/*
* Asterisk -- An open source telephony toolkit.
*
- * Copyright (C) 2009, Digium, Inc.
+ * Copyright (C) 2009 - 2014, Digium, Inc.
*
* Joshua Colp <jcolp@digium.com>
* Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
* \author Joshua Colp <jcolp@digium.com>
* \author Andreas 'MacBrody' Broadmann <andreas.brodmann@gmail.com>
*
- * \brief Multicast RTP Paging Channel
+ * \brief RTP (Multicast and Unicast) Media Channel
*
* \ingroup channel_drivers
*/
#include "asterisk.h"
-ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+ASTERISK_REGISTER_FILE()
-#include <fcntl.h>
-#include <sys/signal.h>
-
-#include "asterisk/lock.h"
#include "asterisk/channel.h"
-#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
-#include "asterisk/sched.h"
-#include "asterisk/io.h"
#include "asterisk/acl.h"
-#include "asterisk/callerid.h"
-#include "asterisk/file.h"
-#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/causes.h"
-
-static const char tdesc[] = "Multicast RTP Paging Channel Driver";
+#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);
-static int multicast_rtp_call(struct ast_channel *ast, const char *dest, int timeout);
-static int multicast_rtp_hangup(struct ast_channel *ast);
-static struct ast_frame *multicast_rtp_read(struct ast_channel *ast);
-static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f);
+static struct ast_channel *unicast_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);
+static int rtp_call(struct ast_channel *ast, const char *dest, int timeout);
+static int rtp_hangup(struct ast_channel *ast);
+static struct ast_frame *rtp_read(struct ast_channel *ast);
+static int rtp_write(struct ast_channel *ast, struct ast_frame *f);
-/* Channel driver declaration */
+/* Multicast channel driver declaration */
static struct ast_channel_tech multicast_rtp_tech = {
.type = "MulticastRTP",
- .description = tdesc,
+ .description = "Multicast RTP Paging Channel Driver",
.requester = multicast_rtp_request,
- .call = multicast_rtp_call,
- .hangup = multicast_rtp_hangup,
- .read = multicast_rtp_read,
- .write = multicast_rtp_write,
+ .call = rtp_call,
+ .hangup = rtp_hangup,
+ .read = rtp_read,
+ .write = rtp_write,
+};
+
+/* Unicast channel driver declaration */
+static struct ast_channel_tech unicast_rtp_tech = {
+ .type = "UnicastRTP",
+ .description = "Unicast RTP Media Channel Driver",
+ .requester = unicast_rtp_request,
+ .call = rtp_call,
+ .hangup = rtp_hangup,
+ .read = rtp_read,
+ .write = rtp_write,
};
/*! \brief Function called when we should read a frame from the channel */
-static struct ast_frame *multicast_rtp_read(struct ast_channel *ast)
+static struct ast_frame *rtp_read(struct ast_channel *ast)
{
- return &ast_null_frame;
+ struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
+ int fdno = ast_channel_fdno(ast);
+
+ switch (fdno) {
+ case 0:
+ return ast_rtp_instance_read(instance, 0);
+ default:
+ return &ast_null_frame;
+ }
}
/*! \brief Function called when we should write a frame to the channel */
-static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f)
+static int rtp_write(struct ast_channel *ast, struct ast_frame *f)
{
struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
}
/*! \brief Function called when we should actually call the destination */
-static int multicast_rtp_call(struct ast_channel *ast, const char *dest, int timeout)
+static int rtp_call(struct ast_channel *ast, const char *dest, int timeout)
{
struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
}
/*! \brief Function called when we should hang the channel up */
-static int multicast_rtp_hangup(struct ast_channel *ast)
+static int rtp_hangup(struct ast_channel *ast)
{
struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
return 0;
}
-/*! \brief Function called when we should prepare to call the destination */
+/*! \brief Function called when we should prepare to call the multicast destination */
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)
{
- char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control;
+ char *parse;
struct ast_rtp_instance *instance;
struct ast_sockaddr control_address;
struct ast_sockaddr destination_address;
struct ast_channel *chan;
struct ast_format_cap *caps = NULL;
struct ast_format *fmt = NULL;
+ AST_DECLARE_APP_ARGS(args,
+ 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");
+ goto failure;
+ }
+ parse = ast_strdupa(data);
