[asterisk-commits] dvossel: trunk r183995 - in /trunk: ./ channels/ configs/

[SVN commits to the Asterisk project]

Author: dvossel
Date: Tue Mar 24 15:01:29 2009
New Revision: 183995

URL: http://svn.digium.com/svn-view/asterisk?view=rev&rev=183995
Log:
SIP preferred codec only feature

Added an option to respond to a SIP invite with only the single most preferred joint codec.  This limits the options of what codecs the other side can use.

(closes issue #12485)
Reported by: bamby
Review: http://reviewboard.digium.com/r/206/



Modified:
    trunk/CHANGES
    trunk/channels/chan_sip.c
    trunk/configs/sip.conf.sample

Modified: trunk/CHANGES
URL: http://svn.digium.com/svn-view/asterisk/trunk/CHANGES?view=diff&rev=183995&r1=183994&r2=183995
==============================================================================
--- trunk/CHANGES (original)
+++ trunk/CHANGES Tue Mar 24 15:01:29 2009
@@ -11,6 +11,11 @@
 ------------------------------------------------------------------------------
 --- 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: trunk/channels/chan_sip.c
URL: http://svn.digium.com/svn-view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=183995&r1=183994&r2=183995
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Mar 24 15:01:29 2009
@@ -1369,6 +1369,7 @@
 /* Space for addition of other realtime flags in the future */
 #define SIP_PAGE2_STATECHANGEQUEUE	(1 << 9)	/*!< D: Unsent state pending change exists */
 
+#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? */
@@ -1398,7 +1399,7 @@
 	(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_UDPTL_DESTINATION | SIP_PAGE2_VIDEOSUPPORT_ALWAYS | SIP_PAGE2_PREFERRED_CODEC)
 
 /*@}*/ 
 
@@ -8064,11 +8065,15 @@
 	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);
@@ -22697,6 +22702,8 @@
 			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")) {
@@ -23432,6 +23439,8 @@
 			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: trunk/configs/sip.conf.sample
URL: http://svn.digium.com/svn-view/asterisk/trunk/configs/sip.conf.sample?view=diff&rev=183995&r1=183994&r2=183995
==============================================================================
--- trunk/configs/sip.conf.sample (original)
+++ trunk/configs/sip.conf.sample Tue Mar 24 15:01:29 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