[asterisk-commits] rizzo: trunk r43479 - /trunk/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: rizzo
Date: Fri Sep 22 09:56:21 2006
New Revision: 43479

URL: http://svn.digium.com/view/asterisk?rev=43479&view=rev
Log:
style fix:
move variable declaration at the beginning of the block.


Modified:
    trunk/channels/chan_sip.c

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=43479&r1=43478&r2=43479&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Fri Sep 22 09:56:21 2006
@@ -4531,6 +4531,9 @@
 	int numberofmediastreams = 0;
 	int debug = sip_debug_test_pvt(p);
 		
+	int found_rtpmap_codecs[32];
+	int last_rtpmap_codec=0;
+
 	if (!p->rtp) {
 		ast_log(LOG_ERROR, "Got SDP but have no RTP session allocated.\n");
 		return -1;
@@ -4702,8 +4705,6 @@
 	 */
 	/* XXX This needs to be done per media stream, since it's media stream specific */
 	iterator = req->sdp_start;
-	int found_rtpmap_codecs[32];
-	int last_rtpmap_codec=0;
 	while ((a = get_sdp_iterate(&iterator, req, "a"))[0] != '\0') {
 		char* mimeSubtype = ast_strdupa(a); /* ensures we have enough space */
 		if (option_debug > 1) {