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Summary:ASTERISK-15734: sip reinvite broken
Reporter:Eugene M. Zheganin (drookie)Labels:
Date Opened:2010-03-03 00:28:09.000-0600Date Closed:2011-06-07 14:00:24
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) console-debug.txt
( 1) wireshark-session.cap
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'.
Comments:By: Paul Belanger (pabelanger) 2010-06-02 13:27:47

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: Paul Belanger (pabelanger) 2010-06-10 15:08:17

Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested.

Further information can be found at http://www.asterisk.org/developers/bug-guidelines