/* * Asterisk -- An open source telephony toolkit. * * Copyright (C) 2009, Digium, Inc. * * Joshua Colp * Andreas 'MacBrody' Brodmann * * See http://www.asterisk.org for more information about * the Asterisk project. Please do not directly contact * any of the maintainers of this project for assistance; * the project provides a web site, mailing lists and IRC * channels for your use. * * This program is free software, distributed under the terms of * the GNU General Public License Version 2. See the LICENSE file * at the top of the source tree. */ /*! * \file * * \brief Multicast RTP Engine * * \author Joshua Colp * \author Andreas 'MacBrody' Brodmann * * \ingroup rtp_engines */ /*** MODULEINFO core ***/ #include "asterisk.h" ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #include #include #include #include #include "asterisk/pbx.h" #include "asterisk/frame.h" #include "asterisk/channel.h" #include "asterisk/acl.h" #include "asterisk/config.h" #include "asterisk/lock.h" #include "asterisk/utils.h" #include "asterisk/cli.h" #include "asterisk/manager.h" #include "asterisk/unaligned.h" #include "asterisk/module.h" #include "asterisk/rtp_engine.h" /*! Command value used for Linksys paging to indicate we are starting */ #define LINKSYS_MCAST_STARTCMD 6 /*! Command value used for Linksys paging to indicate we are stopping */ #define LINKSYS_MCAST_STOPCMD 7 /*! \brief Type of paging to do */ enum multicast_type { /*! Simple multicast enabled client/receiver paging like Snom and Barix uses */ MULTICAST_TYPE_BASIC = 0, /*! More advanced Linksys type paging which requires a start and stop packet */ MULTICAST_TYPE_LINKSYS, }; /*! \brief Structure for a Linksys control packet */ struct multicast_control_packet { /*! Unique identifier for the control packet */ uint32_t unique_id; /*! Actual command in the control packet */ uint32_t command; /*! IP address for the RTP */ uint32_t ip; /*! Port for the RTP */ uint32_t port; }; /*! \brief Structure for a multicast paging instance */ struct multicast_rtp { /*! TYpe of multicast paging this instance is doing */ enum multicast_type type; /*! Socket used for sending the audio on */ int socket; /*! Synchronization source value, used when creating/sending the RTP packet */ unsigned int ssrc; /*! Sequence number, used when creating/sending the RTP packet */ unsigned int seqno; }; /* Forward Declarations */ static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data); static int multicast_rtp_activate(struct ast_rtp_instance *instance); static int multicast_rtp_destroy(struct ast_rtp_instance *instance); static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame); static struct ast_frame *multicast_rtp_read(struct ast_rtp_instance *instance, int rtcp); /* RTP Engine Declaration */ static struct ast_rtp_engine multicast_rtp_engine = { .name = "multicast", .new = multicast_rtp_new, .activate = multicast_rtp_activate, .destroy = multicast_rtp_destroy, .write = multicast_rtp_write, .read = multicast_rtp_read, }; /*! \brief Function called to create a new multicast instance */ static int multicast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data) { struct multicast_rtp *multicast; const char *type = data; if (!(multicast = ast_calloc(1, sizeof(*multicast)))) { return -1; } if (!strcasecmp(type, "basic")) { multicast->type = MULTICAST_TYPE_BASIC; } else if (!strcasecmp(type, "linksys")) { multicast->type = MULTICAST_TYPE_LINKSYS; } else { ast_free(multicast); return -1; } if ((multicast->socket = socket(AF_INET, SOCK_DGRAM, 0)) < 0) { ast_free(multicast); return -1; } multicast->ssrc = ast_random(); ast_rtp_instance_set_data(instance, multicast); return 0; } /*! \brief Helper function which populates a control packet with useful information and sends it */ static int multicast_send_control_packet(struct ast_rtp_instance *instance, struct multicast_rtp *multicast, int command) { struct multicast_control_packet control_packet = { .unique_id = htonl((u_long)time(NULL)), .command = htonl(command), }; struct ast_sockaddr control_address, remote_address; ast_rtp_instance_get_local_address(instance, &control_address); ast_rtp_instance_get_remote_address(instance, &remote_address); /* Ensure the user of us have given us both the control address and destination address */ if (ast_sockaddr_isnull(&control_address) || ast_sockaddr_isnull(&remote_address)) { return -1; } /* The protocol only supports IPv4. */ if (ast_sockaddr_is_ipv6(&remote_address)) { ast_log(LOG_WARNING, "Cannot send control packet for IPv6 " "remote address.\n"); return -1; } control_packet.ip = htonl(ast_sockaddr_ipv4(&remote_address)); control_packet.port = htonl(ast_sockaddr_port(&remote_address)); /* Based on a recommendation by Brian West who did the FreeSWITCH implementation we send control packets twice */ ast_sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, &control_address); ast_sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, &control_address); return 0; } /*! \brief Function called to indicate that audio is now going to flow */ static int multicast_rtp_activate(struct ast_rtp_instance *instance) { struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance); if (multicast->type != MULTICAST_TYPE_LINKSYS) { return 0; } return multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STARTCMD); } /*! \brief Function called to destroy a multicast instance */ static int multicast_rtp_destroy(struct ast_rtp_instance *instance) { struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance); if (multicast->type == MULTICAST_TYPE_LINKSYS) { multicast_send_control_packet(instance, multicast, LINKSYS_MCAST_STOPCMD); } close(multicast->socket); ast_free(multicast); return 0; } /*! \brief Function called to broadcast some audio on a multicast instance */ static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame) { struct multicast_rtp *multicast = ast_rtp_instance_get_data(instance); struct ast_frame *f = frame; struct ast_sockaddr remote_address; int hdrlen = 12, res, codec; unsigned char *rtpheader; /* We only accept audio, nothing else */ if (frame->frametype != AST_FRAME_VOICE) { return 0; } /* Grab the actual payload number for when we create the RTP packet */ if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, &frame->subclass.format, 0)) < 0) { return -1; } /* 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); } /* Construct an RTP header for our packet */ rtpheader = (unsigned char *)(f->data.ptr - hdrlen); put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno++) | (0 << 23))); put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8)); put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc)); /* Finally send it out to the eager phones listening for us */ ast_rtp_instance_get_remote_address(instance, &remote_address); res = ast_sendto(multicast->socket, (void *) rtpheader, f->datalen + hdrlen, 0, &remote_address); if (res < 0) { ast_log(LOG_ERROR, "Multicast RTP Transmission error to %s: %s\n", ast_sockaddr_stringify(&remote_address), strerror(errno)); } /* If we were forced to duplicate the frame free the new one */ if (frame != f) { ast_frfree(f); } return res; } /*! \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_LOAD_ORDER, "Multicast RTP Engine", .load = load_module, .unload = unload_module, .load_pri = AST_MODPRI_CHANNEL_DEPEND, );