[asterisk-bugs] [Asterisk 0013638]: Audio not passing between two Asterisk boxes when OpenSER in the middle

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13638 
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Reported By:                mpiazzatnetbug
Assigned To:                
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Project:                    Asterisk
Issue ID:                   13638
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.22 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-10-07 09:48 CDT
Last Modified:              2008-10-08 01:59 CDT
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Summary:                    Audio not passing between two Asterisk boxes when
OpenSER in the middle
Description: 
This issue is similar to the BUG number 0010481. I have two asterisk
identical each other, and a Openser in the middle to route the call based
on prefix (in product will be noumerous asterisk and some other sip PBX).
Every device is in the same network, so no NAT issue. 
Thia is the scenario

phone1 --> Asterisk1 --> Openser --> Asterisk2 --> phone2
The asterisk is registred on Openser with a Sip trunk, this is the sip
conf for the asterisk:


[phone1]
type=friend
username=phone1
callerid=("" <2678>)
secret=1234
context=users-distretto
callgroup=1
pickupgroup=1
canreinvite=yes
nat=no
host=dynamic


[siptrunk]
type=friend
username=8889
secret=8889
fromuser=8889
host=siptrunk.tnet.it
dtmfmode=rfc2833
fromdomain=siptrunk.tnet.it
context=default
insecure=very
canreinvite=yes

and on the dialplan to dial
exten => _04.,n,Dial(SIP/${EXTEN}@siptrunk)
and to recieve
exten => _04.,1,Goto(user|${EXTEN}|1)

If I have enable canreinvite=yes in the general section, on the phone and
on the siptrunk I can see a lot of 491 message and the RTP stream in not
forwarded correctly.

Of course If I choose canreinvit=no every were evithing is fine.

If I select canreinvite=no in general , and in the user and on the
siptrunk =yes the phone 1 send the rtp stream directly to the asterisk2 and
the phone 2 send the RTP stream to asterisk2.




 
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---------------------------------------------------------------------- 
 (0093342) mpiazzatnetbug (reporter) - 2008-10-08 01:59
 http://bugs.digium.com/view.php?id=13638#c93342 
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Isimplified the scenario. Now I have the two asterisk talking directly
without openser in the middle.

So in a 172.25.18.0/24 network

Phone 1 --> Asterisk (Udine) --> Asterisk (Aviano) --> Phone 2
.60            .67                   .66                  .83
SNOM 300                                                 LInksys

I still see the issue
as Attachment a wireshark trace. In rtpdebug.txt a dump od SIP debug and
in rtpdebug2.txt a dump of rtp debug.
Matteo 

Issue History 
Date Modified    Username       Field                    Change               
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2008-10-08 01:59 mpiazzatnetbug Note Added: 0093342                          
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