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Add support for multicast RTP paging.
(closes issue #11797) Reported by: macbrody Review: https://reviewboard.asterisk.org/r/270/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
261
res/res_rtp_multicast.c
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261
res/res_rtp_multicast.c
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/*
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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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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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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/netsock.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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/*! 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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/*! Simple multicast enabled client/receiver paging like Snom and Barix uses */
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MULTICAST_TYPE_BASIC = 0,
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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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unsigned int seqno;
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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 sched_context *sched, struct sockaddr_in *sin, 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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/*! \brief Function called to create a new multicast instance */
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static int multicast_rtp_new(struct ast_rtp_instance *instance, struct sched_context *sched, struct sockaddr_in *sin, void *data)
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{
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struct multicast_rtp *multicast;
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const char *type = 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 (!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_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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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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/*! \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 sockaddr_in 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 (!control_address.sin_addr.s_addr || !remote_address.sin_addr.s_addr) {
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return -1;
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}
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control_packet.ip = remote_address.sin_addr.s_addr;
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control_packet.port = htonl(ntohs(remote_address.sin_port));
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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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sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, (struct sockaddr *)&control_address, sizeof(control_address));
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sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, (struct sockaddr *)&control_address, sizeof(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 sockaddr_in remote_address;
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int hdrlen = 12, res, codec;
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unsigned char *rtpheader;
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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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if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, frame->subclass)) < 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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/* 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++) | (0 << 23)));
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put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8));
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put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc));
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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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res = sendto(multicast->socket, (void *) rtpheader, f->datalen + hdrlen, 0, (struct sockaddr *) &remote_address, sizeof(remote_address));
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if (res < 0) {
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ast_log(LOG_ERROR, "Multicast RTP Transmission error to %s:%u: %s\n",
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ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port), strerror(errno));
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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)
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{
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ast_rtp_engine_unregister(&multicast_rtp_engine);
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return 0;
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}
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AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multicast RTP Engine");
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