268205318-23022005><FONT face=Arial
size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=268205318-23022005><FONT face=Arial
size=2>Thanks<BR>Sarat.</FONT></SPAN></DIV></FONT></DIV></BODY></HTML>