[asterisk-bugs] [Asterisk 0016340]: RFC 2833 DTMF Events Generated by Polycom IP Phone Running v3.2.1 F/W Not Recognized by Asterisk 1.4

[Asterisk Bug Tracker]

The following issue has been CLOSED 
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https://issues.asterisk.org/view.php?id=16340 
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Reported By:                Jamuel
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16340
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           1.4.27 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
Resolution:                 open
Fixed in Version:           
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Date Submitted:             2009-11-28 15:04 CST
Last Modified:              2009-11-30 13:42 CST
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Summary:                    RFC 2833 DTMF Events Generated by Polycom IP Phone
Running v3.2.1 F/W  Not Recognized by Asterisk 1.4
Description: 
A Polycom SP IP 450 phone running firmware version 3.2.1 configured with
the default configuration settings generates an rtpmap for DTMF events that
is apparently not recognized by Asterisk.

Run the same phone against Asterisk 1.6.1 no problem.

Run the phone on version 3.1.3 firmware against Asterisk 1.4 or 1.6 and no
problem.

In the SDP exchange we are seeing a different rtpmap # for
telephone-event/8000 generated by the Polycom with the 3.2.1 version of
firmware as compared v3.1.3

I will attach packet captures showing that the in the INVITE's SDP the
SPIP generates rtpmap:127 telephone-event/8000 when running fw v3.2.1 and
rtpmap:101 telephone-event/8000 when running v 3.1.3

v3.1.3 works without issue with Asterisk 1.4.27 but while running the
Polycom on v3.2.1 Asterisk does not seem to recognize the rtmap:127 value
and does not respond in-kind in its SDP.

rtpmap:127 is the default value set in the Polycom configuration files
(sip.cfg) for f/w 3.2.1  <DTMF tone.dtmf.rfc2833Payload="127" /> versus in
v3.1.3 f/w <DTMF tone.dtmf.rfc2833Payload="101" />

I realize this sounds like it's a Polycom issue but if rtpmap:127 is a
valid payload type for 2833 RTP telephony-events, then this is a bug in
Asterisk 1.4

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Relationships       ID      Summary
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duplicate of        0016268 [patch] Last line of SDP is not being p...
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 (0114394) lmadsen (administrator) - 2009-11-30 13:42
 https://issues.asterisk.org/view.php?id=16340#c114394 
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This should be fixed in the next set of releases, due out later today. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-30 13:42 lmadsen        Note Added: 0114394                          
2009-11-30 13:42 lmadsen        Status                   new => closed       
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