ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966
497 lines
14 KiB
C
497 lines
14 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2009, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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* Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*!
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* \file
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*
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* \brief Multicast RTP Engine
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*
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* \author Joshua Colp <jcolp@digium.com>
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* \author Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.com>
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*
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* \ingroup rtp_engines
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include <sys/time.h>
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#include <signal.h>
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#include <fcntl.h>
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#include <math.h>
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#include "asterisk/pbx.h"
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#include "asterisk/frame.h"
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#include "asterisk/channel.h"
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#include "asterisk/acl.h"
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#include "asterisk/config.h"
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#include "asterisk/lock.h"
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#include "asterisk/utils.h"
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#include "asterisk/cli.h"
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#include "asterisk/manager.h"
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#include "asterisk/unaligned.h"
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#include "asterisk/module.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/format_cache.h"
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#include "asterisk/multicast_rtp.h"
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#include "asterisk/app.h"
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/*! Command value used for Linksys paging to indicate we are starting */
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#define LINKSYS_MCAST_STARTCMD 6
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/*! Command value used for Linksys paging to indicate we are stopping */
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#define LINKSYS_MCAST_STOPCMD 7
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/*! \brief Type of paging to do */
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enum multicast_type {
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/*! Type has not been set yet */
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MULTICAST_TYPE_UNSPECIFIED = 0,
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/*! Simple multicast enabled client/receiver paging like Snom and Barix uses */
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MULTICAST_TYPE_BASIC,
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/*! More advanced Linksys type paging which requires a start and stop packet */
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MULTICAST_TYPE_LINKSYS,
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};
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/*! \brief Structure for a Linksys control packet */
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struct multicast_control_packet {
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/*! Unique identifier for the control packet */
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uint32_t unique_id;
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/*! Actual command in the control packet */
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uint32_t command;
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/*! IP address for the RTP */
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uint32_t ip;
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/*! Port for the RTP */
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uint32_t port;
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};
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/*! \brief Structure for a multicast paging instance */
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struct multicast_rtp {
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/*! TYpe of multicast paging this instance is doing */
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enum multicast_type type;
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/*! Socket used for sending the audio on */
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int socket;
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/*! Synchronization source value, used when creating/sending the RTP packet */
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unsigned int ssrc;
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/*! Sequence number, used when creating/sending the RTP packet */
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uint16_t seqno;
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unsigned int lastts;
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struct timeval txcore;
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};
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enum {
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OPT_CODEC = (1 << 0),
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OPT_LOOP = (1 << 1),
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OPT_TTL = (1 << 2),
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OPT_IF = (1 << 3),
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};
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enum {
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OPT_ARG_CODEC = 0,
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OPT_ARG_LOOP,
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OPT_ARG_TTL,
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OPT_ARG_IF,
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OPT_ARG_ARRAY_SIZE,
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};
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AST_APP_OPTIONS(multicast_rtp_options, BEGIN_OPTIONS
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/*! Set the codec to be used for multicast RTP */
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AST_APP_OPTION_ARG('c', OPT_CODEC, OPT_ARG_CODEC),
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/*! Set whether multicast RTP is looped back to the sender */
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AST_APP_OPTION_ARG('l', OPT_LOOP, OPT_ARG_LOOP),
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/*! Set the hop count for multicast RTP */
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AST_APP_OPTION_ARG('t', OPT_TTL, OPT_ARG_TTL),
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/*! Set the interface from which multicast RTP is sent */
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AST_APP_OPTION_ARG('i', OPT_IF, OPT_ARG_IF),
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END_OPTIONS );
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struct ast_multicast_rtp_options {
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char *type;
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char *options;
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struct ast_format *fmt;
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struct ast_flags opts;
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char *opt_args[OPT_ARG_ARRAY_SIZE];
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/*! The type and options are stored in this buffer */
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char buf[0];
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};
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struct ast_multicast_rtp_options *ast_multicast_rtp_create_options(const char *type,
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const char *options)
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{
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struct ast_multicast_rtp_options *mcast_options;
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char *pos;
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mcast_options = ast_calloc(1, sizeof(*mcast_options)
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+ strlen(type)
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+ strlen(S_OR(options, "")) + 2);
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if (!mcast_options) {
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return NULL;
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}
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pos = mcast_options->buf;
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/* Safe */
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strcpy(pos, type);
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mcast_options->type = pos;
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pos += strlen(type) + 1;
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if (!ast_strlen_zero(options)) {
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strcpy(pos, options); /* Safe */
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}
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mcast_options->options = pos;
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if (ast_app_parse_options(multicast_rtp_options, &mcast_options->opts,
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mcast_options->opt_args, mcast_options->options)) {
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ast_log(LOG_WARNING, "Error parsing multicast RTP options\n");
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ast_multicast_rtp_free_options(mcast_options);
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return NULL;
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}
