Merged revisions 64754 via svnmerge from

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

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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)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64755 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 7e5499d44e95325561e49922bb7d257d8f2155ce..732e1690ec55f7f38bd9db65a34e020d652600d6 100644 (file)
--- 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))