asterisk/res/res_hep_rtcp.c
George Joseph 747beb1ed1 modules: change module LOAD_FAILUREs to LOAD_DECLINES
In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE.  This prevents asterisk from exiting
if a module can't be loaded.  If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.

A new API was added to logger: ast_is_logger_initialized().  This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout.  If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.

Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
2017-04-12 15:57:21 -06:00

178 lines
4.4 KiB
C

/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2014, Digium, Inc.
*
* Matt Jordan <mjordan@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.
*/
/*!
* \file
* \brief RTCP logging with Homer
*
* \author Matt Jordan <mjordan@digium.com>
*
*/
/*** MODULEINFO
<depend>res_hep</depend>
<support_level>extended</support_level>
***/
#include "asterisk.h"
#include "asterisk/res_hep.h"
#include "asterisk/module.h"
#include "asterisk/netsock2.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/stasis.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/json.h"
#include "asterisk/config.h"
static struct stasis_subscription *stasis_rtp_subscription;
static char *assign_uuid(struct ast_json *json_channel)
{
const char *channel_name = ast_json_string_get(ast_json_object_get(json_channel, "name"));
enum hep_uuid_type uuid_type = hepv3_get_uuid_type();
char *uuid = NULL;
if (!channel_name) {
return NULL;
}
if (uuid_type == HEP_UUID_TYPE_CALL_ID && ast_begins_with(channel_name, "PJSIP")) {
struct ast_channel *chan = ast_channel_get_by_name(channel_name);
char buf[128];
if (chan && !ast_func_read(chan, "CHANNEL(pjsip,call-id)", buf, sizeof(buf))) {
uuid = ast_strdup(buf);
}
ast_channel_cleanup(chan);
}
/* If we couldn't get the call-id or didn't want it, just use the channel name */
if (!uuid) {
uuid = ast_strdup(channel_name);
}
return uuid;
}
static void rtcp_message_handler(struct stasis_message *message)
{
RAII_VAR(struct ast_json *, json_payload, NULL, ast_json_unref);
RAII_VAR(char *, payload, NULL, ast_json_free);
struct ast_json *json_blob;
struct ast_json *json_channel;
struct ast_json *json_rtcp;
struct hepv3_capture_info *capture_info;
struct ast_json *from;
struct ast_json *to;
struct timeval current_time = ast_tvnow();
json_payload = stasis_message_to_json(message, NULL);
if (!json_payload) {
return;
}
json_blob = ast_json_object_get(json_payload, "blob");
if (!json_blob) {
return;
}
json_channel = ast_json_object_get(json_payload, "channel");
if (!json_channel) {
return;
}
json_rtcp = ast_json_object_get(json_payload, "rtcp_report");
if (!json_rtcp) {
return;
}
from = ast_json_object_get(json_blob, "from");
to = ast_json_object_get(json_blob, "to");
if (!from || !to) {
return;
}
payload = ast_json_dump_string(json_rtcp);
if (ast_strlen_zero(payload)) {
return;
}
capture_info = hepv3_create_capture_info(payload, strlen(payload));
if (!capture_info) {
return;
}
ast_sockaddr_parse(&capture_info->src_addr, ast_json_string_get(from), PARSE_PORT_REQUIRE);
ast_sockaddr_parse(&capture_info->dst_addr, ast_json_string_get(to), PARSE_PORT_REQUIRE);
capture_info->uuid = assign_uuid(json_channel);
if (!capture_info->uuid) {
ao2_ref(capture_info, -1);
return;
}
capture_info->capture_time = current_time;
capture_info->capture_type = HEPV3_CAPTURE_TYPE_RTCP;
capture_info->zipped = 0;
hepv3_send_packet(capture_info);
}
static void rtp_topic_handler(void *data, struct stasis_subscription *sub, struct stasis_message *message)
{
struct stasis_message_type *message_type = stasis_message_type(message);
if ((message_type == ast_rtp_rtcp_sent_type()) ||
(message_type == ast_rtp_rtcp_received_type())) {
rtcp_message_handler(message);
}
}
static int load_module(void)
{
if (!ast_module_check("res_hep.so") || !hepv3_is_loaded()) {
ast_log(AST_LOG_WARNING, "res_hep is not loaded or running; declining module load\n");
return AST_MODULE_LOAD_DECLINE;
}
stasis_rtp_subscription = stasis_subscribe(ast_rtp_topic(),
rtp_topic_handler, NULL);
if (!stasis_rtp_subscription) {
return AST_MODULE_LOAD_DECLINE;
}
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
if (stasis_rtp_subscription) {
stasis_rtp_subscription = stasis_unsubscribe_and_join(stasis_rtp_subscription);
}
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "RTCP HEPv3 Logger",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_DEFAULT,
);