res_pjsip_sips_contact.c

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/*
 * 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)
{
    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,
    .requires = "res_pjsip",
);