[asterisk-commits] mmichelson: branch 1.8 r376521 - in /branches/1.8/channels: ./ sip/include/

[SVN commits to the Asterisk project]

Author: mmichelson
Date: Tue Nov 20 10:45:50 2012
New Revision: 376521

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=376521
Log:
Add "Require: timer" to 200 OK responses when appropriate.

The method by which the Require header is added to 200 responses is
inspired by the method that Olle Johansson uses in his darjeeling-prack
branch.

(closes issue ASTERISK-20570)
Reported by Matt Jordan, at the behest of Olle Johansson

Review: https://reviewboard.asterisk.org/r/2172


Modified:
    branches/1.8/channels/chan_sip.c
    branches/1.8/channels/sip/include/sip.h

Modified: branches/1.8/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/chan_sip.c?view=diff&rev=376521&r1=376520&r2=376521
==============================================================================
--- branches/1.8/channels/chan_sip.c (original)
+++ branches/1.8/channels/chan_sip.c Tue Nov 20 10:45:50 2012
@@ -4467,6 +4467,34 @@
 		with_sdp ? send_provisional_keepalive_with_sdp : send_provisional_keepalive, dialog_ref(pvt, "Increment refcount to pass dialog pointer to sched callback"));
 }
 
+static void add_required_respheader(struct sip_request *req)
+{
+	struct ast_str *str;
+	int i;
+
+	if (!req->reqsipoptions) {
+		return;
+	}
+
+	str = ast_str_create(32);
+
+	for (i = 0; i < ARRAY_LEN(sip_options); ++i) {
+		if (!(req->reqsipoptions & sip_options[i].id)) {
+			continue;
+		}
+		if (ast_str_strlen(str) > 0) {
+			ast_str_append(&str, 0, ", ");
+		}
+		ast_str_append(&str, 0, "%s", sip_options[i].text);
+	}
+
+	if (ast_str_strlen(str) > 0) {
+		add_header(req, "Require", ast_str_buffer(str));
+	}
+
+	ast_free(str);
+}
+
 /*! \brief Transmit response on SIP request*/
 static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno)
 {
@@ -10652,11 +10680,26 @@
 	add_supported_header(p, resp);
 
 	/* If this is an invite, add Session-Timers related headers if the feature is active for this session */
-	if (p->method == SIP_INVITE && p->stimer && p->stimer->st_active == TRUE && p->stimer->st_active_peer_ua == TRUE) {
+	if (p->method == SIP_INVITE && p->stimer && p->stimer->st_active == TRUE) {
 		char se_hdr[256];
 		snprintf(se_hdr, sizeof(se_hdr), "%d;refresher=%s", p->stimer->st_interval,
 			p->stimer->st_ref == SESSION_TIMER_REFRESHER_US ? "uas" : "uac");
 		add_header(resp, "Session-Expires", se_hdr);
+		/* RFC 2048, Section 9
+		 * If the refresher parameter in the Session-Expires header field in the
+		 * 2xx response has a value of 'uac', the UAS MUST place a Require
+		 * header field into the response with the value 'timer'.
+		 * ...
+		 * If the refresher parameter in
+		 * the 2xx response has a value of 'uas' and the Supported header field
+		 * in the request contained the value 'timer', the UAS SHOULD place a
+		 * Require header field into the response with the value 'timer'
+		 */
+		if (p->stimer->st_ref == SESSION_TIMER_REFRESHER_THEM ||
+				(p->stimer->st_ref == SESSION_TIMER_REFRESHER_US &&
+				 p->stimer->st_active_peer_ua == TRUE)) {
+			resp->reqsipoptions |= SIP_OPT_TIMER;
+		}
 	}
 
 	if (msg[0] == '2' && (p->method == SIP_SUBSCRIBE || p->method == SIP_REGISTER || p->method == SIP_PUBLISH)) {
@@ -12166,6 +12209,7 @@
 		ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);
 	if (reliable && !p->pendinginvite)
 		p->pendinginvite = seqno;		/* Buggy clients sends ACK on RINGING too */
+	add_required_respheader(&resp);
 	return send_response(p, &resp, reliable, seqno);
 }
 
@@ -23359,7 +23403,7 @@
 			st_active = TRUE;
 			st_interval = st_get_se(p, TRUE);
 			tmp_st_ref = SESSION_TIMER_REFRESHER_US;
-			p->stimer->st_active_peer_ua = FALSE;
+			p->stimer->st_active_peer_ua = (p->sipoptions & SIP_OPT_TIMER) ? TRUE : FALSE;
 			break;
 
 		default:

Modified: branches/1.8/channels/sip/include/sip.h
URL: http://svnview.digium.com/svn/asterisk/branches/1.8/channels/sip/include/sip.h?view=diff&rev=376521&r1=376520&r2=376521
==============================================================================
--- branches/1.8/channels/sip/include/sip.h (original)
+++ branches/1.8/channels/sip/include/sip.h Tue Nov 20 10:45:50 2012
@@ -769,6 +769,7 @@
 	/* XXX Do we need to unref socket.ser when the request goes away? */
 	struct sip_socket socket;          /*!< The socket used for this request */
 	AST_LIST_ENTRY(sip_request) next;
+	unsigned int reqsipoptions; /*!< Items needed for Required header in responses */
 };
 
 /* \brief given a sip_request and an offset, return the char * that resides there