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			191 lines
		
	
	
		
			4.7 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
			
		
		
	
	
			191 lines
		
	
	
		
			4.7 KiB
		
	
	
	
		
			C
		
	
	
	
	
	
| /*
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|  * Asterisk -- An open source telephony toolkit.
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|  *
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|  * Copyright (C) 1999 - 2014, Digium, Inc.
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|  *
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|  * Matt Jordan <mjordan@digium.com>
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|  *
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|  * See http://www.asterisk.org for more information about
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|  * the Asterisk project. Please do not directly contact
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|  * any of the maintainers of this project for assistance;
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|  * the project provides a web site, mailing lists and IRC
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|  * channels for your use.
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|  *
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|  * This program is free software, distributed under the terms of
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|  * the GNU General Public License Version 2. See the LICENSE file
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|  * at the top of the source tree.
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|  */
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| 
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| /*!
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|  * \file
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|  * \brief RTCP logging with Homer
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|  *
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|  * \author Matt Jordan <mjordan@digium.com>
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|  *
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|  */
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| 
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| /*** MODULEINFO
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| 	<depend>res_hep</depend>
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| 	<support_level>extended</support_level>
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|  ***/
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| 
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| #include "asterisk.h"
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| 
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| ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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| 
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| #include "asterisk/res_hep.h"
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| #include "asterisk/module.h"
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| #include "asterisk/netsock2.h"
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| #include "asterisk/channel.h"
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| #include "asterisk/pbx.h"
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| #include "asterisk/stasis.h"
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| #include "asterisk/rtp_engine.h"
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| #include "asterisk/json.h"
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| #include "asterisk/config.h"
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| 
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| static struct stasis_subscription *stasis_rtp_subscription;
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| 
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| static char *assign_uuid(struct ast_json *json_channel)
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| {
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| 	const char *channel_name = ast_json_string_get(ast_json_object_get(json_channel, "name"));
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| 	enum hep_uuid_type uuid_type = hepv3_get_uuid_type();
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| 	char *uuid = NULL;
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| 
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| 	if (!channel_name) {
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| 		return NULL;
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| 	}
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| 
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| 	if (uuid_type == HEP_UUID_TYPE_CALL_ID) {
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| 		struct ast_channel *chan = NULL;
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| 		char buf[128];
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| 
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| 		if (ast_begins_with(channel_name, "PJSIP")) {
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| 			chan = ast_channel_get_by_name(channel_name);
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| 
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| 			if (chan && !ast_func_read(chan, "CHANNEL(pjsip,call-id)", buf, sizeof(buf))) {
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| 				uuid = ast_strdup(buf);
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| 			}
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| 		} else if (ast_begins_with(channel_name, "SIP")) {
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| 			chan = ast_channel_get_by_name(channel_name);
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| 
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| 			if (chan && !ast_func_read(chan, "SIP_HEADER(call-id)", buf, sizeof(buf))) {
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| 				uuid = ast_strdup(buf);
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| 			}
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| 		}
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| 
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| 		ast_channel_cleanup(chan);
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| 	}
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| 
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| 	/* If we couldn't get the call-id or didn't want it, just use the channel name */
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| 	if (!uuid) {
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| 		uuid = ast_strdup(channel_name);
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| 	}
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| 
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| 	return uuid;
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| }
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| 
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| static void rtcp_message_handler(struct stasis_message *message)
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| {
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| 
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| 	RAII_VAR(struct ast_json *, json_payload, NULL, ast_json_unref);
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| 	RAII_VAR(char *,  payload, NULL, ast_json_free);
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| 	struct ast_json *json_blob;
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| 	struct ast_json *json_channel;
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| 	struct ast_json *json_rtcp;
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| 	struct hepv3_capture_info *capture_info;
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| 	struct ast_json *from;
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| 	struct ast_json *to;
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| 	struct timeval current_time = ast_tvnow();
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| 
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| 	json_payload = stasis_message_to_json(message, NULL);
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| 	if (!json_payload) {
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| 		return;
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| 	}
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| 
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| 	json_blob = ast_json_object_get(json_payload, "blob");
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| 	if (!json_blob) {
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| 		return;
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| 	}
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| 
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| 	json_channel = ast_json_object_get(json_payload, "channel");
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| 	if (!json_channel) {
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| 		return;
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| 	}
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| 
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| 	json_rtcp = ast_json_object_get(json_payload, "rtcp_report");
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| 	if (!json_rtcp) {
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| 		return;
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| 	}
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| 
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| 	from = ast_json_object_get(json_blob, "from");
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| 	to = ast_json_object_get(json_blob, "to");
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| 	if (!from || !to) {
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| 		return;
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| 	}
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| 
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| 	payload = ast_json_dump_string(json_rtcp);
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| 	if (ast_strlen_zero(payload)) {
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| 		return;
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| 	}
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| 
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| 	capture_info = hepv3_create_capture_info(payload, strlen(payload));
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| 	if (!capture_info) {
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| 		return;
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| 	}
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| 	ast_sockaddr_parse(&capture_info->src_addr, ast_json_string_get(from), PARSE_PORT_REQUIRE);
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| 	ast_sockaddr_parse(&capture_info->dst_addr, ast_json_string_get(to), PARSE_PORT_REQUIRE);
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| 
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| 	capture_info->uuid = assign_uuid(json_channel);
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| 	if (!capture_info->uuid) {
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| 		ao2_ref(capture_info, -1);
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| 		return;
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| 	}
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| 	capture_info->capture_time = current_time;
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| 	capture_info->capture_type = HEPV3_CAPTURE_TYPE_RTCP;
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| 	capture_info->zipped = 0;
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| 
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| 	hepv3_send_packet(capture_info);
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| }
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| 
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| static void rtp_topic_handler(void *data, struct stasis_subscription *sub, struct stasis_message *message)
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| {
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| 	struct stasis_message_type *message_type = stasis_message_type(message);
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| 
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| 	if ((message_type == ast_rtp_rtcp_sent_type()) ||
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| 		(message_type == ast_rtp_rtcp_received_type())) {
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| 		rtcp_message_handler(message);
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| 	}
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| }
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| 
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| static int load_module(void)
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| {
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| 	if (!ast_module_check("res_hep.so") || !hepv3_is_loaded()) {
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| 		ast_log(AST_LOG_WARNING, "res_hep is not loaded or running; declining module load\n");
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| 		return AST_MODULE_LOAD_DECLINE;
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| 	}
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| 
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| 	stasis_rtp_subscription = stasis_subscribe(ast_rtp_topic(),
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| 		rtp_topic_handler, NULL);
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| 	if (!stasis_rtp_subscription) {
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| 		return AST_MODULE_LOAD_DECLINE;
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| 	}
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| 
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| 	return AST_MODULE_LOAD_SUCCESS;
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| }
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| 
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| static int unload_module(void)
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| {
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| 	if (stasis_rtp_subscription) {
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| 		stasis_rtp_subscription = stasis_unsubscribe_and_join(stasis_rtp_subscription);
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| 	}
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| 
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| 	return 0;
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| }
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| 
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| AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "RTCP HEPv3 Logger",
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| 	.support_level = AST_MODULE_SUPPORT_EXTENDED,
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| 	.load = load_module,
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| 	.unload = unload_module,
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| 	.load_pri = AST_MODPRI_DEFAULT,
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| 	);
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