[asterisk-bugs] [Asterisk 0017712]: TOS_SIP does not get set

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Aug 19 16:07:19 CDT 2010


A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17712 
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Reported By:                nickb
Assigned To:                dvossel
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Project:                    Asterisk
Issue ID:                   17712
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
Asterisk Version:           1.6.2.10 
JIRA:                       SWP-1944 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 fixed
Fixed in Version:           
====================================================================== 
Date Submitted:             2010-07-25 22:54 CDT
Last Modified:              2010-08-19 16:07 CDT
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Summary:                    TOS_SIP does not get set
Description: 
Following set in sip.conf, setting reflected in sip show settings via CLI

tos_sip=CS3
tos_audio=EF

Place an outbound call and capture packets on the local server

RTP Packet Works:
11:39:59.554679 IP (tos 0xb8, ttl  64, id 0, offset 0, flags [DF], proto:
UDP (17), length: 200) LOCAL.26392 > REMOTE.8768: UDP, length 172

Signaling Packet Broken:
11:39:59.633869 IP (tos 0x0, ttl  64, id 35957, offset 0, flags [none],
proto: UDP (17), length: 479) LOCAL.sip > REMOTE.sip: SIP, length: 451
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
has duplicate       0017719 SIP packets not marked with configured ...
====================================================================== 

---------------------------------------------------------------------- 
 (0126173) svnbot (reporter) - 2010-08-19 16:07
 https://issues.asterisk.org/view.php?id=17712#c126173 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 282895

_U  branches/1.8/
U   branches/1.8/channels/chan_sip.c

------------------------------------------------------------------------
r282895 | dvossel | 2010-08-19 16:07:18 -0500 (Thu, 19 Aug 2010) | 25
lines

Merged revisions 282894 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.6.2

................
  r282894 | dvossel | 2010-08-19 16:05:54 -0500 (Thu, 19 Aug 2010) | 18
lines
  
  Merged revisions 282893 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r282893 | dvossel | 2010-08-19 16:03:24 -0500 (Thu, 19 Aug 2010) | 11
lines
    
    tos_sip option was not being set correctly
    
    When tos_sip is used, the tos of the sip socket is only set
    correctly if the socket binding changes on a reload.  If the binding
    stays the same but the TOS changes, the new tos value would not take
    into effect.  This patch fixes that.
    
    
    (closes issue https://issues.asterisk.org/view.php?id=17712)
    Reported by: nickb
  ........
................

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http://svn.digium.com/view/asterisk?view=rev&revision=282895 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-08-19 16:07 svnbot         Checkin                                      
2010-08-19 16:07 svnbot         Note Added: 0126173                          
======================================================================




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