Sarat Vemuri wrote:
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.
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?
Please add your patch to the bug tracker!
http://bugs.digium.com
Thank you for contributing.
/Olle
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