asterisk/res/res_pjsip_transport_websock...

457 lines
14 KiB
C
Raw Normal View History

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2013, Digium, Inc.
*
* Jason Parker <jparker@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.
*/
/*!
* \brief WebSocket transport module
*/
/*** MODULEINFO
<depend>pjproject</depend>
<depend>res_pjsip</depend>
<depend>res_http_websocket</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <pjsip.h>
#include <pjsip_ua.h>
#include "asterisk/module.h"
#include "asterisk/http_websocket.h"
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/taskprocessor.h"
static int transport_type_ws;
static int transport_type_wss;
/*!
* \brief Wrapper for pjsip_transport, for storing the WebSocket session
*/
struct ws_transport {
pjsip_transport transport;
pjsip_rx_data rdata;
struct ast_websocket *ws_session;
};
/*!
* \brief Send a message over the WebSocket connection.
*
* Called by pjsip transport manager.
*/
static pj_status_t ws_send_msg(pjsip_transport *transport,
pjsip_tx_data *tdata,
const pj_sockaddr_t *rem_addr,
int addr_len,
void *token,
pjsip_transport_callback callback)
{
struct ws_transport *wstransport = (struct ws_transport *)transport;
res_http_websocket: Avoid passing strlen() to ast_websocket_write(). We have seen a rash of test failures on a 32-bit build agent. Commit 48698a5e21d7307f61b5fb2bd39fd593bc1423ca solved an obvious problem where we were not encoding a 64-bit value correctly over the wire. This commit, however, did not solve the test failures. In the failing tests, ARI is attempting to send a 537 byte text frame over a websocket. When sending a frame this small, 16 bits are all that is required in order to encode the payload length on the websocket frame. However, ast_websocket_write() thinks that the payload length is greater than 65535 and therefore writes out a 64 bit payload length. Inspecting this payload length, the lower 32 bits are exactly what we would expect it to be, 537 in hex. The upper 32 bits, are junk values that are not expected to be there. In the failure, we are passing the result of strlen() to a function that expects a uint64_t parameter to be passed in. strlen() returns a size_t, which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit unsigned value to somewhere where a 64-bit unsigned value is expected would cause no problems. In fact, in manual runs of failing tests, this works just fine. However, ast_websocket_write() uses the Asterisk optional API, which means that rather than a simple function call, there are a series of macros that are used for its declaration and implementation. These macros may be causing some sort of error to occur when converting from a 32 bit quantity to a 64 bit quantity. This commit changes the logic by making existing ast_websocket_write() calls use ast_websocket_write_string() instead. Within ast_websocket_write_string(), the 64-bit converted strlen is saved in a local variable, and that variable is passed to ast_websocket_write() instead. Note that this commit message is full of speculation rather than certainty. This is because the observed test failures, while always present in automated test runs, never occur when tests are manually attempted on the same test agent. The idea behind this commit is to fix a theoretical issue by performing changes that should, at the least, cause no harm. If it turns out that this change does not fix the failing tests, then this commit should be reverted. Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67
2015-08-03 16:06:07 +00:00
uint64_t len = tdata->buf.cur - tdata->buf.start;
res_http_websocket: Avoid passing strlen() to ast_websocket_write(). We have seen a rash of test failures on a 32-bit build agent. Commit 48698a5e21d7307f61b5fb2bd39fd593bc1423ca solved an obvious problem where we were not encoding a 64-bit value correctly over the wire. This commit, however, did not solve the test failures. In the failing tests, ARI is attempting to send a 537 byte text frame over a websocket. When sending a frame this small, 16 bits are all that is required in order to encode the payload length on the websocket frame. However, ast_websocket_write() thinks that the payload length is greater than 65535 and therefore writes out a 64 bit payload length. Inspecting this payload length, the lower 32 bits are exactly what we would expect it to be, 537 in hex. The upper 32 bits, are junk values that are not expected to be there. In the failure, we are passing the result of strlen() to a function that expects a uint64_t parameter to be passed in. strlen() returns a size_t, which on this 32-bit machine is 32 bits wide. Normally, passing a 32-bit unsigned value to somewhere where a 64-bit unsigned value is expected would cause no problems. In fact, in manual runs of failing tests, this works just fine. However, ast_websocket_write() uses the Asterisk optional API, which means that rather than a simple function call, there are a series of macros that are used for its declaration and implementation. These macros may be causing some sort of error to occur when converting from a 32 bit quantity to a 64 bit quantity. This commit changes the logic by making existing ast_websocket_write() calls use ast_websocket_write_string() instead. Within ast_websocket_write_string(), the 64-bit converted strlen is saved in a local variable, and that variable is passed to ast_websocket_write() instead. Note that this commit message is full of speculation rather than certainty. This is because the observed test failures, while always present in automated test runs, never occur when tests are manually attempted on the same test agent. The idea behind this commit is to fix a theoretical issue by performing changes that should, at the least, cause no harm. If it turns out that this change does not fix the failing tests, then this commit should be reverted. Change-Id: I4458dd87d785ca322b89c152b223a540a3d23e67
2015-08-03 16:06:07 +00:00
if (ast_websocket_write(wstransport->ws_session, AST_WEBSOCKET_OPCODE_TEXT, tdata->buf.start, len)) {
return PJ_EUNKNOWN;
}
return PJ_SUCCESS;
}
/*!
