[asterisk-bugs] [Asterisk 0017878]: [patch] chan_sip fails to remove hold when receving a reINVITE without SDP
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Aug 24 06:31:46 CDT 2010
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=17878
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Reported By: frawd
Assigned To: mnicholson
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Project: Asterisk
Issue ID: 17878
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: minor
Priority: normal
Status: confirmed
Target Version: 1.6.2.13
Asterisk Version: 1.6.2.11
JIRA: SWP-2064
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: 2010-08-17 11:05 CDT
Last Modified: 2010-08-24 06:31 CDT
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Summary: [patch] chan_sip fails to remove hold when receving
a reINVITE without SDP
Description:
Patch for issue https://issues.asterisk.org/view.php?id=14448 never made it to
trunk so all versions above 1.4
suffer this problem.
Here is an updated patch for 1.6.2.11.
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Relationships ID Summary
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related to 0014448 [patch] chan_sip fails to remove hold w...
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(0126259) frawd (reporter) - 2010-08-24 06:31
https://issues.asterisk.org/view.php?id=17878#c126259
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Exactly, it is a request for the receiving side to start the negotiation
(from zero). I don't see a case where we would want to start a new call
with hold...
As you say, the RFC is indicates that the offer should not take account of
the fact that a hold exists, so what do you mean by "agreement to remove
the hold"?; What hold??
The goal of the patch is actually to clear the hold, because it is not
supposed to exist in the first place! :-)
At this point, the choice would be between:
1 - Rewrite the SDP code to make it comply with the RFC.
2 - Add a simple workaround that might be ugly but works and has been
working for a while in 1.4 branch, closes a couple of issues (maybe more),
and does not break anything.
Oh, and 2 can be done while 1 is on its way!!
I would be very interested into helping for 1, but I have very few hours
to spend and low C skills so without any help I won't be able to. Is there
a SIP dev-group where this could be discussed?
Issue History
Date Modified Username Field Change
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2010-08-24 06:31 frawd Note Added: 0126259
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