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-rw-r--r--res/res_pjsip.c28
-rw-r--r--res/res_pjsip/location.c17
-rw-r--r--res/res_pjsip/pjsip_configuration.c10
-rw-r--r--res/res_pjsip/pjsip_options.c10
-rw-r--r--res/res_pjsip_header_funcs.c9
-rw-r--r--res/res_pjsip_registrar.c32
-rw-r--r--res/res_rtp_multicast.c189
-rw-r--r--res/res_rtp_multicast.exports.in6
8 files changed, 284 insertions, 17 deletions
diff --git a/res/res_pjsip.c b/res/res_pjsip.c
index bebe941b5..8fc3c530e 100644
--- a/res/res_pjsip.c
+++ b/res/res_pjsip.c
@@ -1175,6 +1175,28 @@
Asterisk Server name on which SIP endpoint registered.
</para></description>
</configOption>
+ <configOption name="via_addr">
+ <synopsis>IP-address of the last Via header from registration.</synopsis>
+ <description><para>
+ The last Via header should contain the address of UA which sent the request.
+ The IP-address of the last Via header is automatically stored based on data present
+ in incoming SIP REGISTER requests and is not intended to be configured manually.
+ </para></description>
+ </configOption>
+ <configOption name="via_port">
+ <synopsis>IP-port of the last Via header from registration.</synopsis>
+ <description><para>
+ The IP-port of the last Via header is automatically stored based on data present
+ in incoming SIP REGISTER requests and is not intended to be configured manually.
+ </para></description>
+ </configOption>
+ <configOption name="call_id">
+ <synopsis>Call-ID header from registration.</synopsis>
+ <description><para>
+ The Call-ID header is automatically stored based on data present
+ in incoming SIP REGISTER requests and is not intended to be configured manually.
+ </para></description>
+ </configOption>
</configObject>
<configObject name="aor">
<synopsis>The configuration for a location of an endpoint</synopsis>
@@ -1967,6 +1989,12 @@
<parameter name="RegExpire">
<para>Absolute time that this contact is no longer valid after</para>
</parameter>
+ <parameter name="ViaAddress">
+ <para>IP address:port of the last Via header in REGISTER request</para>
+ </parameter>
+ <parameter name="CallID">
+ <para>Content of the Call-ID header in REGISTER request</para>
+ </parameter>
</syntax>
</managerEventInstance>
</managerEvent>
diff --git a/res/res_pjsip/location.c b/res/res_pjsip/location.c
index fd6db6edc..43e6ea40f 100644
--- a/res/res_pjsip/location.c
+++ b/res/res_pjsip/location.c
@@ -121,6 +121,8 @@ static void *contact_alloc(const char *name)
}
ast_string_field_init_extended(contact, reg_server);
+ ast_string_field_init_extended(contact, via_addr);
+ ast_string_field_init_extended(contact, call_id);
/* Dynamic contacts are delimited with ";@" and static ones with "@@" */
if ((aor_separator = strstr(id, ";@")) || (aor_separator = strstr(id, "@@"))) {
@@ -303,6 +305,7 @@ struct ast_sip_contact *ast_sip_location_retrieve_contact(const char *contact_na
int ast_sip_location_add_contact_nolock(struct ast_sip_aor *aor, const char *uri,
struct timeval expiration_time, const char *path_info, const char *user_agent,
+ const char *via_addr, int via_port, const char *call_id,
struct ast_sip_endpoint *endpoint)
{
char name[MAX_OBJECT_FIELD * 2 + 3];
@@ -337,6 +340,15 @@ int ast_sip_location_add_contact_nolock(struct ast_sip_aor *aor, const char *uri
ast_string_field_set(contact, reg_server, ast_config_AST_SYSTEM_NAME);
}
+ if (!ast_strlen_zero(via_addr)) {