* \brief Destroy the pjsip transport.
*
* Called by pjsip transport manager.
*/
static pj_status_t ws_destroy(pjsip_transport *transport)
{
struct ws_transport *wstransport = (struct ws_transport *)transport;
int fd = ast_websocket_fd(wstransport->ws_session);
if (fd > 0) {
ast_websocket_close(wstransport->ws_session, 1000);
shutdown(fd, SHUT_RDWR);
}
ao2_ref(wstransport, -1);
return PJ_SUCCESS;
}
static void transport_dtor(void *arg)
{
struct ws_transport *wstransport = arg;
if (wstransport->ws_session) {
ast_websocket_unref(wstransport->ws_session);
}
if (wstransport->transport.ref_cnt) {
pj_atomic_destroy(wstransport->transport.ref_cnt);
}
if (wstransport->transport.lock) {
pj_lock_destroy(wstransport->transport.lock);
}
if (wstransport->transport.endpt && wstransport->transport.pool) {
pjsip_endpt_release_pool(wstransport->transport.endpt, wstransport->transport.pool);
}
if (wstransport->rdata.tp_info.pool) {
pjsip_endpt_release_pool(wstransport->transport.endpt, wstransport->rdata.tp_info.pool);
}
}
static int transport_shutdown(void *data)
{
struct ws_transport *wstransport = data;
if (!wstransport->transport.is_shutdown && !wstransport->transport.is_destroying) {
pjsip_transport_shutdown(&wstransport->transport);
}
/* Note that the destructor calls PJSIP functions,
* therefore it must be called in a PJSIP thread.
*/
ao2_ref(wstransport, -1);
return 0;
}
struct transport_create_data {
struct ws_transport *transport;
struct ast_websocket *ws_session;
};
/*!
* \brief Create a pjsip transport.
