[asterisk-commits] russell: trunk r101990 - in /trunk: ./ channels/chan_sip.c

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Fri Feb 1 17:08:29 CST 2008


Author: russell
Date: Fri Feb  1 17:08:28 2008
New Revision: 101990

URL: http://svn.digium.com/view/asterisk?view=rev&rev=101990
Log:
Merged revisions 101989 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008) | 5 lines

Change the SDP_SAMPLE_RATE macro.  It turns out that even though G.722 is 16 kHz,
it is supposed to specified as 8 kHz in the RTP, and RTP timestamps are supposed
to be calculated based on 8 kHz.  (Apparently this is due to a bug in a spec, but
people follow it anyway, because it's the spec ...)

........

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=101990&r1=101989&r2=101990
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Feb  1 17:08:28 2008
@@ -7758,7 +7758,12 @@
 
 }
 
-#define SDP_SAMPLE_RATE(x) (x == AST_FORMAT_G722) ? 16000 : 8000
+/*!
+ * \note G.722 actually is supposed to specified as 8 kHz, even though it is
+ * really 16 kHz.  Update this macro for other formats as they are added in
+ * the future.
+ */
+#define SDP_SAMPLE_RATE(x) 8000
 
 /*! \brief Add Session Description Protocol message 
 




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