[asterisk-commits] branch 1.2 r12477 - /branches/1.2/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Thu Mar  9 11:00:36 2006
New Revision: 12477

URL: http://svn.digium.com/view/asterisk?rev=12477&view=rev
Log:
Issue #6576 - SIP_CODEC not used for early media (reported by gpapadop73)

Modified:
    branches/1.2/channels/chan_sip.c

Modified: branches/1.2/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_sip.c?rev=12477&r1=12476&r2=12477&view=diff
==============================================================================
--- branches/1.2/channels/chan_sip.c (original)
+++ branches/1.2/channels/chan_sip.c Thu Mar  9 11:00:36 2006
@@ -2483,12 +2483,34 @@
 	return 0;
 }
 
+/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
+static void try_suggested_sip_codec(struct sip_pvt *p)
+{
+	int fmt;
+	char *codec;
+
+	codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
+	if (!codec) 
+		return;
+
+	fmt = ast_getformatbyname(codec);
+	if (fmt) {
+		ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
+		if (p->jointcapability & fmt) {
+			p->jointcapability &= fmt;
+			p->capability &= fmt;
+		} else
+			ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
+	} else
+		ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
+	return;	
+}
+
 /*! \brief  sip_answer: Answer SIP call , send 200 OK on Invite 
  * Part of PBX interface */
 static int sip_answer(struct ast_channel *ast)
 {
-	int res = 0,fmt;
-	char *codec;
+	int res = 0;
 	struct sip_pvt *p = ast->tech_pvt;
 
 	ast_mutex_lock(&p->lock);
@@ -2496,19 +2518,7 @@
 #ifdef OSP_SUPPORT	
 		time(&p->ospstart);
 #endif
-	
-		codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC");
-		if (codec) {
-			fmt=ast_getformatbyname(codec);
-			if (fmt) {
-				ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
-				if (p->jointcapability & fmt) {
-					p->jointcapability &= fmt;
-					p->capability &= fmt;
-				} else
-					ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
-			} else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
-		}
+		try_suggested_sip_codec(p);	
 
 		ast_setstate(ast, AST_STATE_UP);
 		if (option_debug)
@@ -4514,6 +4524,7 @@
 	respprep(&resp, p, msg, req);
 	if (p->rtp) {
 		ast_rtp_offered_from_local(p->rtp, 0);
+		try_suggested_sip_codec(p);	
 		add_sdp(&resp, p);
 	} else {
 		ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);