[asterisk-commits] oej: trunk r101218 - in /trunk: apps/app_rtppage.c configs/rtppage.conf.sample

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Wed Jan 30 09:30:39 CST 2008


Author: oej
Date: Wed Jan 30 09:30:38 2008
New Revision: 101218

URL: http://svn.digium.com/view/asterisk?view=rev&rev=101218
Log:
Add rtppage() application to do multicast or unicast RTP paging to SIP phones.

(closes issue #11797)
Reported by: macbrody
Patches: 
      app_rtppage-20080130.c uploaded by macbrody (license 352)


Added:
    trunk/apps/app_rtppage.c   (with props)
    trunk/configs/rtppage.conf.sample   (with props)

Added: trunk/apps/app_rtppage.c
URL: http://svn.digium.com/view/asterisk/trunk/apps/app_rtppage.c?view=auto&rev=101218
==============================================================================
--- trunk/apps/app_rtppage.c (added)
+++ trunk/apps/app_rtppage.c Wed Jan 30 09:30:38 2008
@@ -1,0 +1,556 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * Copyright (C) 2007, Andreas 'MacBrody' Brodmann
+ *
+ * Andreas 'MacBrody' Brodmann <andreas.brodmann at gmail.com>
+ *
+ * Information on how multicast paging works with linksys 
+ * phones was used from FreeSWITCH's mod_esf with permission
+ * from Brian West.
+ *
+ * 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 Application to stream a channel's input to a specified uni-/multicast address
+ *
+ * \author Andreas 'MacBrody' Brodmann <andreas.brodmann at gmail.com>
+ * 
+ * \ingroup applications
+ */
+
+/*** MODULEINFO
+	<defaultenabled>yes</defaultenabled>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <netdb.h>
+#include <netinet/in.h>
+#include <arpa/inet.h>
+
+#include "asterisk/file.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/pbx.h"
+#include "asterisk/module.h"
+#include "asterisk/lock.h"
+#include "asterisk/app.h"
+#include "asterisk/config.h"
+#include "asterisk/acl.h"
+
+#define RTP_PT_ULAW    0
+#define RTP_PT_GSM     3
+#define RTP_PT_ALAW    8
+#define RTP_PT_G729   18
+
+/*! \brief Multicast Group Receiver Type object */
+enum grouptype {
+	MGT_BASIC = 1,    /*!< simple multicast enabled client/receiver like snom, barix */
+	MGT_LINKSYS = 2,  /*!< linksys ipphones; they need a start/stop packet */
+	MGT_CISCO = 3     /*!< cisco phones; they need a http request to their internal web server // NOT YET IMPLEMENTED */
+};
+
+/*! \brief Multicast Group object */
+struct mcast_group {
+	char name[32];                        /*!< name of the group */
+	enum grouptype type;                  /*!< type, see grouptype */
+	int socket;                           /*!< socket used for streaming to this group (each group has got its own socket */
+	int ttl;                              /*!< timetolive to be set on this socket */
+	struct sockaddr_in rtp_address;       /*!< address/port pair where the traffic is sent to */
+	struct sockaddr_in control_address;   /*!< address/port for linksys phones to send the start/stop packet to */
+	AST_LIST_ENTRY(mcast_group) list;     /*!< next element int group list */
