[svn-commits] mmichelson: branch group/issue8824 r184025 - in /team/group/issue8824: ./ cha...

[SVN commits to the Digium repositories]

Author: mmichelson
Date: Tue Mar 24 15:42:10 2009
New Revision: 184025

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=184025
Log:
Resolve automerge and reset conflict.


Modified:
    team/group/issue8824/   (props changed)
    team/group/issue8824/CHANGES
    team/group/issue8824/channels/chan_sip.c
    team/group/issue8824/configs/sip.conf.sample

Propchange: team/group/issue8824/
------------------------------------------------------------------------------
    automerge = *

Propchange: team/group/issue8824/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Mar 24 15:42:10 2009
@@ -1,1 +1,1 @@
-/trunk:1-183920
+/trunk:1-184024

Modified: team/group/issue8824/CHANGES
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue8824/CHANGES?view=diff&rev=184025&r1=184024&r2=184025
==============================================================================
--- team/group/issue8824/CHANGES (original)
+++ team/group/issue8824/CHANGES Tue Mar 24 15:42:10 2009
@@ -10,6 +10,11 @@
 ------------------------------------------------------------------------------
 --- Functionality changes from Asterisk 1.6.2 to Asterisk 1.6.3  -------------
 ------------------------------------------------------------------------------
+
+SIP Changes
+-----------
+ * Added preferred_codec_only option in sip.conf. This feature limits the joint
+   codecs sent in response to an INVITE to the single most preferred codec.
 
 Applications
 ------------

Modified: team/group/issue8824/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue8824/channels/chan_sip.c?view=diff&rev=184025&r1=184024&r2=184025
==============================================================================
--- team/group/issue8824/channels/chan_sip.c (original)
+++ team/group/issue8824/channels/chan_sip.c Tue Mar 24 15:42:10 2009
@@ -1423,6 +1423,7 @@
 #define SIP_PAGE2_CONNECTLINEUPDATE_PEND		(1 << 10)
 #define SIP_PAGE2_RPID_IMMEDIATE			(1 << 11)
 
+#define SIP_PAGE2_PREFERRED_CODEC	(1 << 13)	/*!< GDP: Only respond with single most preferred joint codec */
 #define SIP_PAGE2_VIDEOSUPPORT		(1 << 14)	/*!< DP: Video supported if offered? */
 #define SIP_PAGE2_TEXTSUPPORT		(1 << 15)	/*!< GDP: Global text enable */
 #define SIP_PAGE2_ALLOWSUBSCRIBE	(1 << 16)	/*!< GP: Allow subscriptions from this peer? */
@@ -1452,7 +1453,8 @@
 	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_IGNORESDPVERSION | \
 	SIP_PAGE2_VIDEOSUPPORT | SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | \
 	SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_TEXTSUPPORT | SIP_PAGE2_FAX_DETECT | \
-	SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_RPID_IMMEDIATE)
+	SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC | \
+	SIP_PAGE2_RPID_IMMEDIATE)
 
 /*@}*/ 
 
@@ -8129,11 +8131,15 @@
 	p->peercapability = newpeercapability;		        /* The other sides capability in latest offer */
 	p->jointnoncodeccapability = newnoncodeccapability;	/* DTMF capabilities */
 
+	if (ast_test_flag(&p->flags[1], SIP_PAGE2_PREFERRED_CODEC)) { /* respond with single most preferred joint codec, limiting the other side's choice */
+		p->jointcapability = ast_codec_choose(&p->prefs, p->jointcapability, 1);
+	}
+
 	if (p->jointcapability & AST_FORMAT_T140RED) {
-		p->red = 1; 
+		p->red = 1;
 		ast_rtp_red_init(p->trtp, 300, red_data_pt, 2);
 	} else {
-		p->red = 0; 
+		p->red = 0;
 	}
 
 	ast_rtp_pt_copy(p->rtp, newaudiortp);
@@ -23276,6 +23282,8 @@
 			int error =  ast_parse_allow_disallow(&peer->prefs, &peer->capability, v->value, FALSE);
 			if (error)
 				ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
+		} else if (!strcasecmp(v->name, "preferred_codec_only")) {
+			ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_PREFERRED_CODEC);
 		} else if (!strcasecmp(v->name, "registertrying")) {
 			ast_set2_flag(&peer->flags[1], ast_true(v->value), SIP_PAGE2_REGISTERTRYING);
 		} else if (!strcasecmp(v->name, "autoframing")) {
@@ -24011,6 +24019,8 @@
 			int error =  ast_parse_allow_disallow(&default_prefs, &global_capability, v->value, FALSE);
 			if (error)
 				ast_log(LOG_WARNING, "Codec configuration errors found in line %d : %s = %s\n", v->lineno, v->name, v->value);
+		} else if (!strcasecmp(v->name, "preferred_codec_only")) {
+			ast_set2_flag(&global_flags[1], ast_true(v->value), SIP_PAGE2_PREFERRED_CODEC);
 		} else if (!strcasecmp(v->name, "autoframing")) {
 			global_autoframing = ast_true(v->value);
 		} else if (!strcasecmp(v->name, "allowexternaldomains")) {

Modified: team/group/issue8824/configs/sip.conf.sample
URL: http://svn.digium.com/svn-view/asterisk/team/group/issue8824/configs/sip.conf.sample?view=diff&rev=184025&r1=184024&r2=184025
==============================================================================
--- team/group/issue8824/configs/sip.conf.sample (original)
+++ team/group/issue8824/configs/sip.conf.sample Tue Mar 24 15:42:10 2009
@@ -182,6 +182,11 @@
 ;vmexten=voicemail              ; dialplan extension to reach mailbox sets the 
                                 ; Message-Account in the MWI notify message 
                                 ; defaults to "asterisk"
+
+;preferred_codec_only=yes       ; Respond to a SIP invite with the single most preferred codec
+                                ; rather than advertising all joint codec capabilities. This
+                                ; limits the other side's codec choice to exactly what we prefer.
+
 ;disallow=all                   ; First disallow all codecs
 ;allow=ulaw                     ; Allow codecs in order of preference
 ;allow=ilbc                     ; see doc/rtp-packetization for framing options