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 ...)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101990 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index ee9fb33..1332a8e 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -7758,7 +7758,12 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo,
 
 }
 
-#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