+};
+
+/*! \brief RTP header object */
+struct rtp_header {
+	uint16_t flags;
+	uint16_t seqno;
+	uint32_t timestamp;
+	uint32_t ssrc;
+};
+
+/*! \brief Control Packet object as used for linksys phones for start/stop packets */
+struct control_packet {
+	uint32_t unique_id;                    /*!< unique id per command start or stop - not the same for both commands */
+	uint32_t command;                      /*!< the command: 6=start, 7=stop */
+	uint32_t ip;                           /*!< multicast address in network byte order */
+	uint32_t port;                         /*!< udp port to send the data to */
+};
+
+/*! \brief List to hold all the multicast groups defined in the config file */
+static AST_LIST_HEAD_STATIC(groups, mcast_group);
+
+static char *app = "RTPPage";
+static char *synopsis = "RTPPage Application";
+static char *descrip = "  RTPPage(direct|multicast, ip:port[&ip:port]|group[&group2[&group3...]][,codec]): Sends the channel's input to the\n"
+"specified group(s) defined in the config file rtppage.conf.\n"
+"The optional codec may be one of the following:\n"
+"   ulaw - default\n"
+"   alaw\n"
+"   gsm\n"
+"   g729\n"
+"as long as asterisk does not have to translate or respective translators are\n"
+"installed with your asterisk installation. If none or any other codec is\n"
+"specified the application will fall back to ulaw.\n";
+
+static const char config[] = "rtppage.conf";
+static int default_ttl = -1;
+static unsigned int tos = -1;
+
+/*! \brief Read input from channel and send it to the specified group(s) as rtp traffic */
+static int rtppage_exec(struct ast_channel *chan, void *data)
+{
+	int res = 0;
+	struct ast_module_user *u =  NULL;
+	struct ast_frame *f = NULL;
+	char *parse = NULL;
+	char *rest = NULL, *cur = NULL;
+	char *rest2 = NULL;
+	char *ip = NULL, *port = NULL;
+	int ms = -1;
+	unsigned char *databuf = NULL;
+	struct sockaddr_in destaddr;
+	struct mcast_group *group;
+	struct control_packet cpk;
+	struct rtp_header *rtph = NULL;
+	uint8_t rtp_pt = RTP_PT_ULAW;
+	int chan_format = AST_FORMAT_ULAW;
+	uint16_t rtpflags = 0;
+	int ttl = 0;
+	int pagetype = 0;
+	AST_LIST_HEAD(, mcast_group) activegroups;
+
+	/* init active groups */
+	activegroups.first = NULL;
+	activegroups.last = NULL;
+	activegroups.lock = AST_MUTEX_INIT_VALUE;
+
+	/* you can specify three arguments:
+	 * 1) pagetype (0 = direct, 1 = multicast)
+	 * 2) groups, e.g. NameOfGroup or Name1&Name2 etc) / or ip:port in case of direct
+	 * 3) optional: codec, if specified and valid
+	 *    this codec will be used for streaming
+	 */
+	AST_DECLARE_APP_ARGS(args,
+		AST_APP_ARG(pagetype);
+		AST_APP_ARG(groups);
+		AST_APP_ARG(codec);
+	);
+
+	/* make sure there is at least one parameter */
+	if (ast_strlen_zero(data)) {
+		ast_log(LOG_WARNING, "%s requires argument (group(s)[,codec])\n", app);
+		return -1;
+	}
+
+	parse = ast_strdupa(data);
+	AST_STANDARD_APP_ARGS(args, parse);
+
+	/* pagetype is a mandatory parameter */
+	if (!args.pagetype) {
