[asterisk-commits] file: branch 1.6.0 r118561 - in /branches/1.6.0: ./ channels/chan_sip.c

[SVN commits to the Asterisk project]

Author: file
Date: Tue May 27 14:35:47 2008
New Revision: 118561

URL: http://svn.digium.com/view/asterisk?view=rev&rev=118561
Log:
Merged revisions 118560 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
r118560 | file | 2008-05-27 16:34:14 -0300 (Tue, 27 May 2008) | 12 lines

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:
    branches/1.6.0/   (props changed)
    branches/1.6.0/channels/chan_sip.c

Propchange: branches/1.6.0/
------------------------------------------------------------------------------
Binary property 'trunk-merged' - no diff available.

Modified: branches/1.6.0/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.6.0/channels/chan_sip.c?view=diff&rev=118561&r1=118560&r2=118561
==============================================================================
--- branches/1.6.0/channels/chan_sip.c (original)
+++ branches/1.6.0/channels/chan_sip.c Tue May 27 14:35:47 2008
@@ -5792,6 +5792,7 @@
 			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) {
@@ -6656,23 +6657,20 @@
 					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 (sscanf(a, "rtpmap: %u %[^/]/", &codec, mimeSubtype) == 2) {
 			/* We have a rtpmap to handle */