[asterisk-commits] file: branch 12 r401499 - in /branches/12: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Wed Oct 23 06:14:49 2013
New Revision: 401499

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=401499
Log:
Fix an issue where an incompatible audio format may be added to SDP.

If preferred codecs included any non-audio format the code would
mistakenly add the audio format, even if it was not a joint capability
with the remote side.

(closes issue ASTERISK-21131)
Reported by: nbougues
Patches:
	patch_unsupported_codec_1.8.patch uploaded by nbougues (license 6470)
........

Merged revisions 401497 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 401498 from http://svn.asterisk.org/svn/asterisk/branches/11

Modified:
    branches/12/   (props changed)
    branches/12/channels/chan_sip.c

Propchange: branches/12/
------------------------------------------------------------------------------
Binary property 'branch-11-merged' - no diff available.

Modified: branches/12/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/12/channels/chan_sip.c?view=diff&rev=401499&r1=401498&r2=401499
==============================================================================
--- branches/12/channels/chan_sip.c (original)
+++ branches/12/channels/chan_sip.c Wed Oct 23 06:14:49 2013
@@ -13358,10 +13358,11 @@
 		/* Unless otherwise configured, the prefcaps is added before the peer's
 		 * configured codecs.
 		 */
-		if (!ast_test_flag(&p->flags[2], SIP_PAGE3_IGNORE_PREFCAPS) && ast_format_cap_has_joint(tmpcap, p->prefcaps)) {
+		if (!ast_test_flag(&p->flags[2], SIP_PAGE3_IGNORE_PREFCAPS)) {
 			ast_format_cap_iter_start(p->prefcaps);
 			while (!(ast_format_cap_iter_next(p->prefcaps, &tmp_fmt))) {
-				if (AST_FORMAT_GET_TYPE(tmp_fmt.id) != AST_FORMAT_TYPE_AUDIO) {
+				if (AST_FORMAT_GET_TYPE(tmp_fmt.id) != AST_FORMAT_TYPE_AUDIO ||
+					!ast_format_cap_iscompatible(tmpcap, &tmp_fmt)) {
 					continue;
 				}
 				add_codec_to_sdp(p, &tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size);