+ AST_NONSTANDARD_APP_ARGS(args, parse, '/');
- fmt = ast_format_cap_get_format(cap, 0);
-
- ast_sockaddr_setnull(&control_address);
+ if (ast_strlen_zero(args.type)) {
+ ast_log(LOG_ERROR, "Type is required for the 'MulticastRTP' channel\n");
+ goto failure;
+ }
- /* If no type was given we can't do anything */
- if (ast_strlen_zero(multicast_type)) {
+ if (ast_strlen_zero(args.destination)) {
+ ast_log(LOG_ERROR, "Destination is required for the 'MulticastRTP' channel\n");
+ goto failure;
+ }
+ if (!ast_sockaddr_parse(&destination_address, args.destination, PARSE_PORT_REQUIRE)) {
+ ast_log(LOG_ERROR, "Destination address '%s' could not be parsed\n",
+ args.destination);
goto failure;
}
- if (!(destination = strchr(tmp, '/'))) {
+ ast_sockaddr_setnull(&control_address);
+ if (!ast_strlen_zero(args.control)
+ && !ast_sockaddr_parse(&control_address, args.control, PARSE_PORT_REQUIRE)) {
+ ast_log(LOG_ERROR, "Control address '%s' could not be parsed\n", args.control);
goto failure;
}
- *destination++ = '\0';
- if ((control = strchr(destination, '/'))) {
- *control++ = '\0';
- if (!ast_sockaddr_parse(&control_address, control,
- PARSE_PORT_REQUIRE)) {
- goto failure;
- }
+ mcast_options = ast_multicast_rtp_create_options(args.type, args.options);
+ if (!mcast_options) {
+ goto failure;
}
- if (!ast_sockaddr_parse(&destination_address, destination,
- PARSE_PORT_REQUIRE)) {
+ 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 codec available for sending RTP to '%s'\n",
+ args.destination);
goto failure;
}
goto failure;
}
- if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, multicast_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",
+ args.type, args.destination);
goto failure;
}
- if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids, requestor, 0, "MulticastRTP/%p", instance))) {
+ chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
+ requestor, 0, "MulticastRTP/%p", instance);
+ if (!chan) {
ast_rtp_instance_destroy(instance);
goto failure;
}
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;
}
-/*! \brief Function called when our module is loaded */
-static int load_module(void)
+enum {
+ OPT_RTP_CODEC = (1 << 0),
+ OPT_RTP_ENGINE = (1 << 1),
+};
+
+enum {
+ OPT_ARG_RTP_CODEC,
+ OPT_ARG_RTP_ENGINE,
+ /* note: this entry _MUST_ be the last one in the enum */
+ OPT_ARG_ARRAY_SIZE
+};
+
+AST_APP_OPTIONS(unicast_rtp_options, BEGIN_OPTIONS
+ /*! Set the codec to be used for unicast RTP */
+ AST_APP_OPTION_ARG('c', OPT_RTP_CODEC, OPT_ARG_RTP_CODEC),
+ /*! Set the RTP engine to use for unicast RTP */
+ AST_APP_OPTION_ARG('e', OPT_RTP_ENGINE, OPT_ARG_RTP_ENGINE),
+END_OPTIONS );
+
+/*! \brief Function called when we should prepare to call the unicast destination */
+static struct ast_channel *unicast_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)
{
- if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
- return AST_MODULE_LOAD_DECLINE;
+ char *parse;
+ struct ast_rtp_instance *instance;
+ struct ast_sockaddr address;
+ struct ast_sockaddr local_address;
+ struct ast_channel *chan;
+ struct ast_format_cap *caps = NULL;
+ struct ast_format *fmt = NULL;
+ const char *engine_name;
+ AST_DECLARE_APP_ARGS(args,
+ AST_APP_ARG(destination);
+ AST_APP_ARG(options);
+ );
+ struct ast_flags opts = { 0, };
+ char *opt_args[OPT_ARG_ARRAY_SIZE];
+
+ if (ast_strlen_zero(data)) {
+ ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
+ goto failure;
}
- ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
- if (ast_channel_register(&multicast_rtp_tech)) {
- ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
- ao2_ref(multicast_rtp_tech.capabilities, -1);
- multicast_rtp_tech.capabilities = NULL;
- return AST_MODULE_LOAD_DECLINE;
+ parse = ast_strdupa(data);
+ AST_NONSTANDARD_APP_ARGS(args, parse, '/');
+
+ if (ast_strlen_zero(args.destination)) {
+ ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
+ goto failure;
+ }
+ if (!ast_sockaddr_parse(&address, args.destination, PARSE_PORT_REQUIRE)) {
+ ast_log(LOG_ERROR, "Destination '%s' could not be parsed\n", args.destination);
+ goto failure;
}
- return AST_MODULE_LOAD_SUCCESS;
+ if (!ast_strlen_zero(args.options)
+ && ast_app_parse_options(unicast_rtp_options, &opts, opt_args,
+ ast_strdupa(args.options))) {
+ ast_log(LOG_ERROR, "'UnicastRTP' channel options '%s' parse error\n",
+ args.options);
+ goto failure;
