[asterisk-commits] russell: branch 1.4 r101989 - /branches/1.4/channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: russell
Date: Fri Feb  1 17:06:32 2008
New Revision: 101989

URL: http://svn.digium.com/view/asterisk?view=rev&rev=101989
Log:
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:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?view=diff&rev=101989&r1=101988&r2=101989
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Fri Feb  1 17:06:32 2008
@@ -6346,7 +6346,12 @@
 		ast_build_string(a_buf, a_size, "a=fmtp:%d 0-16\r\n", rtp_code);
 }
 
-#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 */
 static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)