[asterisk-bugs] [Asterisk 0018633]: asterisk does not respond with ACK on retransmission of 200 OK after it sent ACK
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Feb 16 15:50:33 CST 2011
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=18633
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Reported By: rrevels
Assigned To:
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Project: Asterisk
Issue ID: 18633
Category: Channels/chan_sip/General
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: Older 1.6.2 - please test a newer version
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2011-01-17 09:31 CST
Last Modified: 2011-02-16 15:50 CST
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Summary: asterisk does not respond with ACK on retransmission
of 200 OK after it sent ACK
Description:
Upstream blocking causes the first ACK to be dropped before hitting
destination. Retransmitted 200s show up in network trace on Asterisk
server but no ACK response is ever generated
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(0132050) lmadsen (administrator) - 2011-02-16 15:50
https://issues.asterisk.org/view.php?id=18633#c132050
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rrevels: OK, so you have told us what version this isn't a problem on, but
have not told us what version this *is* a problem on. You've marked this as
an older version of 1.6.2, and said more recent versions do not exhibit
this problem.
Additionally I do see Asterisk respond to the 200 OK with an ACK in your
message:
<--- SIP read from UDP:216.27.87.216:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
208.44.174.45:5060;received=208.44.174.45;branch=z9hG4bK56604b66;rport=5060
Record-Route: <sip:216.27.87.216;lr;ftag=as10f5522b>
From: "JOE" <sip:9194397461 at 208.44.174.45>;tag=as10f5522b
To: <sip:+19197023718 at 216.27.87.216>;tag=_2462058652-1042412449
Call-ID: 70580fae265510cb21ccfd2430e8ed5b at 208.44.174.45
CSeq: 102 INVITE
Contact: sip:+19197023718 at 12.34.5.94:5060
Accept: application/sdp
Allow: INVITE,ACK,CANCEL,BYE,INFO,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE
Content-Disposition: session; handling=required
Content-Type: application/sdp
Content-Length: 229
v=0
o=305419896 305419896 IN IP4 12.34.5.125
s=-
c=IN IP4 12.34.5.125
t=0 0
m=audio 15632 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=maxptime:40
a=sendrecv
<------------->
--- (13 headers 12 lines) ---
list_route: hop: <sip:216.27.87.216;lr;ftag=as10f5522b>
set_destination: Parsing <sip:216.27.87.216;lr;ftag=as10f5522b> for
address/port to send to
set_destination: set destination to 216.27.87.216, port 5060
Transmitting (no NAT) to 216.27.87.216:5060:
ACK sip:+19197023718 at 12.34.5.94:5060 SIP/2.0
Via: SIP/2.0/UDP 208.44.174.45:5060;branch=z9hG4bK30148e3c;rport
Route: <sip:216.27.87.216;lr;ftag=as10f5522b>
Max-Forwards: 70
From: "JOE" <sip:9194397461 at 208.44.174.45>;tag=as10f5522b
To: <sip:+19197023718 at 216.27.87.216>;tag=_2462058652-1042412449
Contact: <sip:9194397461 at 208.44.174.45>
Call-ID: 70580fae265510cb21ccfd2430e8ed5b at 208.44.174.45
CSeq: 102 ACK
User-Agent: providerVoice
Remote-Party-ID: "JOE"
<sip:9194397461 at 208.44.174.45>;privacy=off;screen=no
Content-Length: 0
To me this looks like either Asterisk isn't sending the response to the
right spot, or that the other end is behind NAT or something. NAT is
typically when I see this type of behaviour.
Issue History
Date Modified Username Field Change
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2011-02-16 15:50 lmadsen Note Added: 0132050
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