[asterisk-commits] file: trunk r118560 - in /trunk: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Tue May 27 14:34:14 2008
New Revision: 118560

URL: http://svn.digium.com/view/asterisk?view=rev&rev=118560
Log:
Merged revisions 118558 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 lines

Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP.
(closes issue #12501)
Reported by: slimey

........

Modified:
    trunk/   (props changed)
    trunk/channels/chan_sip.c

Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.

Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=118560&r1=118559&r2=118560
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue May 27 14:34:14 2008
@@ -6078,6 +6078,7 @@
 			ast_udptl_setqos(p->udptl, global_tos_audio, global_cos_audio);
 		p->maxcallbitrate = default_maxcallbitrate;
 		p->autoframing = global_autoframing;
+		ast_rtp_codec_setpref(p->rtp, &p->prefs);
 	}
 
 	if (useglobal_nat && sin) {
@@ -6989,23 +6990,20 @@
 					ast_debug(1, "Can't read framing from SDP: %s\n", a);
 				}
 			}
-			if (framing && last_rtpmap_codec) {
-				if (p->autoframing) {
-					struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
-					int codec_n;
-					int format = 0;
-					for (codec_n = 0; codec_n < last_rtpmap_codec; codec_n++) {
-						format = ast_rtp_codec_getformat(found_rtpmap_codecs[codec_n]);
-						if (!format)	/* non-codec or not found */
-							continue;
-						ast_debug(1, "Setting framing for %d to %ld\n", format, framing);
-						ast_codec_pref_setsize(pref, format, framing);
-					}
-					ast_rtp_codec_setpref(p->rtp, pref);
+			if (framing && p->autoframing) {
+				struct ast_codec_pref *pref = ast_rtp_codec_getpref(p->rtp);
+				int codec_n;
+				int format = 0;
+				for (codec_n = 0; codec_n < MAX_RTP_PT; codec_n++) {
+					format = ast_rtp_codec_getformat(codec_n);
+					if (!format)	/* non-codec or not found */
+						continue;
+					if (option_debug)
+						ast_log(LOG_DEBUG, "Setting framing for %d to %ld\n", format, framing);
+					ast_codec_pref_setsize(pref, format, framing);
 				}
+				ast_rtp_codec_setpref(p->rtp, pref);
 			}
-			memset(&found_rtpmap_codecs, 0, sizeof(found_rtpmap_codecs));
-			last_rtpmap_codec = 0;
 			continue;
 			
 		} else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {