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

noreply at bugs.digium.com noreply at bugs.digium.com
Mon Jan 21 07:59:11 CST 2008


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:              01-21-2008 07:59 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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---------------------------------------------------------------------- 
 guido-r - 01-21-08 07:59  
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One of the first devices is the AVM Fritz!Mini
SIP Servers are for example dus.net or servers that do not touch the sdp
body and leave the negotiation to the end devices.

Nice to hear that asterisk now supports 16khz audio.
Maybe this thing can be redone. 

Issue History 
Date Modified   Username       Field                    Change               
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01-21-08 07:59  guido-r        Note Added: 0080934                          
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