asterisk/res/res_pjsip_transport_websocket.c
Richard Mudgett 237d341bbd res_pjsip.c: Split ast_sip_push_task_synchronous() to fit expectations.
ast_sip_push_task_synchronous() did not necessarily execute the passed in
task under the specified serializer.  If the current thread is any
registered pjsip thread then it would execute the task immediately instead
of under the specified serializer.  Reentrancy issues could result if the
task does not execute with the right serializer.

The original reason ast_sip_push_task_synchronous() checked to see if the
current thread was a registered pjsip thread was because of a deadlock
with masquerades and the channel technology's fixup callback
(ASTERISK_22936).  A subsequent masquerade deadlock fix (ASTERISK_24356)
involving call pickups avoided the original deadlock situation entirely.
The PJSIP channel technology's fixup callback no longer needed to call
ast_sip_push_task_synchronous().

However, there are a few places where this unexpected behavior is still
required to avoid deadlocks.  The pjsip monitor thread executes callbacks
that do calls to ast_sip_push_task_synchronous() that would deadlock if
the task were actually pushed to the specified serializer.  I ran into one
dealing with the pubsub subscriptions where an ao2 destructor called
ast_sip_push_task_synchronous().

* Split ast_sip_push_task_synchronous() into
ast_sip_push_task_wait_servant() and ast_sip_push_task_wait_serializer().
ast_sip_push_task_wait_servant() has the old behavior of
ast_sip_push_task_synchronous().  ast_sip_push_task_wait_serializer() has
the new behavior where the task is always executed by the specified
serializer or a picked serializer if one is not passed in.  Both functions
behave the same if the current thread is not a SIP servant.

* Redirected ast_sip_push_task_synchronous() to
ast_sip_push_task_wait_servant() to preserve API for released branches.

