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