From 208f3468787c0840a48c1485229465283b7a7d77 Mon Sep 17 00:00:00 2001 From: Russell Bryant Date: Fri, 1 Feb 2008 23:08:28 +0000 Subject: [PATCH] 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 ...) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@101990 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 ee9fb33..1332a8e 100644 --- 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 -- 1.7.9.5