asterisk/res/res_pjsip_sips_contact.c

107 lines
2.8 KiB
C

/*
* 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",
);