[asterisk-bugs] [Asterisk 0013958]: SDP replies incorrect - 'a=inactive' - replied to with 'a=sendrecv'
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Jun 15 16:29:28 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=13958
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Reported By: toc
Assigned To: mnicholson
======================================================================
Project: Asterisk
Issue ID: 13958
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: feature
Priority: normal
Status: closed
Asterisk Version: 1.6.0
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
Resolution: fixed
Fixed in Version:
======================================================================
Date Submitted: 2008-11-24 05:15 CST
Last Modified: 2009-06-15 16:29 CDT
======================================================================
Summary: SDP replies incorrect - 'a=inactive' - replied to
with 'a=sendrecv'
Description:
When asterisk is sent a packet from Microsoft OCS Mediation Server, where
the call is being placed on hold, Asterisk sends an incorrect response.
Mediation server is 192.168.1.208
Asterisk is 192.168.1.250 (on 5061).
OpenSIPS is intercepting to resolve the bug on 192.168.1.250 (on 5060)
The packet sent from OCS is below (intercepted using OpenSIPS):
[ Method INVITE from 192.168.1.208:1276 (2) ]
INVITE sip:01234567 at 192.168.1.250:5061 SIP/2.0
FROM: <sip:First.Last at domain.com>;epid=A4325290A1;tag=f7768ee4ad
TO: <sip:01234567 at 192.168.1.250;user=phone>;tag=as79279f7b
CSEQ: 69 INVITE
CALL-ID: 1ae1e5ef-f96f-482c-9c44-da92c9094ecc
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.1.208:1276;branch=z9hG4bK1d297960
CONTACT:
<sip:domain.com.au:5060;transport=Tcp;maddr=192.168.1.208;ms-opaque=2cea147ecbefbb7f>
CONTENT-LENGTH: 268
SUPPORTED: 100rel
USER-AGENT: RTCC/3.0.0.0 MediationServer
CONTENT-TYPE: application/sdp; charset=utf-8
v=0
o=- 0 0 IN IP4 192.168.1.208
s=session
c=IN IP4 192.168.1.208
b=CT:1000
t=0 0
m=audio 60806 RTP/AVP 0 101
c=IN IP4 192.168.1.208
a=rtcp:60807
a=inactive
a=label:Audio
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
[ End of Request (2) ]
Response from Asterisk:
[ Reply 200 (OK) from 192.168.1.250:5061 concerning INVITE ]
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.250;branch=z9hG4bK3795.6ca29175.0;i=2;received=192.168.1.250
Via: SIP/2.0/TCP 192.168.1.208:1276;branch=z9hG4bK1d297960
From: <sip:First.Last at domain.com.au>;epid=A4325290A1;tag=f7768ee4ad
To: <sip:01234567 at 192.168.1.250;user=phone>;tag=as79279f7b
Call-ID: 1ae1e5ef-f96f-482c-9c44-da92c9094ecc
CSeq: 69 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:01234567 at 192.168.1.250:5061>
Content-Type: application/sdp
Content-Length: 261
v=0
o=root 295744575 295744576 IN IP4 192.168.1.250
s=Asterisk PBX 1.6.0
c=IN IP4 192.168.1.250
t=0 0
m=audio 19848 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
[ End of Reply ]
The asterisk packet contains an incorrect a=sendrecv, it should be
a=inactive as OCS has requested the call to be placed on hold.
The hold works when modifying the packet from a=sendrecv to a=inactive
using OpenSIPS.
======================================================================
----------------------------------------------------------------------
(0106429) svnbot (reporter) - 2009-06-15 16:29
https://issues.asterisk.org/view.php?id=13958#c106429
----------------------------------------------------------------------
Repository: asterisk
Revision: 200724
_U branches/1.6.0/
U branches/1.6.0/channels/chan_sip.c
U branches/1.6.0/configs/sip.conf.sample
------------------------------------------------------------------------
r200724 | kpfleming | 2009-06-15 16:29:27 -0500 (Mon, 15 Jun 2009) | 35
lines
Merged revisions 165180,200689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
........
r165180 | mnicholson | 2008-12-17 12:49:12 -0600 (Wed, 17 Dec 2008) | 14
lines
This patch adds a new 'ignoresdpversion' option to sip.conf. When this
is
enabled (either globally or for a specific peer), chan_sip will treat
any SDP
data it receives as new data and update the media stream accordingly.
By
default, Asterisk will only modify the media stream if the SDP session
version
received is different from the current SDP session version. This option
is
required to interoperate with devices that have non-standard SDP session
version implementations (observed by toc on the bug tracker with
Microsoft OCS
which always uses 0 as the session version).
http://reviewboard.digium.com/r/94/
(closes issue https://issues.asterisk.org/view.php?id=13958)
Reported by: toc
Tested by: toc
........
r200689 | kpfleming | 2009-06-15 15:42:38 -0500 (Mon, 15 Jun 2009) | 12
lines
Accept T.38 re-INVITE responses with invalid SDP versions.
This commit changes the 'incoming SDP version' check logic a bit more;
when
'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE
to
switch to T.38, we'll always accept the peer's SDP response, even if
they
don't properly increment the SDP version number as they should. If this
situation
occurs, a warning message will be generated suggesting that the peer's
configuration be changed to include the 'ignoresdpversion' configuration
option
(although ideally they'd fix their SIP implementation to be RFC
compliant).
AST-221
........
------------------------------------------------------------------------
http://svn.digium.com/view/asterisk?view=rev&revision=200724
Issue History
Date Modified Username Field Change
======================================================================
2009-06-15 16:29 svnbot Checkin
2009-06-15 16:29 svnbot Note Added: 0106429
======================================================================
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