+ }
+
+ if (ast_test_flag(&opts, OPT_RTP_CODEC)
+ && !ast_strlen_zero(opt_args[OPT_ARG_RTP_CODEC])) {
+ fmt = ast_format_cache_get(opt_args[OPT_ARG_RTP_CODEC]);
+ if (!fmt) {
+ ast_log(LOG_ERROR, "Codec '%s' not found for sending RTP to '%s'\n",
+ opt_args[OPT_ARG_RTP_CODEC], args.destination);
+ goto failure;
+ }
+ } else {
+ fmt = ast_format_cap_get_format(cap, 0);
+ if (!fmt) {
+ ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
+ args.destination);
+ goto failure;
+ }
+ }
+
+ caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
+ if (!caps) {
+ goto failure;
+ }
+
+ engine_name = S_COR(ast_test_flag(&opts, OPT_RTP_ENGINE),
+ opt_args[OPT_ARG_RTP_ENGINE], NULL);
+
+ ast_ouraddrfor(&address, &local_address);
+ instance = ast_rtp_instance_new(engine_name, NULL, &local_address, NULL);
+ if (!instance) {
+ ast_log(LOG_ERROR,
+ "Could not create %s RTP instance for sending media to '%s'\n",
+ S_OR(engine_name, "default"), args.destination);
+ goto failure;
+ }
+
+ chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
+ requestor, 0, "UnicastRTP/%s-%p", args.destination, instance);
+ if (!chan) {
+ ast_rtp_instance_destroy(instance);
+ goto failure;
+ }
+ ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan));
+ ast_rtp_instance_set_remote_address(instance, &address);
+ ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(instance, 0));
+
+ ast_channel_tech_set(chan, &unicast_rtp_tech);
+
+ ast_format_cap_append(caps, fmt, 0);
+ ast_channel_nativeformats_set(chan, caps);
+ ast_channel_set_writeformat(chan, fmt);
+ ast_channel_set_rawwriteformat(chan, fmt);
+ ast_channel_set_readformat(chan, fmt);
+ ast_channel_set_rawreadformat(chan, fmt);
+
+ ast_channel_tech_pvt_set(chan, instance);
+
+ pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS",
+ ast_sockaddr_stringify_addr(&local_address));
+ ast_rtp_instance_get_local_address(instance, &local_address);
+ pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT",
+ ast_sockaddr_stringify_port(&local_address));
+
+ ast_channel_unlock(chan);
+
+ ao2_ref(fmt, -1);
+ ao2_ref(caps, -1);
+
+ return chan;
+
+failure:
+ ao2_cleanup(fmt);
+ ao2_cleanup(caps);
+ *cause = AST_CAUSE_FAILURE;
+ return NULL;
}
/*! \brief Function called when our module is unloaded */
static int unload_module(void)
{
ast_channel_unregister(&multicast_rtp_tech);
- ao2_ref(multicast_rtp_tech.capabilities, -1);
+ ao2_cleanup(multicast_rtp_tech.capabilities);
multicast_rtp_tech.capabilities = NULL;
+ ast_channel_unregister(&unicast_rtp_tech);
+ ao2_cleanup(unicast_rtp_tech.capabilities);
+ unicast_rtp_tech.capabilities = NULL;
+
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel",
+/*! \brief Function called when our module is loaded */
+static int load_module(void)
+{
+ if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
+ return AST_MODULE_LOAD_DECLINE;
+ }
+ ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
+ if (ast_channel_register(&multicast_rtp_tech)) {
+ ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
+ unload_module();
+ return AST_MODULE_LOAD_DECLINE;
+ }
+
+ if (!(unicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
+ unload_module();
+ return AST_MODULE_LOAD_DECLINE;
+ }
+ ast_format_cap_append_by_type(unicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
+ if (ast_channel_register(&unicast_rtp_tech)) {
+ ast_log(LOG_ERROR, "Unable to register channel class 'UnicastRTP'\n");
+ unload_module();
+ return AST_MODULE_LOAD_DECLINE;
+ }
+
+ return AST_MODULE_LOAD_SUCCESS;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "RTP Media Channel",
.support_level = AST_MODULE_SUPPORT_CORE,
.load = load_module,
.unload = unload_module,
#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
/*! \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,
};
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);
.read = multicast_rtp_read,
};
+static int set_type(struct multicast_rtp *multicast, const char *type)
+{
+ 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;
- const char *type = data;
+ struct ast_multicast_rtp_options *mcast_options = 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 {
+ if (set_type(multicast, mcast_options->type)) {
ast_free(multicast);
return -1;
}
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);
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,
projects / thirdparty / asterisk.git / commitdiff
