When a frame destined for a MulticastRTP channel does not have timing information (such as when an 'originate' is done), we generate the RTP timestamps ourselves without regard to the number of samples we are about to send. Instead, use the same method as res_rtp_asterisk and 'predict' a timestamp given the number of samples. If the difference between the timestamp that we generate and the one we predict is within a specific threshold, use the predicted timestamp so that we end up with timestamps that are consistent with the number of samples we are actually sending. Change-Id: I2bf0db3541b1573043330421cbb114ff0f22ec1f
568 lines
16 KiB
C
568 lines
16 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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#include "asterisk/smoother.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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struct ast_smoother *smoother;
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};
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#define MAX_TIMESTAMP_SKEW 640
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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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if (multicast->smoother) {
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ast_smoother_free(multicast->smoother);
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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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static int rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *frame, int codec)
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{
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struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance);
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unsigned int ms = calc_txstamp(multicast, &frame->delivery);
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unsigned char *rtpheader;
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struct ast_sockaddr remote_address = { {0,} };
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int rate = rtp_get_rate(frame->subclass.format) / 1000;
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int hdrlen = 12, mark = 0;
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if (ast_format_cmp(frame->subclass.format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) {
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frame->samples /= 2;
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}
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if (ast_test_flag(frame, AST_FRFLAG_HAS_TIMING_INFO)) {
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multicast->lastts = frame->ts * rate;
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} else {
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/* Try to predict what our timestamp should be */
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int pred = multicast->lastts + frame->samples;
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/* Calculate last TS */
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multicast->lastts = multicast->lastts + ms * rate;
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if (ast_tvzero(frame->delivery)) {
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int delta = abs((int) multicast->lastts - pred);
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if (delta < MAX_TIMESTAMP_SKEW) {
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multicast->lastts = pred;
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} else {
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ast_debug(3, "Difference is %d, ms is %u\n", delta, ms);
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mark = 1;
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}
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}
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}
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/* Construct an RTP header for our packet */
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rtpheader = (unsigned char *)(frame->data.ptr - hdrlen);
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put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno) | (mark << 23)));
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put_unaligned_uint32(rtpheader + 4, htonl(multicast->lastts));
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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, frame->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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return -1;
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}
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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_format *format;
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struct ast_frame *f;
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int codec;
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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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|
}
|
|
|
|
/* Grab the actual payload number for when we create the RTP packet */
|
|
codec = ast_rtp_codecs_payload_code_tx(ast_rtp_instance_get_codecs(instance),
|
|
1, frame->subclass.format, 0);
|
|
if (codec < 0) {
|
|
return -1;
|
|
}
|
|
|
|
format = frame->subclass.format;
|
|
if (!multicast->smoother && ast_format_can_be_smoothed(format)) {
|
|
unsigned int smoother_flags = ast_format_get_smoother_flags(format);
|
|
unsigned int framing_ms = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance));
|
|
|
|
if (!framing_ms && (smoother_flags & AST_SMOOTHER_FLAG_FORCED)) {
|
|
framing_ms = ast_format_get_default_ms(format);
|
|
}
|
|
|
|
if (framing_ms) {
|
|
multicast->smoother = ast_smoother_new((framing_ms * ast_format_get_minimum_bytes(format)) / ast_format_get_minimum_ms(format));
|
|
if (!multicast->smoother) {
|
|
ast_log(LOG_WARNING, "Unable to create smoother: format %s ms: %u len %u\n",
|
|
ast_format_get_name(format), framing_ms, ast_format_get_minimum_bytes(format));
|
|
return -1;
|
|
}
|
|
ast_smoother_set_flags(multicast->smoother, smoother_flags);
|
|
}
|
|
}
|
|
|
|
if (multicast->smoother) {
|
|
if (ast_smoother_test_flag(multicast->smoother, AST_SMOOTHER_FLAG_BE)) {
|
|
ast_smoother_feed_be(multicast->smoother, frame);
|
|
} else {
|
|
ast_smoother_feed(multicast->smoother, frame);
|
|
}
|
|
|
|
while ((f = ast_smoother_read(multicast->smoother)) && f->data.ptr) {
|
|
rtp_raw_write(instance, f, codec);
|
|
}
|
|
} else {
|
|
int hdrlen = 12;
|
|
|
|
/* If we do not have space to construct an RTP header duplicate the frame so we get some */
|
|
if (frame->offset < hdrlen) {
|
|
f = ast_frdup(frame);
|
|
} else {
|
|
f = frame;
|
|
}
|
|
|
|
if (f->data.ptr) {
|
|
rtp_raw_write(instance, f, codec);
|
|
}
|
|
|
|
if (f != frame) {
|
|
ast_frfree(f);
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Function called to read from a multicast instance */
|
|
static struct ast_frame *multicast_rtp_read(struct ast_rtp_instance *instance, int rtcp)
|
|
{
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
static int load_module(void)
|
|
{
|
|
if (ast_rtp_engine_register(&multicast_rtp_engine)) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
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,
|
|
);
|