*/
static int transport_create(void *data)
{
struct transport_create_data *create_data = data;
struct ws_transport *newtransport = NULL;
pjsip_endpoint *endpt = ast_sip_get_pjsip_endpoint();
struct pjsip_tpmgr *tpmgr = pjsip_endpt_get_tpmgr(endpt);
pj_pool_t *pool;
pj_str_t buf;
pj_status_t status;
newtransport = ao2_t_alloc_options(sizeof(*newtransport), transport_dtor,
AO2_ALLOC_OPT_LOCK_NOLOCK, "pjsip websocket transport");
if (!newtransport) {
ast_log(LOG_ERROR, "Failed to allocate WebSocket transport.\n");
goto on_error;
}
newtransport->transport.endpt = endpt;
if (!(pool = pjsip_endpt_create_pool(endpt, "ws", 512, 512))) {
ast_log(LOG_ERROR, "Failed to allocate WebSocket endpoint pool.\n");
goto on_error;
}
newtransport->transport.pool = pool;
newtransport->ws_session = create_data->ws_session;
/* Keep the session until transport dies */
ast_websocket_ref(newtransport->ws_session);
status = pj_atomic_create(pool, 0, &newtransport->transport.ref_cnt);
if (status != PJ_SUCCESS) {
goto on_error;
}
status = pj_lock_create_recursive_mutex(pool, pool->obj_name, &newtransport->transport.lock);
if (status != PJ_SUCCESS) {
goto on_error;
}
pj_sockaddr_parse(pj_AF_UNSPEC(), 0, pj_cstr(&buf, ast_sockaddr_stringify(ast_websocket_remote_address(newtransport->ws_session))), &newtransport->transport.key.rem_addr);
newtransport->transport.key.rem_addr.addr.sa_family = pj_AF_INET();
newtransport->transport.key.type = ast_websocket_is_secure(newtransport->ws_session) ? transport_type_wss : transport_type_ws;
newtransport->transport.addr_len = pj_sockaddr_get_len(&newtransport->transport.key.rem_addr);
pj_sockaddr_cp(&newtransport->transport.local_addr, &newtransport->transport.key.rem_addr);
newtransport->transport.local_name.host.ptr = (char *)pj_pool_alloc(pool, newtransport->transport.addr_len+4);
pj_sockaddr_print(&newtransport->transport.key.rem_addr, newtransport->transport.local_name.host.ptr, newtransport->transport.addr_len+4, 0);
newtransport->transport.local_name.host.slen = pj_ansi_strlen(newtransport->transport.local_name.host.ptr);
newtransport->transport.local_name.port = pj_sockaddr_get_port(&newtransport->transport.key.rem_addr);
newtransport->transport.type_name = (char *)pjsip_transport_get_type_name(newtransport->transport.key.type);
newtransport->transport.flag = pjsip_transport_get_flag_from_type((pjsip_transport_type_e)newtransport->transport.key.type);
newtransport->transport.info = (char *)pj_pool_alloc(newtransport->transport.pool, 64);
newtransport->transport.tpmgr = tpmgr;
newtransport->transport.send_msg = &ws_send_msg;
newtransport->transport.destroy = &ws_destroy;
status = pjsip_transport_register(newtransport->transport.tpmgr,
(pjsip_transport *)newtransport);
if (status != PJ_SUCCESS) {
goto on_error;
}
/* Add a reference for pjsip transport manager */
ao2_ref(newtransport, +1);
newtransport->rdata.tp_info.transport = &newtransport->transport;
newtransport->rdata.tp_info.pool = pjsip_endpt_create_pool(endpt, "rtd%p",
PJSIP_POOL_RDATA_LEN, PJSIP_POOL_RDATA_INC);
if (!newtransport->rdata.tp_info.pool) {
ast_log(LOG_ERROR, "Failed to allocate WebSocket rdata.\n");
pjsip_transport_destroy((pjsip_transport *)newtransport);
goto on_error;
}
create_data->transport = newtransport;
return 0;
on_error:
ao2_cleanup(newtransport);
return -1;
}
struct transport_read_data {
struct ws_transport *transport;
char *payload;
uint64_t payload_len;
};
/*!
* \brief Pass WebSocket data into pjsip transport manager.
