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 ...)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@101989 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 13351e7fbd01f825d8d6bbb67dd1a81f131eb1fc..ff09964e57c57b66711ccf2957051710f8edf5e7 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -6346,7 +6346,12 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_
                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)