Files
asterisk/res/res_hep_rtcp.c
Matt Jordan e06a23681c res_hep: Provide an option to pick the UUID type
At one point in time, it seemed like a good idea to use the Asterisk
channel name as the HEP correlation UUID. In particular, it felt like
this would be a useful identifier to tie PJSIP messages and RTCP
messages together, along with whatever other data we may eventually send
to Homer. This also had the benefit of keeping the correlation UUID
channel technology agnostic.

In practice, it isn't as useful as hoped, for two reasons:
1) The first INVITE request received doesn't have a channel. As a
   result, there is always an 'odd message out', leading it to be
   potentially uncorrelated in Homer.
2) Other systems sending capture packets (Kamailio) use the SIP Call-ID.
   This causes RTCP information to be uncorrelated to the SIP message
   traffic seen by those capture nodes.

In order to support both (in case someone is trying to use res_hep_rtcp
with a non-PJSIP channel), this patch adds a new option, uuid_type, with
two valid values - 'call-id' and 'channel'. The uuid_type option is used
by a module to determine the preferred UUID type. When available, that
source of a correlation UUID is used; when not, the more readily available
source is used.

For res_hep_pjsip:
 - uuid_type = call-id: the module uses the SIP Call-ID header value
 - uuid_type = channel: the module uses the channel name if available,
                        falling back to SIP Call-ID if not
For res_hep_rtcp:
 - uuid_type = call-id: the module uses the SIP Call-ID header if the
                        channel type is PJSIP and we have a channel,
                        falling back to the Stasis event provided
                        channel name if not
 - uuid_type = channel: the module uses the channel name

ASTERISK-25352 #close

Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c
2016-05-14 09:42:20 -05:00

176 lines
4.2 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"
ASTERISK_REGISTER_FILE()
#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)
{
stasis_rtp_subscription = stasis_subscribe(ast_rtp_topic(),
rtp_topic_handler, NULL);
if (!stasis_rtp_subscription) {
return AST_MODULE_LOAD_FAILURE;
}
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,
);