From: Kevin P. Fleming <kpfleming@digium.com>
Date: Wed, 8 Nov 2006 18:07:16 +0000 (+0000)
Subject: add simple fix for SDP to report proper sample rate for G.722 media sessions
X-Git-Tag: 1.4.0-beta4~179
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=10978d0390b5f69cd4e3d47585e182e46daa0515;p=thirdparty%2Fasterisk.git

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


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@47333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index ff915c70ff..2064fe6ff7 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -5976,6 +5976,8 @@ 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
+
 /*! \brief Add Session Description Protocol message */
 static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
 {
@@ -6119,31 +6121,33 @@ 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: */
@@ -6155,7 +6159,7 @@ 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);