[asterisk-bugs] [Asterisk 0010331]: [patch] PCMA/16000 and PCMU/16000 support (hd telephony)

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10331 
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Reported By:                guido-r
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10331
Category:                   Channels/chan_sip/CodecHandling
Reproducibility:            N/A
Severity:                   feature
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.9  
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 77765 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             07-30-2007 04:34 CDT
Last Modified:              12-11-2007 02:57 CST
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Summary:                    [patch] PCMA/16000 and PCMU/16000 support (hd
telephony)
Description: 
With this patch PCMA/16000 and PCMU/16000 will be accepted during the SDP
negotiation. Prior asterisk ignored the sample rate in the rtpmap lines of
the SDP body and changed the 16000 to 8000.. and so the negotiation between
the UAs failed.
Now the sample rate is parsed too, and for now PCMA/16000 and PCMU/16000
are supported by the new option flag AST_OPT_16000.

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---------------------------------------------------------------------- 
 oej - 12-11-07 02:57  
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You are wrong. Asterisk sets up the call between each phone and asterisk
and needs to be able to handle the call in the case of hold, transfers and
other situations. After we have a call setup, we can reinvite the call
directly between the end points.

There are ways to handle codecs in passthrough mode, but then we need to
be able to declare them as frame types too. 

Issue History 
Date Modified   Username       Field                    Change               
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12-11-07 02:57  oej            Note Added: 0075183                          
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