[Home]

Summary:ASTERISK-14944: [patch] failed to negiotate t38
Reporter:Maciej Krajewski (jamicque)Labels:
Date Opened:2009-10-06 10:16:35Date Closed:2009-10-12 10:35:52
Priority:CriticalRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/T.38
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) asterisk.log
( 1) debug.patch
( 2) full.log
( 3) full.new
( 4) full.SPA
( 5) log.newest
( 6) mk.cap
Description:I found a problem today, after upgrading from 1.4.x to 1.6.0.15.
When I try to send fax using t38 with ZOIPER softphone thru asterisk to mu VoIP operator, asterisk fails to negotiate T38.

I found that after the ZOIPER send reinvite with t38, asterisk sends this reinvite to operator. And then Asterisk send reinvite to operator with g711 ??


The asterisk LOG and cap file is in attachment.
Comments:By: Kevin P. Fleming (kpfleming) 2009-10-06 14:43:33

Clearly something went wrong, but your log does not contain enough information to try to figure it out. Please ensure that your logger.conf has all logger levels enabled for the 'full' log file (notice,warning,verbose,debug,error) and test a call again after enabling 'core set verbose 10', 'core set debug 10' and 'sip set debug on'. Thanks.
By: Maciej Krajewski (jamicque) 2009-10-07 04:03:48

Hi,
what I've found is that when I'm sending fax behind NAT

ZOIPER ---NAT--> ASTERISK -NAT---> COMETREX

fax fails.
Sending fax thru the same network ZOIPER ----> ASTERISK ---NAT ---> COMETREX fails.

localnet and externip are set.

I've attached the log.
By: Maciej Krajewski (jamicque) 2009-10-07 04:39:17

I've done some new test, and problem with negotiating t38 can happen in the second scenario also:
ZOIPER ----> ASTERISK ---NAT ---> COMETREX

I've attached log full.new from this scenario.
By: Maciej Krajewski (jamicque) 2009-10-07 05:31:49

In full.SPA is a scenario SIPURA --- ASTERISK ---NAT----SIPURA
and the same problem... reinvites t38, and the asterisk reinvites one side to g711...
By: Kevin P. Fleming (kpfleming) 2009-10-09 16:30:02

In full.log, the Zoiper client fails to ACK the 200 OK sent by Asterisk (which is retransmitted six times), so the call fails completely.

In full.new, I can see the problem you are talking about. Do you have 'directmedia' (also known as 'canreinvite') enabled for these SIP peers? If so, please run a test with that disabled for one or both of them as I suspect it will correct this broken behavior and we'll know what needs to be fixed.
By: Maciej Krajewski (jamicque) 2009-10-09 16:44:43

Hi,
actually canreinvite is disabled on both peers.
By: Kevin P. Fleming (kpfleming) 2009-10-09 17:01:25

OK, on closer inspection of the log I see now that something is changing the T.38 state of the 'jamicque' peer's channel to T38_DISABLED, which causes the reinvite back to audio. Please apply the attached patch and test again; this patch will cause the T.38 state change message to include the function (and line number) that triggered the change. Thanks!
By: Maciej Krajewski (jamicque) 2009-10-09 18:02:32

The newest log log.patch added
By: Maciej Krajewski (jamicque) 2009-10-09 18:03:55

I had to change the name to log.newest coz i can't attach files with patch extension :)
By: Maciej Krajewski (jamicque) 2009-10-10 06:01:02

the same problem occurs in 1.4.26.2
By: Kevin P. Fleming (kpfleming) 2009-10-12 09:26:22

Thanks for the log; the problem is now apparent, and is due to some code we've been planning to remove anyway. I'll commit fixes to trunk and the 1.6.x branches shortly. Thanks for testing!
By: Digium Subversion (svnbot) 2009-10-12 09:29:11

Repository: asterisk
Revision: 223652

U   trunk/apps/app_fax.c
U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r223652 | kpfleming | 2009-10-12 09:29:10 -0500 (Mon, 12 Oct 2009) | 13 lines

Remove automatic switching from T.38 to voice mode in chan_sip.

chan_sip has some code to automatically switch from T.38 mode to voice mode when
a voice frame is written to the channel while it is in T.38 mode; this was
intended to handle the situation when a FAX transmission has ended and the channel
is not yet hung up, but is causing problems at the beginning of FAX sessions as
well when there are still voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover, and changes app_fax to
explicitly switch off T.38 mode when the FAX transmission process ends.

