[asterisk-commits] russell: trunk r60011 - in /trunk: channels/ doc/

[asterisk-commits at lists.digium.com]

Author: russell
Date: Tue Apr  3 17:33:03 2007
New Revision: 60011

URL: http://svn.digium.com/view/asterisk?view=rev&rev=60011
Log:
Add support for RTP packetization in chan_jingle and chan_gtalk.
(issue #9416, phsultan)

Modified:
    trunk/channels/chan_gtalk.c
    trunk/channels/chan_jingle.c
    trunk/doc/rtp-packetization.txt

Modified: trunk/channels/chan_gtalk.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_gtalk.c?view=diff&rev=60011&r1=60010&r2=60011
==============================================================================
--- trunk/channels/chan_gtalk.c (original)
+++ trunk/channels/chan_gtalk.c Tue Apr  3 17:33:03 2007
@@ -857,6 +857,9 @@
 	if (!(tmp = ast_calloc(1, sizeof(*tmp)))) {
 		return NULL;
 	}
+
+	memcpy(&tmp->prefs, &client->prefs, sizeof(struct ast_codec_pref));
+
 	if (sid) {
 		ast_copy_string(tmp->sid, sid, sizeof(tmp->sid));
 		ast_copy_string(tmp->them, them, sizeof(tmp->them));
@@ -917,6 +920,11 @@
 		what = i->capability;
 	else
 		what = global_capability;
+
+	/* Set Frame packetization */
+	if (i->rtp)
+		ast_rtp_codec_setpref(i->rtp, &i->prefs);
+
 	tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
 	fmt = ast_best_codec(tmp->nativeformats);
 

Modified: trunk/channels/chan_jingle.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_jingle.c?view=diff&rev=60011&r1=60010&r2=60011
==============================================================================
--- trunk/channels/chan_jingle.c (original)
+++ trunk/channels/chan_jingle.c Tue Apr  3 17:33:03 2007
@@ -726,6 +726,9 @@
 	if (!(tmp = ast_calloc(1, sizeof(*tmp)))) {
 		return NULL;
 	}
+
+	memcpy(&tmp->prefs, &client->prefs, sizeof(tmp->prefs));
+
 	if (sid) {
 		ast_copy_string(tmp->sid, sid, sizeof(tmp->sid));
 		ast_copy_string(tmp->from, from, sizeof(tmp->from));
@@ -777,6 +780,11 @@
 		what = i->capability;
 	else
 		what = global_capability;
+
+	/* Set Frame packetization */
+	if (i->rtp)
+		ast_rtp_codec_setpref(i->rtp, &i->prefs);
+
 	tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | (i->jointcapability & AST_FORMAT_VIDEO_MASK);
 	fmt = ast_best_codec(tmp->nativeformats);
 

Modified: trunk/doc/rtp-packetization.txt
URL: http://svn.digium.com/view/asterisk/trunk/doc/rtp-packetization.txt?view=diff&rev=60011&r1=60010&r2=60011
==============================================================================
--- trunk/doc/rtp-packetization.txt (original)
+++ trunk/doc/rtp-packetization.txt Tue Apr  3 17:33:03 2007
@@ -11,6 +11,8 @@
     chan_skinny
     chan_h323
     chan_ooh323 (Asterisk-Addons)
+    chan_gtalk
+    chan_jingle
 
 Configuration
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