multicast RTP: Add dialing options

This adds a new parameter to the end of a multicast RTP dialing string.
This parameter defines the following options:

* i: Set the interface from which multicast RTP is sent
* l: Set whether multicast packets are looped back to the sender
* t: Set the TTL for multicast packets
* c: Set the codec to use for RTP

ASTERISK-26068 #close
Reported by Mark Michelson

Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219
This commit is contained in:
Mark Michelson 2016-05-26 15:14:50 -05:00 committed by Richard Mudgett
parent a6b16d7029
commit bb0f4a6310
4 changed files with 263 additions and 13 deletions

View File

@ -43,6 +43,7 @@ ASTERISK_REGISTER_FILE()
#include "asterisk/rtp_engine.h"
#include "asterisk/causes.h"
#include "asterisk/format_cache.h"
#include "asterisk/multicast_rtp.h"
/* Forward declarations */
static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause);
@ -132,7 +133,9 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
AST_APP_ARG(type);
AST_APP_ARG(destination);
AST_APP_ARG(control);
AST_APP_ARG(options);
);
struct ast_multicast_rtp_options *mcast_options = NULL;
if (ast_strlen_zero(data)) {
ast_log(LOG_ERROR, "A multicast type and destination must be given to the 'MulticastRTP' channel\n");
@ -163,7 +166,15 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
goto failure;
}
fmt = ast_format_cap_get_format(cap, 0);
mcast_options = ast_multicast_rtp_create_options(args.type, args.options);
if (!mcast_options) {
goto failure;
}
fmt = ast_multicast_rtp_options_get_format(mcast_options);
if (!fmt) {
fmt = ast_format_cap_get_format(cap, 0);
}
if (!fmt) {
ast_log(LOG_ERROR, "No format available for sending RTP to '%s'\n",
args.destination);
@ -175,7 +186,7 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
goto failure;
}
instance = ast_rtp_instance_new("multicast", NULL, &control_address, args.type);
instance = ast_rtp_instance_new("multicast", NULL, &control_address, mcast_options);
if (!instance) {
ast_log(LOG_ERROR,
"Could not create '%s' multicast RTP instance for sending media to '%s'\n",
@ -207,12 +218,14 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
ao2_ref(fmt, -1);
ao2_ref(caps, -1);
ast_multicast_rtp_free_options(mcast_options);
return chan;
failure:
ao2_cleanup(fmt);
ao2_cleanup(caps);
ast_multicast_rtp_free_options(mcast_options);
*cause = AST_CAUSE_FAILURE;
return NULL;
}

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@ -0,0 +1,58 @@
/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 2016, Digium, Inc.
*
* Mark Michelson <mmichelson@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.
*/
#ifndef MULTICAST_RTP_H_
#define MULTICAST_RTP_H_
struct ast_multicast_rtp_options;
/*!
* \brief Create multicast RTP options.
*
* These are passed to the multicast RTP engine on its creation.
*
* \param type The type of multicast RTP, either "basic" or "linksys"
* \param options Miscellaneous options
* \retval NULL Failure
* \retval non-NULL success
*/
struct ast_multicast_rtp_options *ast_multicast_rtp_create_options(const char *type,
const char *options);
/*!
* \brief Free multicast RTP options
*
* This function is NULL-tolerant
*
* \param mcast_options Options to free
*/
void ast_multicast_rtp_free_options(struct ast_multicast_rtp_options *mcast_options);
/*!
* \brief Get format specified in multicast options
*
* Multicast options allow for a format to be selected.
* This function accesses the selected format and creates
* an ast_format structure for it.
*
* \param mcast_options The options where a codec was specified
* \retval NULL No format specified in the options
* \revval non-NULL The format to use for communication
*/
struct ast_format *ast_multicast_rtp_options_get_format(struct ast_multicast_rtp_options *mcast_options);
#endif /* MULTICAST_RTP_H_ */