(closes issue ASTERISK-14944)
Reported by: jamicque

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223652
By: Digium Subversion (svnbot) 2009-10-12 09:31:41

Repository: asterisk
Revision: 223653

_U  branches/1.6.0/
U   branches/1.6.0/apps/app_fax.c
U   branches/1.6.0/channels/chan_sip.c

------------------------------------------------------------------------
r223653 | kpfleming | 2009-10-12 09:31:41 -0500 (Mon, 12 Oct 2009) | 20 lines

Merged revisions 223652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
 r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines
 
 Remove automatic switching from T.38 to voice mode in chan_sip.
 
 chan_sip has some code to automatically switch from T.38 mode to voice mode when
 a voice frame is written to the channel while it is in T.38 mode; this was
 intended to handle the situation when a FAX transmission has ended and the channel
 is not yet hung up, but is causing problems at the beginning of FAX sessions as
 well when there are still voice frames 'in flight' at the time the T.38 negotiation
 completes. This patch removes the automatic switchover, and changes app_fax to
 explicitly switch off T.38 mode when the FAX transmission process ends.
 
 (closes issue ASTERISK-14944)
 Reported by: jamicque
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223653
By: Digium Subversion (svnbot) 2009-10-12 09:36:03

Repository: asterisk
Revision: 223654

_U  branches/1.6.1/
U   branches/1.6.1/apps/app_fax.c
U   branches/1.6.1/channels/chan_sip.c

------------------------------------------------------------------------
r223654 | kpfleming | 2009-10-12 09:36:03 -0500 (Mon, 12 Oct 2009) | 20 lines

Merged revisions 223652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
 r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines
 
 Remove automatic switching from T.38 to voice mode in chan_sip.
 
 chan_sip has some code to automatically switch from T.38 mode to voice mode when
 a voice frame is written to the channel while it is in T.38 mode; this was
 intended to handle the situation when a FAX transmission has ended and the channel
 is not yet hung up, but is causing problems at the beginning of FAX sessions as
 well when there are still voice frames 'in flight' at the time the T.38 negotiation
 completes. This patch removes the automatic switchover, and changes app_fax to
 explicitly switch off T.38 mode when the FAX transmission process ends.
 
 (closes issue ASTERISK-14944)
 Reported by: jamicque
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223654
By: Maciej Krajewski (jamicque) 2009-10-12 09:40:38

kpfleming the same thing happens in 1.4.26.2
By: Digium Subversion (svnbot) 2009-10-12 09:40:59

Repository: asterisk
Revision: 223655

_U  branches/1.6.2/
U   branches/1.6.2/apps/app_fax.c
U   branches/1.6.2/channels/chan_sip.c

------------------------------------------------------------------------
r223655 | kpfleming | 2009-10-12 09:40:58 -0500 (Mon, 12 Oct 2009) | 20 lines

Merged revisions 223652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

........
 r223652 | kpfleming | 2009-10-12 09:25:29 -0500 (Mon, 12 Oct 2009) | 13 lines
 
 Remove automatic switching from T.38 to voice mode in chan_sip.
 
 chan_sip has some code to automatically switch from T.38 mode to voice mode when
 a voice frame is written to the channel while it is in T.38 mode; this was
 intended to handle the situation when a FAX transmission has ended and the channel
 is not yet hung up, but is causing problems at the beginning of FAX sessions as
 well when there are still voice frames 'in flight' at the time the T.38 negotiation
 completes. This patch removes the automatic switchover, and changes app_fax to
 explicitly switch off T.38 mode when the FAX transmission process ends.
 
 (closes issue ASTERISK-14944)
 Reported by: jamicque
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223655
By: Digium Subversion (svnbot) 2009-10-12 10:34:23

Repository: asterisk
Revision: 223692

U   branches/1.4/channels/chan_sip.c

------------------------------------------------------------------------
r223692 | kpfleming | 2009-10-12 10:34:23 -0500 (Mon, 12 Oct 2009) | 13 lines

Remove automatic switching from T.38 to voice mode in chan_sip.
 
chan_sip has some code to automatically switch from T.38 mode to voice mode when
a voice frame is written to the channel while it is in T.38 mode; this was
intended to handle the situation when a FAX transmission has ended and the channel
is not yet hung up, but is causing problems at the beginning of FAX sessions as
well when there are still voice frames 'in flight' at the time the T.38 negotiation
completes. This patch removes the automatic switchover.
 
(issue ASTERISK-14944)
Reported by: jamicque


------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223692
By: Digium Subversion (svnbot) 2009-10-12 10:35:52

Repository: asterisk
Revision: 223693

_U  trunk/

------------------------------------------------------------------------
r223693 | kpfleming | 2009-10-12 10:35:52 -0500 (Mon, 12 Oct 2009) | 19 lines

Recorded merge of revisions 223692 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
 r223692 | kpfleming | 2009-10-12 10:30:40 -0500 (Mon, 12 Oct 2009) | 13 lines
 
 Remove automatic switching from T.38 to voice mode in chan_sip.
   
 chan_sip has some code to automatically switch from T.38 mode to voice mode when
 a voice frame is written to the channel while it is in T.38 mode; this was
 intended to handle the situation when a FAX transmission has ended and the channel
 is not yet hung up, but is causing problems at the beginning of FAX sessions as
 well when there are still voice frames 'in flight' at the time the T.38 negotiation
 completes. This patch removes the automatic switchover.
   
 (issue ASTERISK-14944)
 Reported by: jamicque
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=223693