[asterisk-bugs] [Asterisk 0015182]: [patch] T.38 invite does not always comply with RFC 2327
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Jul 23 09:08:41 CDT 2009
A NOTE has been added to this issue.
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https://issues.asterisk.org/view.php?id=15182
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Reported By: CGMChris
Assigned To:
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Project: Asterisk
Issue ID: 15182
Category: Channels/chan_sip/T.38
Reproducibility: sometimes
Severity: minor
Priority: normal
Status: ready for testing
Target Version: 1.4.27
Asterisk Version: 1.4.25
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
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Date Submitted: 2009-05-22 09:31 CDT
Last Modified: 2009-07-23 09:08 CDT
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Summary: [patch] T.38 invite does not always comply with RFC
2327
Description:
See http://www.faqs.org/rfcs/rfc2327.html, section 6:
"Note: For transports based on UDP, the value should be in the range 1024
to 65535 inclusive. For RTP compliance it should be an even number."
My SIP provider drops the call when my signaling specifies an odd numbered
media port for T.38 (per RFC 2327).
In most of my tests, the media port in the T.38 invite is even:
1. From my provider to Asterisk
2. From Asterisk to my ATA
3. From my ATA back to Asterisk
At this point, (step 4) Asterisk seems to be prone to generating an odd
numbered port for signaling to my provider.
I *think* this issue has not been noticed widespread because ALG on most
routers corrects this behavior. I do NOT have a problem using T.38 with a
cheap $60 Linksys router, however, using an off-brand router from Hong Kong
with no ALG support, I cannot get T.38 to work at all.
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(0108114) globalnetinc (reporter) - 2009-07-23 09:08
https://issues.asterisk.org/view.php?id=15182#c108114
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Is there any way that this patch can be ported to 1.6.1? Our SIP provider
uses Sonus as well and we have been unable to get T.38 to work. They
always drop the call.
Issue History
Date Modified Username Field Change
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2009-07-23 09:08 globalnetinc Note Added: 0108114
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