From 26365fdecaae908c099a02f63ed3994fc9f1c1e6 Mon Sep 17 00:00:00 2001 From: Russell Bryant Date: Fri, 1 Feb 2008 23:06:32 +0000 Subject: [PATCH] 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 --- channels/chan_sip.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 13351e7fbd..ff09964e57 100644 --- 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)