[asterisk-bugs] [Asterisk 0015182]: T.38 invite does not always comply with RFC 2327

[Asterisk Bug Tracker]

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:                     new
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:              2009-06-12 13:15 CDT
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Summary:                    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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 (0106345) mmichelson (administrator) - 2009-06-12 13:15
 https://issues.asterisk.org/view.php?id=15182#c106345 
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I have a couple of problems with this issue report.

1) RFC 2327 is deprecated in favor of RFC 4566, which does not have any
sort of restriction regarding the use of even numbered ports.

2) Even if you ignore the existence of RFC 4566 and follow RFC 2327, the
passage you quoted says that RTP ports should have even numbers. It does
not carry such a restriction for UDPTL. I'll admit though that the language
used is a bit ambiguous and that I may be misinterpreting it.

So to me anyway, this seems like buggy behavior by your SIP provider, not
Asterisk.

That being said, it wouldn't be all *that* difficult to make sure that the
port selected is even. I can work up a patch for you to try. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-06-12 13:15 mmichelson     Note Added: 0106345                          
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