+		ast_log(LOG_WARNING, "%s requires arguments (pagetype, group(s) | ip:port[,codec])\n", app);
+		return(-1);
+	}
+	if (!strcasecmp(args.pagetype, "direct")) {
+		pagetype = 0;
+	} else if (!strcasecmp(args.pagetype, "multicast")) {
+		pagetype = 1;
+	} else {
+		ast_log(LOG_ERROR, "%s is an invalid grouptype! valid types are: direct, multicast.\n", args.pagetype);
+		return(-1);
+	}
+
+	/* group is a mandatory parameter */
+	if (!args.groups) {
+		ast_log(LOG_WARNING, "%s requires arguments (pagetype, group(s) | ip:port[,codec])\n", app);
+		return(-1);
+	}
+
+	/* setup variables for the desired codec */
+	if (args.codec) {
+		if (!strcasecmp(args.codec, "ulaw")) {
+			/* use default settings */
+		} else if (!strcasecmp(args.codec, "alaw")) {
+			rtp_pt = RTP_PT_ALAW;
+			chan_format = AST_FORMAT_ALAW;
+		} else if (!strcasecmp(args.codec, "gsm")) {
+			rtp_pt = RTP_PT_GSM;
+			chan_format = AST_FORMAT_GSM;
+		} else if (!strcasecmp(args.codec, "g729")) {
+			rtp_pt = RTP_PT_G729;
+			chan_format = AST_FORMAT_G729A;
+		} else {
+			/* use ulaw as fallback */
+			rtp_pt = RTP_PT_ULAW;
+			chan_format = AST_FORMAT_ULAW;
+		}
+	}
+
+	u = ast_module_user_add(chan);
+
+	/* Check if the channel is answered, if not
+	 * do answer it */
+	if (chan->_state != AST_STATE_UP) {
+		res = ast_answer(chan);
+		if (res) {
+			ast_log(LOG_WARNING, "Could not answer channel '%s'\n", chan->name);
+			goto end;
+		}
+	}
+
+	/* allocate memory for the rtp send buffer */
+	if ((databuf = ast_calloc(1, 172)) == NULL) {
+		ast_log(LOG_WARNING, "Failed to allocate memory for the data buffer, give up\n");
+		goto end;
+	}
+
+	/* initialize rtp buffer header
+	 * with rtp version and
+	 * payload type
+	 */
+	rtph = (struct rtp_header *)databuf;
+	rtpflags  = (0x02 << 14); /* rtp v2 */
+	rtpflags  = (rtpflags & 0xFF80) |  rtp_pt;  
+	rtph->flags = htons(rtpflags);
+	rtph->ssrc =  htonl((u_long)time(NULL));
+	
+	/* first create a temporary table for this page session
+	 * containing all groups which will be used
+	 */
+	AST_LIST_LOCK(&groups);
+	rest = ast_strdup(args.groups);
+	if (pagetype == 0) {
+		/* a direct page call. this can actually be used
+		 * for multicast paging too by passing the ip:port as
+		 * argument 2 
+		 */
+		while ((cur = strsep(&rest, "&"))) {
+			struct mcast_group *agroup = ast_calloc(1, sizeof(*agroup));
+			rest2 = ast_strdup(cur);
+			ip = strsep(&rest2, ":");
+			port = strsep(&rest2, ":");
+			if (ip == NULL || port == NULL) {
+				ast_log(LOG_WARNING, "invalid ip:port pair in call to RTPPage (%s)!\n", cur);
+				free(agroup);
+				continue;
+			}
+			agroup->rtp_address.sin_family = AF_INET;
+			agroup->rtp_address.sin_port = htons(atoi(port));
+			if (inet_pton(AF_INET, ip, &agroup->rtp_address.sin_addr) <= 0) {
+				ast_log(LOG_WARNING, "invalid ip in call to RTPPage (%s)!\n", cur);
+				free(agroup);
+				continue;
+			}
+			agroup->type = MGT_BASIC;
+			agroup->socket = -1;
+			agroup->ttl = -1;
+			AST_LIST_INSERT_TAIL(&activegroups, agroup, list);
+		}
