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Merged revisions 47333 via svnmerge from

Browse Source 
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r47333 | kpfleming | 2006-11-08 12:07:16 -0600 (Wed, 08 Nov 2006) | 2 lines

add simple fix for SDP to report proper sample rate for G.722 media sessions

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47337 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kevin P. Fleming
2006-11-08 18:28:53 +00:00
parent c5780b19c8
commit 02c6f507bb
1 changed files with 13 additions and 9 deletions
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channels/chan_sip.c
+13 -9
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@@ -6108,6 +6108,8 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo, struct socka
}
#define SDP_SAMPLE_RATE(x) (x == AST_FORMAT_G722) ? 16000 : 8000
/*! \brief Add Session Description Protocol message */
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
{
@@ -6232,31 +6234,33 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
Note that p->prefcodec can include video codecs, so mask them out
*/
if (capability & p->prefcodec) {
add_codec_to_sdp(p, p->prefcodec & AST_FORMAT_AUDIO_MASK, 8000,
int codec = p->prefcodec & AST_FORMAT_AUDIO_MASK;
add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec),
&m_audio_next, &m_audio_left,
&a_audio_next, &a_audio_left,
debug, &min_audio_packet_size);
alreadysent |= p->prefcodec & AST_FORMAT_AUDIO_MASK;
alreadysent |= codec;
}
/* Start by sending our preferred audio codecs */
for (x = 0; x < 32; x++) {
int pref_codec;
int codec;
if (!(pref_codec = ast_codec_pref_index(&p->prefs, x)))
if (!(codec = ast_codec_pref_index(&p->prefs, x)))
break;
if (!(capability & pref_codec))
if (!(capability & codec))
continue;
if (alreadysent & pref_codec)
if (alreadysent & codec)
continue;
add_codec_to_sdp(p, pref_codec, 8000,
add_codec_to_sdp(p, codec, SDP_SAMPLE_RATE(codec),
&m_audio_next, &m_audio_left,
&a_audio_next, &a_audio_left,
debug, &min_audio_packet_size);
alreadysent |= pref_codec;
alreadysent |= codec;
}
/* Now send any other common audio and video codecs, and non-codec formats: */
@@ -6268,7 +6272,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
continue;
if (x <= AST_FORMAT_MAX_AUDIO)
add_codec_to_sdp(p, x, 8000,
add_codec_to_sdp(p, x, SDP_SAMPLE_RATE(x),
&m_audio_next, &m_audio_left,
&a_audio_next, &a_audio_left,
debug, &min_audio_packet_size);