+ ast_string_field_set(contact, via_addr, via_addr);
+ }
+ contact->via_port = via_port;
+
+ if (!ast_strlen_zero(call_id)) {
+ ast_string_field_set(contact, call_id, call_id);
+ }
+
contact->endpoint = ao2_bump(endpoint);
return ast_sorcery_create(ast_sip_get_sorcery(), contact);
@@ -344,6 +356,7 @@ int ast_sip_location_add_contact_nolock(struct ast_sip_aor *aor, const char *uri
int ast_sip_location_add_contact(struct ast_sip_aor *aor, const char *uri,
struct timeval expiration_time, const char *path_info, const char *user_agent,
+ const char *via_addr, int via_port, const char *call_id,
struct ast_sip_endpoint *endpoint)
{
int res;
@@ -356,6 +369,7 @@ int ast_sip_location_add_contact(struct ast_sip_aor *aor, const char *uri,
ao2_wrlock(lock);
res = ast_sip_location_add_contact_nolock(aor, uri, expiration_time, path_info, user_agent,
+ via_addr, via_port, call_id,
endpoint);
ao2_unlock(lock);
ast_named_lock_put(lock);
@@ -1120,6 +1134,9 @@ int ast_sip_initialize_sorcery_location(void)
ast_sorcery_object_field_register(sorcery, "contact", "outbound_proxy", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_contact, outbound_proxy));
ast_sorcery_object_field_register(sorcery, "contact", "user_agent", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_contact, user_agent));
ast_sorcery_object_field_register(sorcery, "contact", "reg_server", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_contact, reg_server));
+ ast_sorcery_object_field_register(sorcery, "contact", "via_addr", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_contact, via_addr));
+ ast_sorcery_object_field_register(sorcery, "contact", "via_port", "0", OPT_UINT_T, 0, FLDSET(struct ast_sip_contact, via_port));
+ ast_sorcery_object_field_register(sorcery, "contact", "call_id", "", OPT_STRINGFIELD_T, 0, STRFLDSET(struct ast_sip_contact, call_id));
ast_sorcery_object_field_register(sorcery, "aor", "type", "", OPT_NOOP_T, 0, 0);
ast_sorcery_object_field_register(sorcery, "aor", "minimum_expiration", "60", OPT_UINT_T, 0, FLDSET(struct ast_sip_aor, minimum_expiration));
diff --git a/res/res_pjsip/pjsip_configuration.c b/res/res_pjsip/pjsip_configuration.c
index 8b6fe61d8..3c4949573 100644
--- a/res/res_pjsip/pjsip_configuration.c
+++ b/res/res_pjsip/pjsip_configuration.c
@@ -131,7 +131,7 @@ static int persistent_endpoint_update_state(void *obj, void *arg, int flags)
}
}
- ast_verb(1, "Endpoint %s is now Reachable\n", ast_endpoint_get_resource(endpoint));
+ ast_verb(2, "Endpoint %s is now Reachable\n", ast_endpoint_get_resource(endpoint));
} else {
ast_endpoint_set_state(endpoint, AST_ENDPOINT_OFFLINE);
blob = ast_json_pack("{s: s}", "peer_status", "Unreachable");
@@ -144,7 +144,7 @@ static int persistent_endpoint_update_state(void *obj, void *arg, int flags)
}
}
- ast_verb(1, "Endpoint %s is now Unreachable\n", ast_endpoint_get_resource(endpoint));
+ ast_verb(2, "Endpoint %s is now Unreachable\n", ast_endpoint_get_resource(endpoint));
}
ast_free(regcontext);
@@ -173,7 +173,7 @@ static void persistent_endpoint_contact_created_observer(const void *object)
contact_status->status = CREATED;
- ast_verb(1, "Contact %s/%s has been created\n",contact->aor, contact->uri);
+ ast_verb(2, "Contact %s/%s has been created\n",contact->aor, contact->uri);
ao2_callback(persistent_endpoints, OBJ_NODATA, persistent_endpoint_update_state, contact_status);
