diff options
author | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-02-01 23:06:32 +0000 |
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committer | russell <russell@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-02-01 23:06:32 +0000 |
commit | d76b2bde7902b2bee69c5a1ca18a94b1ac35c4fa (patch) | |
tree | c469f2f2c67f7fbc1cb852a50ce189422331370e | |
parent | e1115249eb2e81adda75c83d77f9621fa491525c (diff) |
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: http://svn.digium.com/svn/asterisk/branches/1.4@101989 f38db490-d61c-443f-a65b-d21fe96a405b
-rw-r--r-- | channels/chan_sip.c | 7 |
1 files changed, 6 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 13351e7fb..ff09964e5 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) |