Asterisk - The Open Source Telephony Project
21.4.1
|
RTP (Multicast and Unicast) Media Channel. More...
#include "asterisk.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/acl.h"
#include "asterisk/app.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/causes.h"
#include "asterisk/format_cache.h"
#include "asterisk/multicast_rtp.h"
#include "asterisk/dns_core.h"
Go to the source code of this file.
Enumerations | |
enum | { OPT_RTP_CODEC = (1 << 0), OPT_RTP_ENGINE = (1 << 1), OPT_RTP_GLUE = (1 << 2) } |
enum | { OPT_ARG_RTP_CODEC, OPT_ARG_RTP_ENGINE, OPT_ARG_ARRAY_SIZE } |
Functions | |
static void | __reg_module (void) |
static void | __unreg_module (void) |
struct ast_module * | AST_MODULE_SELF_SYM (void) |
static void | chan_rtp_get_codec (struct ast_channel *chan, struct ast_format_cap *result) |
Function called by RTP engine to get peer capabilities. | |
static enum ast_rtp_glue_result | chan_rtp_get_rtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance) |
Function called by RTP engine to get local audio RTP peer. | |
static enum ast_rtp_glue_result | chan_rtp_get_vrtp_peer (struct ast_channel *chan, struct ast_rtp_instance **instance) |
Function called by RTP engine to get local audio RTP peer. | |
static int | chan_rtp_set_rtp_peer (struct ast_channel *chan, struct ast_rtp_instance *rtp, struct ast_rtp_instance *vrtp, struct ast_rtp_instance *tpeer, const struct ast_format_cap *cap, int nat_active) |
Function called by RTP engine to change where the remote party should send media. More... | |
static struct ast_format * | derive_format_from_cap (struct ast_format_cap *cap) |
static int | load_module (void) |
Function called when our module is loaded. | |
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) |
Function called when we should prepare to call the multicast destination. | |
static int | rtp_call (struct ast_channel *ast, const char *dest, int timeout) |
Function called when we should actually call the destination. | |
static int | rtp_hangup (struct ast_channel *ast) |
Function called when we should hang the channel up. | |
static struct ast_frame * | rtp_read (struct ast_channel *ast) |
Function called when we should read a frame from the channel. | |
static int | rtp_write (struct ast_channel *ast, struct ast_frame *f) |
Function called when we should write a frame to the channel. | |
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) |
Function called when we should prepare to call the unicast destination. | |
static int | unload_module (void) |
Function called when our module is unloaded. | |
Variables | |
static struct ast_module_info | __mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "RTP Media Channel" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = "da6642af068ee5e6490c5b1d2cc1d238" , .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DRIVER, .requires = "res_rtp_multicast", } |
static const struct ast_module_info * | ast_module_info = &__mod_info |
static const struct ast_datastore_info | chan_rtp_datastore_info |
static struct ast_channel_tech | multicast_rtp_tech |
static struct ast_rtp_glue | unicast_rtp_glue |
Local glue for interacting with the RTP engine core. | |
static const struct ast_app_option | unicast_rtp_options [128] = { [ 'c' ] = { .flag = OPT_RTP_CODEC , .arg_index = OPT_ARG_RTP_CODEC + 1 }, [ 'e' ] = { .flag = OPT_RTP_ENGINE , .arg_index = OPT_ARG_RTP_ENGINE + 1 }, [ 'g' ] = { .flag = OPT_RTP_GLUE }, } |
static struct ast_channel_tech | unicast_rtp_tech |
RTP (Multicast and Unicast) Media Channel.
Definition in file chan_rtp.c.
|
static |
Function called by RTP engine to change where the remote party should send media.
chan_rtp is not able to actually update the peer, so this function has no effect.
Definition at line 432 of file chan_rtp.c.
|
static |
Definition at line 272 of file chan_rtp.c.