+	} else if (pagetype == 1) {
+		/* a multicast page call */
+		while ((cur = strsep(&rest, "&"))) {
+			AST_LIST_TRAVERSE(&groups, group, list) {
+				if (!strcasecmp(group->name, cur)) {
+					struct mcast_group *agroup = ast_calloc(1, sizeof(*agroup));
+					memcpy(agroup->name, group->name, 32);
+					agroup->type = group->type;
+					agroup->socket = group->socket;
+					agroup->ttl = group->ttl;
+					memcpy(&agroup->rtp_address, &group->rtp_address, sizeof(agroup->rtp_address));
+					memcpy(&agroup->control_address, &group->control_address, sizeof(agroup->control_address));
+					AST_LIST_INSERT_TAIL(&activegroups, agroup, list);
+				}
+			}
+		}
+	}
+	AST_LIST_UNLOCK(&groups);
+
+	/* now initialize these groups, e.g. create a udp socket for each,
+	 * set ttl and tos if requested by config, and
+	 * in case of linksys type groups send the multicast start signal
+	 */
+	AST_LIST_TRAVERSE(&activegroups, group, list) {
+		group->socket = socket(AF_INET, SOCK_DGRAM, 0);
+		/* set ttl if configured
+		 * ttl can be configured either globally in the
+		 * category 'general' or locally within
+		 * the respective groups
+		 */
+		if (group->ttl >= 0 || default_ttl >= 0) {
+			ttl = default_ttl;
+			if (group->ttl >= 0) {
+				ttl = group->ttl;
+			}
+			if (setsockopt(group->socket, IPPROTO_IP, IP_TTL, &ttl, sizeof(ttl)) < 0) {
+				ast_log(LOG_WARNING, "Failed to set ttl on socket for group %s!\n", group->name);
+			}
+		}
+		/* set tos if requested 
+		 * tos can only be configured globally ('general')
+		 */
+		if (tos >= 0) {
+			if (setsockopt(group->socket, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)) < 0) {
+				ast_log(LOG_WARNING, "Failed to set tos field on socket for group %s!\n", group->name);
+			}
+		}
+		/* for linksys device groups send multicast start command */
+		if (group->type == MGT_LINKSYS) {
+			cpk.unique_id = htonl((u_long)time(NULL));
+			cpk.command = htonl(6);  /* multicast start command */
+			memcpy(&cpk.ip, &group->rtp_address.sin_addr, sizeof(cpk.ip));
+			cpk.port = htonl(ntohs(group->rtp_address.sin_port));
+			memcpy(&destaddr, &group->control_address, sizeof(destaddr));
+			sendto(group->socket, &cpk, sizeof(cpk), 0, (struct sockaddr *)&destaddr, sizeof(destaddr));
+			sendto(group->socket, &cpk, sizeof(cpk), 0, (struct sockaddr *)&destaddr, sizeof(destaddr));
+		}
+	}
+
+	/* Set read format as configured - this codec will be used for streaming */
+	res = ast_set_read_format(chan, chan_format);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set channel read mode, giving up\n");
+		res = -1;
+		goto end;
+	}
+
+	/* Play a beep to let the caller know he can start talking */
+	res = ast_streamfile(chan, "beep", chan->language);
+	if (!res) {
+		res = ast_waitstream(chan, "");
+	} else {
+		ast_log(LOG_WARNING, "ast_streamfile failed on %s\n", chan->name);
+	}
+	ast_stopstream(chan);
+
+	/* main loop: 
+	 * read frames from the input channel and, if they are voice frames,
+	 * send them to all requested multi-/unicast listeners.