ao2_cleanup(contact_status);
@@ -192,7 +192,7 @@ static void persistent_endpoint_contact_deleted_observer(const void *object)
return;
}
- ast_verb(1, "Contact %s/%s has been deleted\n", contact->aor, contact->uri);
+ ast_verb(2, "Contact %s/%s has been deleted\n", contact->aor, contact->uri);
ast_statsd_log_string_va("PJSIP.contacts.states.%s", AST_STATSD_GAUGE,
"-1", 1.0, ast_sip_get_contact_status_label(contact_status->status));
ast_statsd_log_string_va("PJSIP.contacts.states.%s", AST_STATSD_GAUGE,
@@ -220,7 +220,7 @@ static void persistent_endpoint_contact_status_observer(const void *object)
}
if (contact_status->status != contact_status->last_status) {
- ast_verb(1, "Contact %s/%s is now %s. RTT: %.3f msec\n", contact_status->aor, contact_status->uri,
+ ast_verb(3, "Contact %s/%s is now %s. RTT: %.3f msec\n", contact_status->aor, contact_status->uri,
ast_sip_get_contact_status_label(contact_status->status),
contact_status->rtt / 1000.0);
diff --git a/res/res_pjsip/pjsip_options.c b/res/res_pjsip/pjsip_options.c
index 62640fe4e..1114336bd 100644
--- a/res/res_pjsip/pjsip_options.c
+++ b/res/res_pjsip/pjsip_options.c
@@ -1156,6 +1156,16 @@ static int format_contact_status(void *obj, void *arg, int flags)
ast_str_append(&buf, 0, "URI: %s\r\n", contact->uri);
ast_str_append(&buf, 0, "UserAgent: %s\r\n", contact->user_agent);
ast_str_append(&buf, 0, "RegExpire: %ld\r\n", contact->expiration_time.tv_sec);
+ if (!ast_strlen_zero(contact->via_addr)) {
+ ast_str_append(&buf, 0, "ViaAddress: %s", contact->via_addr);
+ if (contact->via_port) {
+ ast_str_append(&buf, 0, ":%d", contact->via_port);
+ }
+ ast_str_append(&buf, 0, "\r\n");
+ }
+ if (!ast_strlen_zero(contact->call_id)) {
+ ast_str_append(&buf, 0, "CallID: %s\r\n", contact->call_id);
+ }
ast_str_append(&buf, 0, "Status: %s\r\n", ast_sip_get_contact_status_label(status->status));
if (status->status == UNKNOWN) {
ast_str_append(&buf, 0, "RoundtripUsec: N/A\r\n");
diff --git a/res/res_pjsip_header_funcs.c b/res/res_pjsip_header_funcs.c
index 7d164b12a..648f1c860 100644
--- a/res/res_pjsip_header_funcs.c
+++ b/res/res_pjsip_header_funcs.c
@@ -39,7 +39,8 @@
/*** DOCUMENTATION
<function name="PJSIP_HEADER" language="en_US">
<synopsis>
- Gets, adds, updates or removes the specified SIP header from a PJSIP session.
+ Gets headers from an inbound PJSIP channel. Adds, updates or removes the
+ specified SIP header from an outbound PJSIP channel.
</synopsis>
<syntax>
<parameter name="action" required="true">
@@ -75,6 +76,10 @@
</syntax>
<description>
+ <para>PJSIP_HEADER allows you to read specific SIP headers from the inbound
+ PJSIP channel as well as write(add, update, remove) headers on the outbound
+ channel. One exception is that you can read headers that you have already
+ added on the outbound channel.</para>
<para>Examples:</para>
<para>;</para>
<para>; Set 'somevar' to the value of the 'From' header.</para>
@@ -120,7 +125,7 @@
<note><para>If you call PJSIP_HEADER in a normal dialplan context you'll be
operating on the <emphasis>caller's (incoming)</emphasis> channel which
- may not be what you want. To operate on the <emphasis>callee's (outgoing)</emphasis>
+ may not be what you want. To operate on the <emphasis>callee's (outgoing)</emphasis>
channel call PJSIP_HEADER in a pre-dial handler. </para>
<para>Example:</para>
<para>;</para>