ASTERISK_26806

Change-Id: Id040fa42c0e5972f4c8deef380921461d213b9f3
2018-04-12 17:34:16 -05:00

518 lines
16 KiB
C

/*
* 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_wss;
static int transport_type_wss_ipv6;
/*!
* \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;
uint64_t len = tdata->buf.cur - tdata->buf.start;
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_tp_state_callback state_cb;
pjsip_endpoint *endpt = ast_sip_get_pjsip_endpoint();
struct pjsip_tpmgr *tpmgr = pjsip_endpt_get_tpmgr(endpt);
char *ws_addr_str;
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;
}
/* Give websocket transport a unique name for its lifetime */
snprintf(newtransport->transport.obj_name, PJ_MAX_OBJ_NAME, "ws%p",
&newtransport->transport);
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;
}
/*
* The type_name here is mostly used by log messages eihter in
* pjproject or Asterisk. Other places are reconstituting subscriptions
* after a restart (which could never work for a websocket connection anyway),
* received MESSAGE requests to set PJSIP_TRANSPORT, and most importantly
* by pjproject when generating the Via header.
*/
newtransport->transport.type_name = ast_websocket_is_secure(newtransport->ws_session)
? "WSS" : "WS";
ws_addr_str = ast_sockaddr_stringify(ast_websocket_remote_address(newtransport->ws_session));
ast_debug(4, "Creating websocket transport for %s:%s\n",
newtransport->transport.type_name, ws_addr_str);
newtransport->transport.info = (char *) pj_pool_alloc(newtransport->transport.pool,
strlen(newtransport->transport.type_name) + strlen(ws_addr_str) + sizeof(" to "));
sprintf(newtransport->transport.info, "%s to %s", newtransport->transport.type_name, ws_addr_str);
pj_sockaddr_parse(pj_AF_UNSPEC(), 0, pj_cstr(&buf, ws_addr_str), &newtransport->transport.key.rem_addr);
if (newtransport->transport.key.rem_addr.addr.sa_family == pj_AF_INET6()) {
newtransport->transport.key.type = transport_type_wss_ipv6;
} else {
newtransport->transport.key.type = transport_type_wss;
}
newtransport->transport.addr_len = pj_sockaddr_get_len(&newtransport->transport.key.rem_addr);
ws_addr_str = ast_sockaddr_stringify(ast_websocket_local_address(newtransport->ws_session));
pj_sockaddr_parse(pj_AF_UNSPEC(), 0, pj_cstr(&buf, ws_addr_str), &newtransport->transport.local_addr);
pj_strdup2(pool, &newtransport->transport.local_name.host, ast_sockaddr_stringify_host(ast_websocket_local_address(newtransport->ws_session)));
newtransport->transport.local_name.port = ast_sockaddr_port(ast_websocket_local_address(newtransport->ws_session));
newtransport->transport.flag = pjsip_transport_get_flag_from_type((pjsip_transport_type_e)newtransport->transport.key.type);
newtransport->transport.dir = PJSIP_TP_DIR_INCOMING;
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;
/* Notify application of transport state */
state_cb = pjsip_tpmgr_get_state_cb(newtransport->transport.tpmgr);
if (state_cb) {
pjsip_transport_state_info state_info;
memset(&state_info, 0, sizeof(state_info));
state_cb(&newtransport->transport, PJSIP_TP_STATE_CONNECTED, &state_info);
}
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_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;
}
static int get_write_timeout(void)
{
int write_timeout = -1;
struct ao2_container *transport_states;
transport_states = ast_sip_get_transport_states();
if (transport_states) {
struct ao2_iterator it_transport_states = ao2_iterator_init(transport_states, 0);
struct ast_sip_transport_state *transport_state;
for (; (transport_state = ao2_iterator_next(&it_transport_states)); ao2_cleanup(transport_state)) {
struct ast_sip_transport *transport;
if (transport_state->type != AST_TRANSPORT_WS && transport_state->type != AST_TRANSPORT_WSS) {
continue;
}
transport = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "transport", transport_state->id);
if (!transport) {
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_iterator_destroy(&it_transport_states);
ao2_cleanup(transport_states);
}
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;
}
static struct ast_taskprocessor *create_websocket_serializer(void)
{
char tps_name[AST_TASKPROCESSOR_MAX_NAME + 1];
/* Create name with seq number appended. */
ast_taskprocessor_build_name(tps_name, sizeof(tps_name), "pjsip/websocket");
return ast_sip_create_serializer(tps_name);
}
/*! \brief WebSocket connection handler. */
static void websocket_cb(struct ast_websocket *session, struct ast_variable *parameters, struct ast_variable *headers)
{
struct ast_taskprocessor *serializer;
struct transport_create_data create_data;
struct ws_transport *transport;
struct transport_read_data read_data;
if (ast_websocket_set_nonblock(session)) {
ast_websocket_unref(session);
return;
}
if (ast_websocket_set_timeout(session, get_write_timeout())) {
ast_websocket_unref(session);
return;
}
serializer = create_websocket_serializer();
if (!serializer) {
ast_websocket_unref(session);
return;
}
create_data.ws_session = session;
if (ast_sip_push_task_wait_serializer(serializer, transport_create, &create_data)) {
ast_log(LOG_ERROR, "Could not create WebSocket transport.\n");
ast_taskprocessor_unreference(serializer);
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_wait_serializer(serializer, transport_read, &read_data);
} else if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
break;
}
}
ast_sip_push_task_wait_serializer(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 };
pjsip_contact_hdr *contact;
long type = rdata->tp_info.transport->key.type;
if (type != (long) transport_type_wss && type != (long) transport_type_wss_ipv6) {
return PJ_FALSE;
}
contact = pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL);
if (contact
&& !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);
const pj_str_t *txp_str = &STR_WS;
ast_debug(4, "%s re-writing Contact URI from %.*s:%d%s%.*s to %s:%d;transport=%s\n",
pjsip_rx_data_get_info(rdata),
(int)pj_strlen(&uri->host), pj_strbuf(&uri->host), uri->port,
pj_strlen(&uri->transport_param) ? ";transport=" : "",
(int)pj_strlen(&uri->transport_param), pj_strbuf(&uri->transport_param),
rdata->pkt_info.src_name ?: "", rdata->pkt_info.src_port, pj_strbuf(txp_str));
pj_cstr(&uri->host, rdata->pkt_info.src_name);
uri->port = rdata->pkt_info.src_port;
pj_strdup(rdata->tp_info.pool, &uri->transport_param, txp_str);
}
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)
{
/*
* We only need one transport type name (ws) defined. Firefox
* and Chrome do not support anything other than secure websockets
* anymore.
*
* Also we really cannot have two transports with the same name
* and address family because it would be ambiguous. Outgoing
* requests may try to find the transport by name and pjproject
* only finds the first one registered.
*/
pjsip_transport_register_type(PJSIP_TRANSPORT_RELIABLE | PJSIP_TRANSPORT_SECURE, "ws", 5060, &transport_type_wss);
pjsip_transport_register_type(PJSIP_TRANSPORT_RELIABLE | PJSIP_TRANSPORT_SECURE | PJSIP_TRANSPORT_IPV6, "ws", 5060, &transport_type_wss_ipv6);
if (ast_sip_register_service(&websocket_module) != PJ_SUCCESS) {
return AST_MODULE_LOAD_DECLINE;
}
ast_sip_session_register_supplement(&websocket_supplement);
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,
.requires = "res_pjsip,res_http_websocket",
);