[svn-commits] jpeeler: branch jpeeler/asterisk-sigwork-trunk r203252 - in /team/jpeeler/ast...
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Thu Jun 25 14:21:05 CDT 2009
Author: jpeeler
Date: Thu Jun 25 14:21:01 2009
New Revision: 203252
URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=203252
Log:
add missing files
Added:
team/jpeeler/asterisk-sigwork-trunk/channels/chan_multicast_rtp.c (with props)
team/jpeeler/asterisk-sigwork-trunk/res/res_rtp_multicast.c (with props)
Added: team/jpeeler/asterisk-sigwork-trunk/channels/chan_multicast_rtp.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/jpeeler/asterisk-sigwork-trunk/channels/chan_multicast_rtp.c?view=auto&rev=203252
==============================================================================
--- team/jpeeler/asterisk-sigwork-trunk/channels/chan_multicast_rtp.c (added)
+++ team/jpeeler/asterisk-sigwork-trunk/channels/chan_multicast_rtp.c Thu Jun 25 14:21:01 2009
@@ -1,0 +1,184 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2009, Digium, Inc.
+ *
+ * Joshua Colp <jcolp at digium.com>
+ * Andreas 'MacBrody' Brodmann <andreas.brodmann at gmail.com>
+ *
+ * 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
+ *
+ * \author Joshua Colp <jcolp at digium.com>
+ * \author Andreas 'MacBrody' Broadmann <andreas.brodmann at gmail.com>
+ *
+ * \brief Multicast RTP Paging Channel
+ *
+ * \ingroup channel_drivers
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <fcntl.h>
+#include <sys/signal.h>
+
+#include "asterisk/lock.h"
+#include "asterisk/channel.h"
+#include "asterisk/config.h"
+#include "asterisk/module.h"
+#include "asterisk/pbx.h"
+#include "asterisk/sched.h"
+#include "asterisk/io.h"
+#include "asterisk/acl.h"
+#include "asterisk/callerid.h"
+#include "asterisk/file.h"
+#include "asterisk/cli.h"
+#include "asterisk/app.h"
+#include "asterisk/rtp_engine.h"
+#include "asterisk/causes.h"
+
+static const char tdesc[] = "Multicast RTP Paging Channel Driver";
+
+/* Forward declarations */
+static struct ast_channel *multicast_rtp_request(const char *type, int format, void *data, int *cause);
+static int multicast_rtp_call(struct ast_channel *ast, char *dest, int timeout);
+static int multicast_rtp_hangup(struct ast_channel *ast);
+static struct ast_frame *multicast_rtp_read(struct ast_channel *ast);
+static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f);
+
+/* Channel driver declaration */
+static const struct ast_channel_tech multicast_rtp_tech = {
+ .type = "MulticastRTP",
+ .description = tdesc,
+ .capabilities = -1,
+ .requester = multicast_rtp_request,
+ .call = multicast_rtp_call,
+ .hangup = multicast_rtp_hangup,
+ .read = multicast_rtp_read,
+ .write = multicast_rtp_write,
+};
+
+/*! \brief Function called when we should read a frame from the channel */
+static struct ast_frame *multicast_rtp_read(struct ast_channel *ast)
+{
+ return &ast_null_frame;
+}
+
+/*! \brief Function called when we should write a frame to the channel */
+static int multicast_rtp_write(struct ast_channel *ast, struct ast_frame *f)
+{
+ struct ast_rtp_instance *instance = ast->tech_pvt;
+
+ return ast_rtp_instance_write(instance, f);
+}
+
+/*! \brief Function called when we should actually call the destination */
+static int multicast_rtp_call(struct ast_channel *ast, char *dest, int timeout)
+{
+ struct ast_rtp_instance *instance = ast->tech_pvt;
+
+ ast_queue_control(ast, AST_CONTROL_ANSWER);
+
+ return ast_rtp_instance_activate(instance);
+}
+
+/*! \brief Function called when we should hang the channel up */
+static int multicast_rtp_hangup(struct ast_channel *ast)
+{
+ struct ast_rtp_instance *instance = ast->tech_pvt;
+
+ ast_rtp_instance_destroy(instance);
+
+ ast->tech_pvt = NULL;
+
+ return 0;
+}
+
+/*! \brief Function called when we should prepare to call the destination */
+static struct ast_channel *multicast_rtp_request(const char *type, int format, void *data, int *cause)
+{
+ char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control;
+ struct ast_rtp_instance *instance;
+ struct sockaddr_in control_address = { .sin_family = AF_INET, }, destination_address = { .sin_family = AF_INET, };
+ struct ast_channel *chan;
+ int fmt = ast_best_codec(format);
+
+ /* If no type was given we can't do anything */
+ if (ast_strlen_zero(multicast_type)) {
+ goto failure;
+ }
+
+ if (!(destination = strchr(tmp, '/'))) {