View File

@ -54,6 +54,8 @@ ASTERISK_REGISTER_FILE()
#include "asterisk/module.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/format_cache.h"
#include "asterisk/multicast_rtp.h"
#include "asterisk/app.h"
/*! Command value used for Linksys paging to indicate we are starting */
#define LINKSYS_MCAST_STARTCMD 6
@ -63,8 +65,10 @@ ASTERISK_REGISTER_FILE()
/*! \brief Type of paging to do */
enum multicast_type {
/*! Type has not been set yet */
MULTICAST_TYPE_UNSPECIFIED = 0,
/*! Simple multicast enabled client/receiver paging like Snom and Barix uses */
MULTICAST_TYPE_BASIC = 0,
MULTICAST_TYPE_BASIC,
/*! More advanced Linksys type paging which requires a start and stop packet */
MULTICAST_TYPE_LINKSYS,
};
@ -95,6 +99,91 @@ struct multicast_rtp {
struct timeval txcore;
};
enum {
OPT_CODEC = (1 << 0),
OPT_LOOP = (1 << 1),
OPT_TTL = (1 << 2),
OPT_IF = (1 << 3),
};
enum {
OPT_ARG_CODEC = 0,
OPT_ARG_LOOP,
OPT_ARG_TTL,
OPT_ARG_IF,
OPT_ARG_ARRAY_SIZE,
};
AST_APP_OPTIONS(multicast_rtp_options, BEGIN_OPTIONS
/*! Set the codec to be used for multicast RTP */
AST_APP_OPTION_ARG('c', OPT_CODEC, OPT_ARG_CODEC),
/*! Set whether multicast RTP is looped back to the sender */
AST_APP_OPTION_ARG('l', OPT_LOOP, OPT_ARG_LOOP),
/*! Set the hop count for multicast RTP */
AST_APP_OPTION_ARG('t', OPT_TTL, OPT_ARG_TTL),
/*! Set the interface from which multicast RTP is sent */
AST_APP_OPTION_ARG('i', OPT_IF, OPT_ARG_IF),
END_OPTIONS );
struct ast_multicast_rtp_options {
char *type;
char *options;
struct ast_format *fmt;
struct ast_flags opts;
char *opt_args[OPT_ARG_ARRAY_SIZE];
/*! The type and options are stored in this buffer */
char buf[0];
};
struct ast_multicast_rtp_options *ast_multicast_rtp_create_options(const char *type,
const char *options)
{
struct ast_multicast_rtp_options *mcast_options;
char *pos;
mcast_options = ast_calloc(1, sizeof(*mcast_options)
+ strlen(type)
+ strlen(options) + 2);
if (!mcast_options) {
return NULL;
}
pos = mcast_options->buf;
/* Safe */
strcpy(pos, type);
mcast_options->type = pos;
pos += strlen(type) + 1;
/* Safe */
strcpy(pos, options);
mcast_options->options = pos;
if (ast_app_parse_options(multicast_rtp_options, &mcast_options->opts,
mcast_options->opt_args, mcast_options->options)) {
ast_log(LOG_WARNING, "Error parsing multicast RTP options\n");
ast_multicast_rtp_free_options(mcast_options);
return NULL;
}
return mcast_options;
}
void ast_multicast_rtp_free_options(struct ast_multicast_rtp_options *mcast_options)
{
ast_free(mcast_options);
}
struct ast_format *ast_multicast_rtp_options_get_format(struct ast_multicast_rtp_options *mcast_options)
{
if (ast_test_flag(&mcast_options->opts, OPT_CODEC)
&& !ast_strlen_zero(mcast_options->opt_args[OPT_ARG_CODEC])) {
return ast_format_cache_get(mcast_options->opt_args[OPT_ARG_CODEC]);
}
return NULL;
}
/* Forward Declarations */
static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data);
static int multicast_rtp_activate(struct ast_rtp_instance *instance);
@ -112,21 +201,93 @@ static struct ast_rtp_engine multicast_rtp_engine = {
.read = multicast_rtp_read,
};
/*! \brief Function called to create a new multicast instance */
static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
