[asterisk-bugs] [Asterisk 0010743]: Separate RTP pool for remote vs. local connections

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Nov 6 10:39:35 CST 2007


The following issue has been RESOLVED. 
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http://bugs.digium.com/view.php?id=10743 
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Reported By:                jamessan
Assigned To:                file
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Project:                    Asterisk
Issue ID:                   10743
Category:                   Core/RTP
Reproducibility:            always
Severity:                   feature
Priority:                   normal
Status:                     resolved
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 82568 
Disclaimer on File?:        N/A 
Request Review:              
Resolution:                 suspended
Fixed in Version:           
====================================================================== 
Date Submitted:             09-17-2007 10:25 CDT
Last Modified:              11-06-2007 10:39 CST
====================================================================== 
Summary:                    Separate RTP pool for remote vs. local connections
Description: 
Currently, Asterisk only has the notion of a single pool of RTP ports.  In
my setup (and I doubt this is uncommon), I have phones using my Asterisk
system from both the local network and remote network.  I'd like to be able
to limit the number of calls that can be made to/from the remote network
since my Asterisk system isn't very powerful and local<->remote calls
require Asterisk acting as a middle-man for the entirety of the call.  This
isn't a problem with local<->local calls because the phones REINVITE to
each other and don't involve Asterisk except for SIP signaling.

Simply limiting Asterisk's current RTP pool isn't an option because I
don't want to impose an artificial limitation on how many local calls can
be made.  Restricting the number of ports that are forwarded to Asterisk
from my firewall also isn't an option as the RTP port is chosen at random
and if Asterisk chooses one that I haven't forwarded, part of the audio
stream is lost.

To address this, I've created a patch for rtp.{c,h} which has allows the
user to specify remotertpstart/remotertpend along with the rtpstart/rtpend
options in rtp.conf.  I've also patched chan_sip.c to use the remote RTP
pool when it detects that the call leg it is processing is an external
call.
====================================================================== 

---------------------------------------------------------------------- 
 file - 11-06-07 10:39  
---------------------------------------------------------------------- 
I'm agreeing with Corydon and Qwell on this. I believe it can be
accomplished using groups, and the proposed patch seems very specialized.
Suspended for now, peace! 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
11-06-07 10:39  file           Status                   new => resolved     
11-06-07 10:39  file           Resolution               open => suspended   
11-06-07 10:39  file           Assigned To               => file            
11-06-07 10:39  file           Note Added: 0073196                          
======================================================================




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