[asterisk-bugs] [Asterisk 0016949]: sip reinvite broken

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16949 
====================================================================== 
Reported By:                drookie
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16949
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0.24 
JIRA:                       SWP-983 
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-03-03 00:28 CST
Last Modified:              2010-06-02 13:27 CDT
====================================================================== 
Summary:                    sip reinvite broken
Description: 
Actually I have asterisk 1.6.0.25, but this interface doesn't allow to
report it.
OS: FreeBSD.

Sample scheme:
PSTN <--E1---> cisco <---SIP---> asterisk <---SIP---> linksys ata

In the supplied wireshark cap file:
Cisco: 192.168.3.40
Asterisk: 192.168.3.20 (main), 192.168.3.28(carp)
Linksys: 192.168.3.197

Cisco has only g711 codec enabled on its voip dial-peer leg.
Linksys has g723 and g729 as preferred codecs, but other codecs _aren't_
restricted.
Asterisk settings:

Cisco:
[kosm65-gw1]
type=peer
insecure=invite,port
host=192.168.3.40
context=kosm65-gw1-incoming
dtmfmode=rfc2833
disallow=all
allow=alaw
canreinvite=yes

Linksys:
[rybalko81-voip3]
username=rybalko81-voip3
secret=somesecret
type=friend
host=dynamic
insecure=invite,port
context=ordinary
disallow=all
allow=g729
allow=g723
dtmfmode=auto
canreinvite=yes

Problem description:
I place call from Linksys to PSTN (number 92931575). If other than
g729/g723 codecs aren't restricted in the Linksys config (it has also g726
and g711), it sends them in the initial SDP. And then asterisk sends
reinvites: to cisco for Linksys address and with g711 codec, and to Linksys
for Cisco address _but_ with codecs g729/g723 (I could understand if it
sends reinvite with g711 codec which is present in Linksys's SDP and which
can be understood by Cisco, but it definitely reinvites with codecs from
asterisk config). All of this results in one-way audio (PSTN cannot hear
Linksys), as Linksys can understand g711 and Cisco cannot understand g72x.

Workaround:
If other than g723/g729 codecs are restricted on the Linksys side,
matching the asterisk config, no reinvite occurs. Same effect when
canreinvite is set to 'no'.
====================================================================== 

---------------------------------------------------------------------- 
 (0122821) pabelanger (manager) - 2010-06-02 13:27
 https://issues.asterisk.org/view.php?id=16949#c122821 
---------------------------------------------------------------------- 
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
 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-06-02 13:27 pabelanger     Note Added: 0122821                          
2010-06-02 13:27 pabelanger     Status                   acknowledged =>
feedback
======================================================================