*/
static int transport_read(void *data)
{
struct transport_read_data *read_data = data;
struct ws_transport *newtransport = read_data->transport;
struct ast_websocket *session = newtransport->ws_session;
pjsip_rx_data *rdata = &newtransport->rdata;
int recvd;
pj_str_t buf;
int pjsip_pkt_len;
pj_gettimeofday(&rdata->pkt_info.timestamp);
pjsip_pkt_len = PJSIP_MAX_PKT_LEN < read_data->payload_len ? PJSIP_MAX_PKT_LEN : read_data->payload_len;
pj_memcpy(rdata->pkt_info.packet, read_data->payload, pjsip_pkt_len);
rdata->pkt_info.len = pjsip_pkt_len;
rdata->pkt_info.zero = 0;
pj_sockaddr_parse(pj_AF_UNSPEC(), 0, pj_cstr(&buf, ast_sockaddr_stringify(ast_websocket_remote_address(session))), &rdata->pkt_info.src_addr);
rdata->pkt_info.src_addr.addr.sa_family = pj_AF_INET();
rdata->pkt_info.src_addr_len = sizeof(rdata->pkt_info.src_addr);
pj_ansi_strcpy(rdata->pkt_info.src_name, ast_sockaddr_stringify_host(ast_websocket_remote_address(session)));
rdata->pkt_info.src_port = ast_sockaddr_port(ast_websocket_remote_address(session));
recvd = pjsip_tpmgr_receive_packet(rdata->tp_info.transport->tpmgr, rdata);
pj_pool_reset(rdata->tp_info.pool);
return (read_data->payload_len == recvd) ? 0 : -1;
}
res_http_websocket: Close websocket correctly and use careful fwrite When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. Note that this version of the patch, unlike r417310 in Asterisk 11, exposes configuration options beyond just chan_sip's sip.conf. Configuration options to configure the write timeout have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ ........ Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-26 12:21:14 +00:00
static int get_write_timeout(void)
{
int write_timeout = -1;
struct ao2_container *transports;
transports = ast_sorcery_retrieve_by_fields(ast_sip_get_sorcery(), "transport", AST_RETRIEVE_FLAG_ALL, NULL);
if (transports) {
struct ao2_iterator it_transports = ao2_iterator_init(transports, 0);
struct ast_sip_transport *transport;
for (; (transport = ao2_iterator_next(&it_transports)); ao2_cleanup(transport)) {
if (transport->type != AST_TRANSPORT_WS && transport->type != AST_TRANSPORT_WSS) {
continue;
}
ast_debug(5, "Found %s transport with write timeout: %d\n",
transport->type == AST_TRANSPORT_WS ? "WS" : "WSS",
transport->write_timeout);
write_timeout = MAX(write_timeout, transport->write_timeout);
}
ao2_cleanup(transports);
}
if (write_timeout < 0) {
write_timeout = AST_DEFAULT_WEBSOCKET_WRITE_TIMEOUT;
}
ast_debug(1, "Write timeout for WS/WSS transports: %d\n", write_timeout);
return write_timeout;
}
/*!
\brief WebSocket connection handler.
*/
static void websocket_cb(struct ast_websocket *session, struct ast_variable *parameters, struct ast_variable *headers)
{
struct ast_taskprocessor *serializer = NULL;
struct transport_create_data create_data;
struct ws_transport *transport = NULL;
struct transport_read_data read_data;
if (ast_websocket_set_nonblock(session)) {
ast_websocket_unref(session);
return;
}
res_http_websocket: Close websocket correctly and use careful fwrite When a client takes a long time to process information received from Asterisk, a write operation using fwrite may fail to write all information. This causes the underlying file stream to be in an unknown state, such that the socket must be disconnected. Unfortunately, there are two problems with this in Asterisk's existing websocket code: 1. Periodically, during the read loop, Asterisk must write to the connected websocket to respond to pings. As such, Asterisk maintains a reference to the session during the loop. When ast_http_websocket_write fails, it may cause the session to decrement its ref count, but this in and of itself does not break the read loop. The read loop's write, on the other hand, does not break the loop if it fails. This causes the socket to get in a 'stuck' state, preventing the client from reconnecting to the server. 2. More importantly, however, is that the fwrite in ast_http_websocket_write fails with a large volume of data when the client takes awhile to process the information. When it does fail, it fails writing only a portion of the bytes. With some debugging, it was shown that this was failing in a similar fashion to ASTERISK-12767. Switching this over to ast_careful_fwrite with a long enough timeout solved the problem. Note that this version of the patch, unlike r417310 in Asterisk 11, exposes configuration options beyond just chan_sip's sip.conf. Configuration options to configure the write timeout have also been added to pjsip.conf and ari.conf. #ASTERISK-23917 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3624/ ........ Merged revisions 417310 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 417311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-26 12:21:14 +00:00
if (ast_websocket_set_timeout(session, get_write_timeout())) {
ast_websocket_unref(session);
return;
}
if (!(serializer = ast_sip_create_serializer())) {
ast_websocket_unref(session);
return;
}
create_data.ws_session = session;
if (ast_sip_push_task_synchronous(serializer, transport_create, &create_data)) {
ast_log(LOG_ERROR, "Could not create WebSocket transport.\n");
ast_websocket_unref(session);
return;
}
transport = create_data.transport;
read_data.transport = transport;
while (ast_wait_for_input(ast_websocket_fd(session), -1) > 0) {
enum ast_websocket_opcode opcode;
int fragmented;
if (ast_websocket_read(session, &read_data.payload, &read_data.payload_len, &opcode, &fragmented)) {
break;
}
if (opcode == AST_WEBSOCKET_OPCODE_TEXT || opcode == AST_WEBSOCKET_OPCODE_BINARY) {
ast_sip_push_task_synchronous(serializer, transport_read, &read_data);
} else if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
break;
}
}
ast_sip_push_task_synchronous(serializer, transport_shutdown, transport);
ast_taskprocessor_unreference(serializer);
ast_websocket_unref(session);
}
/*!