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return mcast_options;
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}
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void ast_multicast_rtp_free_options(struct ast_multicast_rtp_options *mcast_options)
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{
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ast_free(mcast_options);
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}
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struct ast_format *ast_multicast_rtp_options_get_format(struct ast_multicast_rtp_options *mcast_options)
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{
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if (ast_test_flag(&mcast_options->opts, OPT_CODEC)
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&& !ast_strlen_zero(mcast_options->opt_args[OPT_ARG_CODEC])) {
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return ast_format_cache_get(mcast_options->opt_args[OPT_ARG_CODEC]);
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}
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return NULL;
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}
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/* Forward Declarations */
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static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data);
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static int multicast_rtp_activate(struct ast_rtp_instance *instance);
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static int multicast_rtp_destroy(struct ast_rtp_instance *instance);
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static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
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static struct ast_frame *multicast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
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/* RTP Engine Declaration */
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static struct ast_rtp_engine multicast_rtp_engine = {
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.name = "multicast",
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.new = multicast_rtp_new,
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.activate = multicast_rtp_activate,
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.destroy = multicast_rtp_destroy,
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.write = multicast_rtp_write,
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.read = multicast_rtp_read,
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};
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static int set_type(struct multicast_rtp *multicast, const char *type)
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{
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if (!strcasecmp(type, "basic")) {
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multicast->type = MULTICAST_TYPE_BASIC;
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} else if (!strcasecmp(type, "linksys")) {
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multicast->type = MULTICAST_TYPE_LINKSYS;
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} else {
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ast_log(LOG_WARNING, "Unrecognized multicast type '%s' specified.\n", type);
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return -1;
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}
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return 0;
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}
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static void set_ttl(int sock, const char *ttl_str)
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{
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int ttl;
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if (ast_strlen_zero(ttl_str)) {
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return;
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}
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ast_debug(3, "Setting multicast TTL to %s\n", ttl_str);
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if (sscanf(ttl_str, "%30d", &ttl) < 1) {
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ast_log(LOG_WARNING, "Invalid multicast ttl option '%s'\n", ttl_str);
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return;
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}
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if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_TTL, &ttl, sizeof(ttl)) < 0) {
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ast_log(LOG_WARNING, "Could not set multicast ttl to '%s': %s\n",
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ttl_str, strerror(errno));
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}
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}
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static void set_loop(int sock, const char *loop_str)
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{
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unsigned char loop;
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if (ast_strlen_zero(loop_str)) {
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return;
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}
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ast_debug(3, "Setting multicast loop to %s\n", loop_str);
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if (sscanf(loop_str, "%30hhu", &loop) < 1) {
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ast_log(LOG_WARNING, "Invalid multicast loop option '%s'\n", loop_str);
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return;
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}
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if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_LOOP, &loop, sizeof(loop)) < 0) {
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ast_log(LOG_WARNING, "Could not set multicast loop to '%s': %s\n",
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loop_str, strerror(errno));
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}
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}
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static void set_if(int sock, const char *if_str)
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{
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struct in_addr iface;
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if (ast_strlen_zero(if_str)) {
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return;
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}
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ast_debug(3, "Setting multicast if to %s\n", if_str);
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if (!inet_aton(if_str, &iface)) {
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ast_log(LOG_WARNING, "Cannot parse if option '%s'\n", if_str);
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}
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if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_IF, &iface, sizeof(iface)) < 0) {
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ast_log(LOG_WARNING, "Could not set multicast if to '%s': %s\n",
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if_str, strerror(errno));
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}
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}
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/*! \brief Function called to create a new multicast instance */
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static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
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{
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struct multicast_rtp *multicast;
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struct ast_multicast_rtp_options *mcast_options = data;
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if (!(multicast = ast_calloc(1, sizeof(*multicast)))) {
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return -1;
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}
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if (set_type(multicast, mcast_options->type)) {
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ast_free(multicast);
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return -1;
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}
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if ((multicast->socket = socket(AF_INET, SOCK_DGRAM, 0)) < 0) {
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ast_free(multicast);
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return -1;
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}
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if (ast_test_flag(&mcast_options->opts, OPT_LOOP)) {
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set_loop(multicast->socket, mcast_options->opt_args[OPT_ARG_LOOP]);
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}
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if (ast_test_flag(&mcast_options->opts, OPT_TTL)) {
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set_ttl(multicast->socket, mcast_options->opt_args[OPT_ARG_TTL]);
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}
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if (ast_test_flag(&mcast_options->opts, OPT_IF)) {
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set_if(multicast->socket, mcast_options->opt_args[OPT_ARG_IF]);
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}
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multicast->ssrc = ast_random();
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ast_rtp_instance_set_data(instance, multicast);
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return 0;
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}
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static int rtp_get_rate(struct ast_format *format)
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{
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return ast_format_cmp(format, ast_format_g722) == AST_FORMAT_CMP_EQUAL ?
