[asterisk-bugs] [Asterisk 0017072]: Asterisk 1.6.1.18 and 1.6.2.6 RTP BUG

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
====================================================================== 
https://issues.asterisk.org/view.php?id=17072 
====================================================================== 
Reported By:                kvveltho
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   17072
Category:                   Core/General
Reproducibility:            always
Severity:                   block
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.1.18 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-03-22 08:11 CDT
Last Modified:              2010-03-22 08:11 CDT
====================================================================== 
Summary:                    Asterisk 1.6.1.18 and 1.6.2.6 RTP BUG
Description: 
When having more load (about 60 CC calls),  the following message start the
flood the console:
[Mar 22 11:43:43] ERROR[10983]: rtp.c:2563 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Mar 22 11:43:43] WARNING[10983]: chan_sip.c:6806 sip_alloc: Unable to
create RTP audio session: Address already in use
[Mar 22 11:43:44] ERROR[10983]: rtp.c:2563 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Mar 22 11:43:44] WARNING[10983]: chan_sip.c:6806 sip_alloc: Unable to
create RTP audio session: Address already in use
[Mar 22 11:43:45] ERROR[10983]: rtp.c:2563 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Mar 22 11:43:45] WARNING[10983]: chan_sip.c:6806 sip_alloc: Unable to
create RTP audio session: Address already in use
[Mar 22 11:43:45] ERROR[10983]: rtp.c:2563 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Mar 22 11:43:45] WARNING[10983]: chan_sip.c:6806 sip_alloc: Unable to
create RTP audio session: Address already in use
[Mar 22 11:43:46] ERROR[10983]: rtp.c:2563 ast_rtp_new_with_bindaddr: No
RTP ports remaining. Can't setup media stream for this call.
[Mar 22 11:43:46] WARNING[10983]: chan_sip.c:6806 sip_alloc: Unable to
create RTP audio session: Address already in use


We have plenty of RTP ports both on asterisk and our router.  See attached
rtp.conf

====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-03-22 08:11 kvveltho       Asterisk Version          => 1.6.1.18        
2010-03-22 08:11 kvveltho       Regression                => No              
2010-03-22 08:11 kvveltho       SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
======================================================================