[asterisk-bugs] [Asterisk 0013958]: SDP replies incorrect - 'a=inactive' - replied to with 'a=sendrecv'

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Dec 10 17:33:23 CST 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13958 
====================================================================== 
Reported By:                toc
Assigned To:                mnicholson
====================================================================== 
Project:                    Asterisk
Issue ID:                   13958
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.6.0 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-11-24 05:15 CST
Last Modified:              2008-12-10 17:33 CST
====================================================================== 
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.
====================================================================== 

---------------------------------------------------------------------- 
 (0096164) mnicholson (administrator) - 2008-12-10 17:33
 http://bugs.digium.com/view.php?id=13958#c96164 
---------------------------------------------------------------------- 
Ok.  After looking at this a little further, it appears asterisk is not
honoring the new SDP data because the the SDP session version number (field
'o' is not incremented).  If this is the case you should see this message
in the debug log:

SDP version number same as previous SDP

To work around this, we could add some non standard behavior to asterisk
so that it ignores the session version and always treats the SDP info as
new.  You should also be able to work around this by increasing the SDP
version number each time a packet from your OCS server goes through your
proxy.  See http://tools.ietf.org/html/rfc4566#section-5.2 for info on the
format of that 'o=...' line. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-12-10 17:33 mnicholson     Note Added: 0096164                          
======================================================================




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