From: David Vossel <dvossel@digium.com>
Date: Tue, 24 Mar 2009 20:01:29 +0000 (+0000)
Subject: SIP preferred codec only feature
X-Git-Tag: 11.0.0-beta1~5189
X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=da2230adf081d932294ffe5f2629ad66cc338917;p=thirdparty%2Fasterisk.git

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/




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@183995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---

diff --git a/CHANGES b/CHANGES
index 404248b7ee..91877d3d03 100644
--- a/CHANGES
+++ b/CHANGES
@@ -12,6 +12,11 @@
 --- 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
 ------------
  * Added progress option to the app_dial D() option.  When progress DTMF is
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 0266070c95..c8a79b8da1 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -1369,6 +1369,7 @@ struct sip_auth {
 /* 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 @@ struct sip_auth {
 	(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 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
 	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 @@ static struct sip_peer *build_peer(const char *name, struct ast_variable *v, str
 			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 @@ static int reload_config(enum channelreloadreason reason)
 			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")) {
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index c7ae9c06fe..a67a2092d1 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -182,6 +182,11 @@ srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
 ;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