+ goto failure;
+ }
+ *destination++ = '\0';
+
+ if (ast_parse_arg(destination, PARSE_INADDR | PARSE_PORT_REQUIRE, &destination_address)) {
+ goto failure;
+ }
+
+ if ((control = strchr(destination, '/'))) {
+ *control++ = '\0';
+ if (ast_parse_arg(control, PARSE_INADDR | PARSE_PORT_REQUIRE, &control_address)) {
+ goto failure;
+ }
+ }
+
+ if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, multicast_type))) {
+ goto failure;
+ }
+
+ if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", 0, "MulticastRTP/%p", instance))) {
+ ast_rtp_instance_destroy(instance);
+ goto failure;
+ }
+
+ ast_rtp_instance_set_remote_address(instance, &destination_address);
+
+ chan->tech = &multicast_rtp_tech;
+ chan->nativeformats = fmt;
+ chan->writeformat = fmt;
+ chan->readformat = fmt;
+ chan->rawwriteformat = fmt;
+ chan->rawreadformat = fmt;
+ chan->tech_pvt = instance;
+
+ return chan;
+
+failure:
+ *cause = AST_CAUSE_FAILURE;
+ return NULL;
+}
+
+/*! \brief Function called when our module is loaded */
+static int load_module(void)
+{
+ if (ast_channel_register(&multicast_rtp_tech)) {
+ ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
+ return AST_MODULE_LOAD_DECLINE;
+ }
+
+ return AST_MODULE_LOAD_SUCCESS;
+}
+
+/*! \brief Function called when our module is unloaded */
+static int unload_module(void)
+{
+ ast_channel_unregister(&multicast_rtp_tech);
+
+ return 0;
+}
+
+AST_MODULE_INFO_STANDARD(ASTERISK_GPL_KEY, "Multicast RTP Paging Channel");
Propchange: team/jpeeler/asterisk-sigwork-trunk/channels/chan_multicast_rtp.c
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Propchange: team/jpeeler/asterisk-sigwork-trunk/channels/chan_multicast_rtp.c
------------------------------------------------------------------------------
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Added: team/jpeeler/asterisk-sigwork-trunk/res/res_rtp_multicast.c
URL: http://svn.asterisk.org/svn-view/asterisk/team/jpeeler/asterisk-sigwork-trunk/res/res_rtp_multicast.c?view=auto&rev=203252
==============================================================================
--- team/jpeeler/asterisk-sigwork-trunk/res/res_rtp_multicast.c (added)
+++ team/jpeeler/asterisk-sigwork-trunk/res/res_rtp_multicast.c Thu Jun 25 14:21:01 2009
@@ -1,0 +1,261 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2009, Digium, Inc.
+ *
+ * Joshua Colp <jcolp at digium.com>
+ * Andreas 'MacBrody' Brodmann <andreas.brodmann at gmail.com>
+ *
+ * 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 <jcolp at digium.com>
+ * \author Andreas 'MacBrody' Brodmann <andreas.brodmann at gmail.com>
+ */
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <sys/time.h>
+#include <signal.h>
+#include <fcntl.h>
+#include <math.h>
+
+#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/netsock.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 sched_context *sched, struct sockaddr_in *sin, 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 sched_context *sched, struct sockaddr_in *sin, 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 sockaddr_in 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 (!control_address.sin_addr.s_addr || !remote_address.sin_addr.s_addr) {
+ return -1;
+ }
+
+ control_packet.ip = remote_address.sin_addr.s_addr;
+ control_packet.port = htonl(ntohs(remote_address.sin_port));
+
+ /* Based on a recommendation by Brian West who did the FreeSWITCH implementation we send control packets twice */
+ sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, (struct sockaddr *)&control_address, sizeof(control_address));
+ sendto(multicast->socket, &control_packet, sizeof(control_packet), 0, (struct sockaddr *)&control_address, sizeof(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 sockaddr_in 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)) < 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 = sendto(multicast->socket, (void *) rtpheader, f->datalen + hdrlen, 0, (struct sockaddr *) &remote_address, sizeof(remote_address));
+
+ if (res < 0) {
+ ast_log(LOG_ERROR, "Multicast RTP Transmission error to %s:%u: %s\n",
+ ast_inet_ntoa(remote_address.sin_addr), ntohs(remote_address.sin_port), 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_STANDARD(ASTERISK_GPL_KEY, "Multicast RTP Engine");
Propchange: team/jpeeler/asterisk-sigwork-trunk/res/res_rtp_multicast.c
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