diff --git a/res/res_pjsip_registrar.c b/res/res_pjsip_registrar.c
index cbc33ab80..0e14ab786 100644
--- a/res/res_pjsip_registrar.c
+++ b/res/res_pjsip_registrar.c
@@ -447,6 +447,13 @@ static int rx_task_core(struct rx_task_data *task_data, struct ao2_container *co
char *user_agent = NULL;
pjsip_user_agent_hdr *user_agent_hdr;
pjsip_expires_hdr *expires_hdr;
+ pjsip_via_hdr *via_hdr;
+ pjsip_via_hdr *via_hdr_last;
+ char *via_addr = NULL;
+ int via_port = 0;
+ pjsip_cid_hdr *call_id_hdr;
+ char *call_id = NULL;
+ size_t alloc_size;
/* So we don't count static contacts against max_contacts we prune them out from the container */
ao2_callback(contacts, OBJ_NODATA | OBJ_UNLINK | OBJ_MULTIPLE, registrar_prune_static, NULL);
@@ -484,11 +491,32 @@ static int rx_task_core(struct rx_task_data *task_data, struct ao2_container *co
user_agent_hdr = pjsip_msg_find_hdr_by_name(task_data->rdata->msg_info.msg, &USER_AGENT, NULL);
if (user_agent_hdr) {
- size_t alloc_size = pj_strlen(&user_agent_hdr->hvalue) + 1;
+ alloc_size = pj_strlen(&user_agent_hdr->hvalue) + 1;
user_agent = ast_alloca(alloc_size);
ast_copy_pj_str(user_agent, &user_agent_hdr->hvalue, alloc_size);
}
+ /* Find the first Via header */
+ via_hdr = via_hdr_last = (pjsip_via_hdr*) pjsip_msg_find_hdr(task_data->rdata->msg_info.msg, PJSIP_H_VIA, NULL);
+ if (via_hdr) {
+ /* Find the last Via header */
+ while ( (via_hdr = (pjsip_via_hdr*) pjsip_msg_find_hdr(task_data->rdata->msg_info.msg,
+ PJSIP_H_VIA, via_hdr->next)) != NULL) {
+ via_hdr_last = via_hdr;
+ }
+ alloc_size = pj_strlen(&via_hdr_last->sent_by.host) + 1;
+ via_addr = ast_alloca(alloc_size);
+ ast_copy_pj_str(via_addr, &via_hdr_last->sent_by.host, alloc_size);
+ via_port=via_hdr_last->sent_by.port;
+ }
+
+ call_id_hdr = (pjsip_cid_hdr*) pjsip_msg_find_hdr(task_data->rdata->msg_info.msg, PJSIP_H_CALL_ID, NULL);
+ if (call_id_hdr) {
+ alloc_size = pj_strlen(&call_id_hdr->id) + 1;
+ call_id = ast_alloca(alloc_size);
+ ast_copy_pj_str(call_id, &call_id_hdr->id, alloc_size);
+ }
+
/* Iterate each provided Contact header and add, update, or delete */
while ((contact_hdr = pjsip_msg_find_hdr(task_data->rdata->msg_info.msg, PJSIP_H_CONTACT, contact_hdr ? contact_hdr->next : NULL))) {
int expiration;
@@ -520,7 +548,7 @@ static int rx_task_core(struct rx_task_data *task_data, struct ao2_container *co
if (ast_sip_location_add_contact_nolock(task_data->aor, contact_uri, ast_tvadd(ast_tvnow(),
ast_samp2tv(expiration, 1)), path_str ? ast_str_buffer(path_str) : NULL,
- user_agent, task_data->endpoint)) {
+ user_agent, via_addr, via_port, call_id, task_data->endpoint)) {
ast_log(LOG_ERROR, "Unable to bind contact '%s' to AOR '%s'\n",
contact_uri, aor_name);
continue;
diff --git a/res/res_rtp_multicast.c b/res/res_rtp_multicast.c
index 192f3d137..5c419d3e7 100644
--- a/res/res_rtp_multicast.c
+++ b/res/res_rtp_multicast.c
@@ -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,
};
+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;
}
@@ -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,
diff --git a/res/res_rtp_multicast.exports.in b/res/res_rtp_multicast.exports.in
new file mode 100644
index 000000000..995a1802e
--- /dev/null
+++ b/res/res_rtp_multicast.exports.in
@@ -0,0 +1,6 @@
+{
+ global:
+ LINKER_SYMBOL_PREFIXast_multicast_rtp*;
+ local:
+ *;
+};
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-02 09:45:49 +0000