* \brief Store the transport a message came in on, so it can be used for outbound messages to that contact.
*/
static pj_bool_t websocket_on_rx_msg(pjsip_rx_data *rdata)
{
static const pj_str_t STR_WS = { "ws", 2 };
static const pj_str_t STR_WSS = { "wss", 3 };
pjsip_contact_hdr *contact;
long type = rdata->tp_info.transport->key.type;
if (type != (long)transport_type_ws && type != (long)transport_type_wss) {
return PJ_FALSE;
}
if ((contact = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL)) && !contact->star &&
(PJSIP_URI_SCHEME_IS_SIP(contact->uri) || PJSIP_URI_SCHEME_IS_SIPS(contact->uri))) {
pjsip_sip_uri *uri = pjsip_uri_get_uri(contact->uri);
pj_cstr(&uri->host, rdata->pkt_info.src_name);
uri->port = rdata->pkt_info.src_port;
ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan ........ Merged revisions 431717 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-12 20:34:37 +00:00
ast_debug(4, "Re-wrote Contact URI host/port to %.*s:%d\n",
(int)pj_strlen(&uri->host), pj_strbuf(&uri->host), uri->port);
pj_strdup(rdata->tp_info.pool, &uri->transport_param, (type == (long)transport_type_ws) ? &STR_WS : &STR_WSS);
}
rdata->msg_info.via->rport_param = 0;
return PJ_FALSE;
}
static pjsip_module websocket_module = {
.name = { "WebSocket Transport Module", 26 },
.id = -1,
.priority = PJSIP_MOD_PRIORITY_TRANSPORT_LAYER,
.on_rx_request = websocket_on_rx_msg,
.on_rx_response = websocket_on_rx_msg,
};
/*! \brief Function called when an INVITE goes out */
static void websocket_outgoing_invite_request(struct ast_sip_session *session, struct pjsip_tx_data *tdata)
{
if (session->inv_session->state == PJSIP_INV_STATE_NULL) {
pjsip_dlg_add_usage(session->inv_session->dlg, &websocket_module, NULL);
}
}
/*! \brief Supplement for adding Websocket functionality to dialog */
static struct ast_sip_session_supplement websocket_supplement = {
.method = "INVITE",
.priority = AST_SIP_SUPPLEMENT_PRIORITY_FIRST + 1,
.outgoing_request = websocket_outgoing_invite_request,
};
static int load_module(void)
{
CHECK_PJSIP_MODULE_LOADED();
pjsip_transport_register_type(PJSIP_TRANSPORT_RELIABLE, "WS", 5060, &transport_type_ws);
pjsip_transport_register_type(PJSIP_TRANSPORT_RELIABLE | PJSIP_TRANSPORT_SECURE, "WSS", 5060, &transport_type_wss);
if (ast_sip_register_service(&websocket_module) != PJ_SUCCESS) {
return AST_MODULE_LOAD_DECLINE;
}
if (ast_sip_session_register_supplement(&websocket_supplement)) {
ast_sip_unregister_service(&websocket_module);
return AST_MODULE_LOAD_DECLINE;
}
if (ast_websocket_add_protocol("sip", websocket_cb)) {
ast_sip_session_unregister_supplement(&websocket_supplement);
ast_sip_unregister_service(&websocket_module);
return AST_MODULE_LOAD_DECLINE;
}
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
ast_sip_unregister_service(&websocket_module);
ast_sip_session_unregister_supplement(&websocket_supplement);
ast_websocket_remove_protocol("sip", websocket_cb);
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP WebSocket Transport Support",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
);