Files
asterisk/res/res_hep_rtcp.c
Joshua Colp 3c36c29c81 res_hep_rtcp: Provide chan_sip Call-ID for RTCP messages.
This change adds the required logic to allow the SIP
Call-ID to be placed into the HEP RTCP traffic if the
chan_sip module is used. In cases where the option is
enabled but the channel is not either SIP or PJSIP then
the code will fallback to the channel name as done
previously.

Based on the change on Nir's branch at:
team/nirs/hep-chan-sip-support

ASTERISK-26427

Change-Id: I09ffa5f6e2fdfd99ee999650ba4e0a7aad6dc40d
2017-05-09 05:38:59 -05:00

188 lines
4.6 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) {
struct ast_channel *chan = NULL;
char buf[128];
if (ast_begins_with(channel_name, "PJSIP")) {
chan = ast_channel_get_by_name(channel_name);
if (chan && !ast_func_read(chan, "CHANNEL(pjsip,call-id)", buf, sizeof(buf))) {
uuid = ast_strdup(buf);
}
} else if (ast_begins_with(channel_name, "SIP")) {
chan = ast_channel_get_by_name(channel_name);
if (chan && !ast_func_read(chan, "SIP_HEADER(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,
);