[Home]

Summary:ASTERISK-14731: [patch] sip session timer: Does not work if initial INVITE min-se timer is too small
Reporter:Johann Steinwendtner (steinwej)Labels:
Date Opened:2009-09-23 10:55:23Date Closed:2015-01-19 20:10:02.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
is duplicated byASTERISK-23373 [patch]Security: Open FD exhaustion with chan_sip Session-Timers
Environment:Attachments:( 0) full_sip_session-timer.gz
( 1) session_timer_422.patch
Description:Asterisk sip with session timer enabled.
sip.conf:
session-timers=accept
session-expires=600
session-minse=180


Patton box connected to asterisk. Patton sends INVITE with session timer 90

asterisk responds with 422 session interval too small
patton reinvites with the proposed session timer.
asterisk send 200 ok, nothing happens. no tones or anything.
When patton sends BYE, asterisk sends ACK
But
sip channels remains, audio ports are not released

voip-1*CLI> sip show channels
Peer             User/ANR    Call ID          Format           Hold     Last Message  
91.128.104.50    (None)      302e3db1464e650  0x0 (nothing)    No       Rx: OPTIONS              
91.128.104.50    test_user   9e2ec18f1622d61  0x8 (alaw)       No       Rx: BYE                  
2 active SIP dialogs
voip-1*CLI>

voip-1*CLI> core show channels
Channel              Location             State   Application(Data)            
SIP/test_user-b7     01229922640@from_sip Down    (None)                        
1 active channel
0 active calls
0 calls processed
voip-1*CLI>
Comments:By: Johann Steinwendtner (steinwej) 2009-09-23 13:03:25

The problem was not there in 1.6.0.10. See patch.
By: Leif Madsen (lmadsen) 2009-09-24 09:07:43

I'm not 100% sure if 15890 is related, but I'm marking it as such for now as they appear similar.
By: Paul Belanger (pabelanger) 2010-06-01 10:50:45

Is this still an issue with the 1.6.2 branch (see below).
---
Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.6.0 and 1.6.1 branches has ended. For continued maintenance support please move to the 1.6.2 branch.

More information on this change can be found in the release announcement: http://www.asterisk.org/node/49924
By: Johann Steinwendtner (steinwej) 2010-06-07 13:28:28

Yes. Tested with 1.6.2.8
By: Corey Farrell (coreyfarrell) 2014-11-09 05:29:45.669-0600

I believe this is related to ASTERISK-23373.  The channel/RTP leaks should be resolved by this.  Is the REINVITE issue resolved?  Note 1.6 is out of support, so you need to be running 1.8.26.1, 11.8.1, 12.1.1 or higher.
By: Matt Jordan (mjordan) 2015-01-19 20:09:58.312-0600

I'm going to go with Corey on this. If it turns out I was wrong, please feel free to comment here (whoever) and we'll reopen the issue.