+	 */
+	for (;;) {
+		ms = ast_waitfor(chan, 1000);
+		if (ms < 0) {
+			ast_log(LOG_DEBUG, "Hangup detected\n");
+			goto end;
+		}
+		f = ast_read(chan);
+		if (!f)
+			break;
+
+		/* if the speaker pressed '#', then quit */
+		if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '#')) {
+			res = 0;
+			ast_frfree(f);
+			ast_log(LOG_DEBUG, "Received DTMF key: %d\n", f->subclass);
+			goto end;
+		}
+
+		if (f->frametype == AST_FRAME_VOICE) {
+			/* update the rtp header */
+			rtph = (struct rtp_header *)databuf;
+			rtph->seqno = htons(f->seqno);
+			rtph->timestamp = htonl(f->ts * 8);
+			memcpy(databuf+12, f->data, f->datalen);
+
+			/* now send that frame to the destination groups */
+			AST_LIST_TRAVERSE(&activegroups, group, list) {
+				memcpy(&destaddr, &group->rtp_address, sizeof(destaddr));
+				if (sendto(group->socket, databuf, f->datalen+12, 0, (struct sockaddr *)&destaddr, sizeof(destaddr)) <= 0) {
+					ast_log(LOG_DEBUG, "sendto() failed!\n");
+				}
+			}
+		}
+		ast_frfree(f);
+		f = NULL;
+	}
+
+end:
+
+	/* send a stop multicast signal to all linksys devices */
+	AST_LIST_TRAVERSE(&activegroups, group, list) {
+		if (group->socket > 0) {
+			if (group->type == MGT_LINKSYS) {
+				cpk.unique_id = htonl((u_long)time(NULL));
+				cpk.command = htonl(7);  /* multicast stop command */
+				memcpy(&cpk.ip, &group->rtp_address.sin_addr, sizeof(cpk.ip));
+				cpk.port = htonl(ntohs(group->rtp_address.sin_port));
+				memcpy(&destaddr, &group->control_address, sizeof(destaddr));
+				sendto(group->socket, &cpk, 8, 0, (struct sockaddr *)&destaddr, sizeof(destaddr));
+				sendto(group->socket, &cpk, 8, 0, (struct sockaddr *)&destaddr, sizeof(destaddr));
+			}
+			close(group->socket);
+		}
+	}
+
+	/* free activegroups list */
+	while ((group = AST_LIST_REMOVE_HEAD(&activegroups, list))) {
+		free(group);
+	}
+
+	/* free the rtp data buffer */
+	if (databuf != NULL) {
+		free(databuf);
+	}
+
+	ast_module_user_remove(u);
+	ast_log(LOG_DEBUG, "Exit RTPPage(%s)\n", args.groups);
+
+	return res;
+}
+
+static int load_config(int reload) {
+
+	int res = 0;
+	const char *cat = NULL;
+	struct ast_config *cfg = NULL;
+	struct mcast_group *group = NULL;
+	const char *var = NULL;
+	struct ast_flags config_flags = { 0 };
+
+	AST_LIST_LOCK(&groups);
+	if (reload) {
+		/* if this is a reload, then free the config structure before
+		 * filling it again 
+		 */
+		while ((group = AST_LIST_REMOVE_HEAD(&groups, list))) {
+			free(group);
+		}
+
+		/* reset default_ttl & tos */
+		default_ttl = -1; /* means not set */
+		tos = -1;
+	}
+
+	/* load config file */
+	if (!(cfg = ast_config_load(config, config_flags))) {
+		ast_log(LOG_NOTICE, "Failed to load config!\n");
+		AST_LIST_UNLOCK(&groups);
+		return(-1);
+	}
+
+	while ((cat = ast_category_browse(cfg, cat)) != NULL) {
+		/* 'general' is reserved for generic options */
+		if (!strcasecmp(cat, "general")) {
+			var = ast_variable_retrieve(cfg, cat, "ttl");
+			if (var) {
+				default_ttl = atoi(var);
+			}
+			var = ast_variable_retrieve(cfg, cat, "tos");
+			if (var) {
+				ast_str2tos(var, &tos);
+			}
+			continue;
+		}
+
+		group = ast_calloc(1, sizeof(*group));
+		var = ast_variable_retrieve(cfg, cat, "type");
+		if (!strcasecmp(var, "basic")) {
+			ast_copy_string(group->name, cat, sizeof(group->name));
+			group->type = MGT_BASIC;
+			group->socket = -1;
+			group->ttl = -1;
+			if (ast_variable_retrieve(cfg, cat, "ttl") != NULL) {