static int set_type(struct multicast_rtp *multicast, const char *type)
{
struct multicast_rtp *multicast;
const char *type = data;
if (!(multicast = ast_calloc(1, sizeof(*multicast)))) {
return -1;
}
if (!strcasecmp(type, "basic")) {
multicast->type = MULTICAST_TYPE_BASIC;
} else if (!strcasecmp(type, "linksys")) {
multicast->type = MULTICAST_TYPE_LINKSYS;
} else {
ast_log(LOG_WARNING, "Unrecognized multicast type '%s' specified.\n", type);
return -1;
}
return 0;
}
static void set_ttl(int sock, const char *ttl_str)
{
int ttl;
if (ast_strlen_zero(ttl_str)) {
return;
}
ast_debug(3, "Setting multicast TTL to %s\n", ttl_str);
if (sscanf(ttl_str, "%30d", &ttl) < 1) {
ast_log(LOG_WARNING, "Inavlid multicast ttl option '%s'\n", ttl_str);
return;
}
if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_TTL, &ttl, sizeof(ttl)) < 0) {
ast_log(LOG_WARNING, "Could not set multicast ttl to '%s': %s\n",
ttl_str, strerror(errno));
}
}
static void set_loop(int sock, const char *loop_str)
{
unsigned char loop;
if (ast_strlen_zero(loop_str)) {
return;
}
ast_debug(3, "Setting multicast loop to %s\n", loop_str);
if (sscanf(loop_str, "%30hhu", &loop) < 1) {
ast_log(LOG_WARNING, "Invalid multicast loop option '%s'\n", loop_str);
return;
}
if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_LOOP, &loop, sizeof(loop)) < 0) {
ast_log(LOG_WARNING, "Could not set multicast loop to '%s': %s\n",
loop_str, strerror(errno));
}
}
static void set_if(int sock, const char *if_str)
{
struct in_addr iface;
if (ast_strlen_zero(if_str)) {
return;
}
ast_debug(3, "Setting multicast if to %s\n", if_str);
if (!inet_aton(if_str, &iface)) {
ast_log(LOG_WARNING, "Cannot parse if option '%s'\n", if_str);
}
if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_IF, &iface, sizeof(iface)) < 0) {
ast_log(LOG_WARNING, "Could not set multicast if to '%s': %s\n",
if_str, strerror(errno));
}
}
/*! \brief Function called to create a new multicast instance */
static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
{
struct multicast_rtp *multicast;
struct ast_multicast_rtp_options *mcast_options = data;
if (!(multicast = ast_calloc(1, sizeof(*multicast)))) {
return -1;
}
if (set_type(multicast, mcast_options->type)) {
ast_free(multicast);
return -1;
}
@ -136,6 +297,18 @@ static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched
return -1;
}
if (ast_test_flag(&mcast_options->opts, OPT_LOOP)) {
set_loop(multicast->socket, mcast_options->opt_args[OPT_ARG_LOOP]);
}
if (ast_test_flag(&mcast_options->opts, OPT_TTL)) {
set_ttl(multicast->socket, mcast_options->opt_args[OPT_ARG_TTL]);
}
if (ast_test_flag(&mcast_options->opts, OPT_IF)) {
set_if(multicast->socket, mcast_options->opt_args[OPT_ARG_IF]);
}
multicast->ssrc = ast_random();
ast_rtp_instance_set_data(instance, multicast);
@ -316,7 +489,7 @@ static int unload_module(void)
return 0;
}
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Engine",
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Multicast RTP Engine",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,

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@ -0,0 +1,6 @@
{
global:
LINKER_SYMBOL_PREFIXast_multicast_rtp*;
local:
*;
};