From a769766c538372ec28ddd35377047193a58236cf Mon Sep 17 00:00:00 2001
From: Joshua Colp <jcolp@digium.com>
Date: Thu, 17 May 2007 16:11:26 +0000
Subject: [PATCH] Merged revisions 64754 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r64754 | file | 2007-05-17 12:10:12 -0400 (Thu, 17 May 2007) | 2 lines

Even more direct RTP setup fixes! Don't allow a codec that isn't supported to creep into the SDP of either side. (issue #9446 reported by marcelbarbulescu)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64755 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 channels/chan_sip.c | 5 +++++
 1 file changed, 5 insertions(+)

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 7e5499d44e..732e1690ec 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -17960,6 +17960,11 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
 		p->redircodecs = codecs;
 		changed = 1;
 	}
+	if ((p->capability & codecs) != p->capability) {
+		p->jointcapability &= codecs;
+		p->capability &= codecs;
+		changed = 1;
+	}
 	if (changed && !ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
 		if (chan->_state != AST_STATE_UP) {	/* We are in early state */
 			if (!ast_test_flag(&p->flags[0], SIP_NO_HISTORY))