+				group->ttl = atoi(ast_variable_retrieve(cfg, cat, "ttl"));
+			}
+			memset(&group->rtp_address, 0, sizeof(group->rtp_address));
+			group->rtp_address.sin_family = AF_INET;
+			group->rtp_address.sin_port = htons(atoi(ast_variable_retrieve(cfg, cat, "rtp_port")));
+			if (inet_pton(AF_INET, ast_variable_retrieve(cfg, cat, "rtp_address"), &group->rtp_address.sin_addr) <= 0) {
+				ast_log(LOG_NOTICE, "Invalid ip address in group %s!\n", cat);
+				ast_free(group);
+				group = NULL;
+				continue;
+			}
+		} else if (!strcasecmp(var, "linksys")) {
+			ast_copy_string(group->name, cat, sizeof(group->name));
+			group->type = MGT_LINKSYS;
+			group->socket = -1;
+			group->ttl = -1;
+			if (ast_variable_retrieve(cfg, cat, "ttl") != NULL) {
+				group->ttl = atoi(ast_variable_retrieve(cfg, cat, "ttl"));
+			}
+			memset(&group->rtp_address, 0, sizeof(group->rtp_address));
+			group->rtp_address.sin_family = AF_INET;
+			group->rtp_address.sin_port = htons(atoi(ast_variable_retrieve(cfg, cat, "rtp_port")));
+			if (inet_pton(AF_INET, ast_variable_retrieve(cfg, cat, "rtp_address"), &group->rtp_address.sin_addr) <= 0) {
+				ast_log(LOG_NOTICE, "Invalid ip address in group %s!\n", cat);
+				ast_free(group);
+				group = NULL;
+				continue;
+			}
+			memset(&group->control_address, 0, sizeof(group->control_address));
+			group->control_address.sin_family = AF_INET;
+			group->control_address.sin_port = htons(atoi(ast_variable_retrieve(cfg, cat, "control_port")));
+			if (inet_pton(AF_INET, ast_variable_retrieve(cfg, cat, "control_address"), &group->control_address.sin_addr) <= 0) {
+				ast_log(LOG_NOTICE, "Invalid ip address in group %s!\n", cat);
+				ast_free(group);
+				group = NULL;
+				continue;
+			}
+		} else {
+			group->type = -1;
+			group->socket = -1;
+			group->ttl = -1;
+			ast_log(LOG_NOTICE, "Invalid mcast group %s!\n", cat);
+			continue;
+		}
+
+		/* now add it to the linked list */
+		AST_LIST_INSERT_TAIL(&groups, group, list);
+		ast_log(LOG_NOTICE, "loaded category %s\n", group->name);
+		group = NULL;
+		var = NULL;
+	}
+
+	AST_LIST_UNLOCK(&groups);
+
+	ast_config_destroy(cfg);
+
+	return(res);
+}
+
+static int unload_module(void)
+{
+	int res;
+	res = ast_unregister_application(app);
+	ast_module_user_hangup_all();
+	return res;	
+}
+
+static int load_module(void)
+{
+
+	load_config(0);
+	return ast_register_application(app, rtppage_exec, synopsis, descrip);
+}
+
+static int reload(void)
+{
+	return load_config(1);
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "RTPPage Application",
+	.load = load_module,
+	.unload = unload_module,
+	.reload = reload,
+);
+
+

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Added: trunk/configs/rtppage.conf.sample
URL: http://svn.digium.com/view/asterisk/trunk/configs/rtppage.conf.sample?view=auto&rev=101218
==============================================================================
--- trunk/configs/rtppage.conf.sample (added)
+++ trunk/configs/rtppage.conf.sample Wed Jan 30 09:30:38 2008
@@ -1,0 +1,20 @@
+; Configuration for the rtppage() application 
+; that sends audio in multicast or unicast mode to phones
+; for paging
+
+[general]
+ttl=10
+tos=ef
+
+[testgroup]
+type=basic
+rtp_address=192.168.83.147
+rtp_port=12346
+
+[linksysgroup]
+type=linksys
+rtp_address=224.168.168.168
+rtp_port=34567
+control_address=224.168.168.168
+control_port=6061
+

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