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8000 : ast_format_get_sample_rate(format);
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}
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static unsigned int calc_txstamp(struct multicast_rtp *rtp, struct timeval *delivery)
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{
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struct timeval t;
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long ms;
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if (ast_tvzero(rtp->txcore)) {
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rtp->txcore = ast_tvnow();
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rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
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}
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t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
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if ((ms = ast_tvdiff_ms(t, rtp->txcore)) < 0) {
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ms = 0;
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}
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rtp->txcore = t;
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return (unsigned int) ms;
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}
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/*! \brief Helper function which populates a control packet with useful information and sends it */
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static int multicast_send_control_packet(struct ast_rtp_instance *instance, struct multicast_rtp *multicast, int command)
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{
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struct multicast_control_packet control_packet = { .unique_id = htonl((u_long)time(NULL)),
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.command = htonl(command),
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};
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struct ast_sockaddr control_address, remote_address;
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ast_rtp_instance_get_local_address(instance, &control_address);
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ast_rtp_instance_get_remote_address(instance, &remote_address);
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/* Ensure the user of us have given us both the control address and destination address */
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if (ast_sockaddr_isnull(&control_address) ||
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ast_sockaddr_isnull(&remote_address)) {
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return -1;
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}
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/* The protocol only supports IPv4. */
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if (ast_sockaddr_is_ipv6(&remote_address)) {
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ast_log(LOG_WARNING, "Cannot send control packet for IPv6 "
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"remote address.\n");
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return -1;
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}
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control_packet.ip = htonl(ast_sockaddr_ipv4(&remote_address));
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control_packet.port = htonl(ast_sockaddr_port(&remote_address));
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/* Based on a recommendation by Brian West who did the FreeSWITCH implementation we send control packets twice */
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ast_sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, &control_address);
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ast_sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, &control_address);
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return 0;
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}
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/*! \brief Function called to indicate that audio is now going to flow */
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static int multicast_rtp_activate(struct ast_rtp_instance *instance)
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{
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struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
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if (multicast->type != MULTICAST_TYPE_LINKSYS) {
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return 0;
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}
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return multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STARTCMD);
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}
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/*! \brief Function called to destroy a multicast instance */
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static int multicast_rtp_destroy(struct ast_rtp_instance *instance)
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{
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struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
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if (multicast->type == MULTICAST_TYPE_LINKSYS) {
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multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STOPCMD);
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}
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close(multicast->socket);
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ast_free(multicast);
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return 0;
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}
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/*! \brief Function called to broadcast some audio on a multicast instance */
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static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
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{
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struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
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struct ast_frame *f = frame;
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struct ast_sockaddr remote_address;
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int hdrlen = 12, res = 0, codec;
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unsigned char *rtpheader;
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unsigned int ms = calc_txstamp(multicast, &frame->delivery);
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int rate = rtp_get_rate(frame->subclass.format) / 1000;
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/* We only accept audio, nothing else */
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if (frame->frametype != AST_FRAME_VOICE) {
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return 0;
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}
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/* Grab the actual payload number for when we create the RTP packet */
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codec = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance),
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1, frame->subclass.format, 0);
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if (codec < 0) {
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return -1;
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}
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/* If we do not have space to construct an RTP header duplicate the frame so we get some */
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if (frame->offset < hdrlen) {
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f = ast_frdup(frame);
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}
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/* Calucate last TS */
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multicast->lastts = multicast->lastts + ms * rate;
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/* Construct an RTP header for our packet */
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rtpheader = (unsigned char *)(f->data.ptr - hdrlen);
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put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno)));
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if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO)) {
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put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8));
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} else {
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put_unaligned_uint32(rtpheader + 4, htonl(multicast->lastts));
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}
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put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc));
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/* Increment sequence number and wrap to 0 if it overflows 16 bits. */
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multicast->seqno = 0xFFFF & (multicast->seqno + 1);
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/* Finally send it out to the eager phones listening for us */
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ast_rtp_instance_get_remote_address(instance, &remote_address);
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if (ast_sendto(multicast->socket, (void *) rtpheader, f->datalen + hdrlen, 0, &remote_address) < 0) {
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ast_log(LOG_ERROR, "Multicast RTP Transmission error to %s: %s\n",
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ast_sockaddr_stringify(&remote_address),
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strerror(errno));
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res = -1;
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}
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/* If we were forced to duplicate the frame free the new one */
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if (frame != f) {
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ast_frfree(f);
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}
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return res;
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}
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/*! \brief Function called to read from a multicast instance */
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static struct ast_frame *multicast_rtp_read(struct ast_rtp_instance *instance, int rtcp)
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{
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return &ast_null_frame;
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}
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static int load_module(void)
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{
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if (ast_rtp_engine_register(&multicast_rtp_engine)) {
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return AST_MODULE_LOAD_DECLINE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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static int unload_module(void)
|
|
{
|
|
ast_rtp_engine_unregister(&multicast_rtp_engine);
|
|
|
|
return 0;
|
|
}
|
|
|
|
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,
|
|
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
|
|
);
|