1 files changed, 107 insertions, 0 deletions
diff --git a/res/res_pjsip_sips_contact.c b/res/res_pjsip_sips_contact.c
new file mode 100644
index 000000000..f8e554cf4
--- /dev/null
+++ b/res/res_pjsip_sips_contact.c
@@ -0,0 +1,107 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2015, Digium, Inc.
+ *
+ * Mark Michelson <mmichelson@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*** MODULEINFO
+	<depend>pjproject</depend>
+	<depend>res_pjsip</depend>
+	<support_level>core</support_level>
+ ***/
+
+#include "asterisk.h"
+
+#include <pjsip.h>
+
+#include "asterisk/res_pjsip.h"
+#include "asterisk/module.h"
+
+/*!
+ * \brief Upgrade Contact URIs on outgoing SIP requests to SIPS if required.
+ *
+ * The rules being used here are according to RFC 3261 section 8.1.1.8. In
+ * brief, if the request URI is SIPS or the topmost Route header is SIPS,
+ * then the Contact header we send must also be SIPS.
+ */
+static pj_status_t sips_contact_on_tx_request(pjsip_tx_data *tdata)
+{
+	pjsip_contact_hdr *contact;
+	pjsip_route_hdr *route;
+	pjsip_sip_uri *contact_uri;
+
+	contact = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_CONTACT, NULL);
+	if (!contact) {
+		return PJ_SUCCESS;
+	}
+
+	contact_uri = pjsip_uri_get_uri(contact->uri);
+	if (PJSIP_URI_SCHEME_IS_SIPS(contact_uri)) {
+		/* If the Contact header is already SIPS, then we don't need to do anything */
+		return PJ_SUCCESS;
+	}
+
+	if (PJSIP_URI_SCHEME_IS_SIPS(tdata->msg->line.req.uri)) {
+		ast_debug(1, "Upgrading contact URI on outgoing SIP request to SIPS due to SIPS Request URI\n");
+		pjsip_sip_uri_set_secure(contact_uri, PJ_TRUE);
+		return PJ_SUCCESS;
+	}
+
+	route = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_ROUTE, NULL);
+	if (!route) {
+		return PJ_SUCCESS;
+	}
+
+	if (!PJSIP_URI_SCHEME_IS_SIPS(&route->name_addr)) {
+		return PJ_SUCCESS;
+	}
+
+	/* Our Contact header is not a SIPS URI, but our topmost Route header is. */
+	ast_debug(1, "Upgrading contact URI on outgoing SIP request to SIPS due to SIPS Route header\n");
+	pjsip_sip_uri_set_secure(contact_uri, PJ_TRUE);
+
+	return PJ_SUCCESS;
+}
+
+static pjsip_module sips_contact_module = {
+	.name = {"SIPS Contact", 12 },
+	.id = -1,
+	.priority = PJSIP_MOD_PRIORITY_TSX_LAYER - 2,
+	.on_tx_request = sips_contact_on_tx_request,
+};
+
+static int unload_module(void)
+{
+	ast_sip_unregister_service(&sips_contact_module);
+	return 0;
+}
+
+static int load_module(void)
+{
+	CHECK_PJSIP_MODULE_LOADED();
+
+	if (ast_sip_register_service(&sips_contact_module)) {
+		return AST_MODULE_LOAD_DECLINE;
+	}
+
+	return AST_MODULE_LOAD_SUCCESS;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "UAC SIPS Contact support",
+		.support_level = AST_MODULE_SUPPORT_CORE,
+		.load = load_module,
+		.unload = unload_module,
+		.load_pri = AST_MODPRI_APP_DEPEND,
+	       );