From: Joshua Colp Date: Fri, 12 Sep 2014 17:42:15 +0000 (+0000) Subject: chan_rtp: Add unicast RTP support. X-Git-Tag: 14.0.0-beta1~1665 X-Git-Url: http://git.ipfire.org/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=02295456efde5b930b3b6f40454dad3a9aca0524;p=thirdparty%2Fasterisk.git chan_rtp: Add unicast RTP support. This module supports sending both unicast and multicast RTP to a specified target. Multicast functionality is the same as chan_multicast_rtp was. In the case of unicast a specific IP address and port can be specified, along with optional RTP engine and format in the form of: UnicastRTP/:// This can be useful for sending a copy of a media stream to another application for processing. Review: https://reviewboard.asterisk.org/r/3981/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- diff --git a/channels/chan_multicast_rtp.c b/channels/chan_multicast_rtp.c deleted file mode 100644 index 267baabf1a..0000000000 --- a/channels/chan_multicast_rtp.c +++ /dev/null @@ -1,223 +0,0 @@ -/* - * 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 - * - * \author Joshua Colp - * \author Andreas 'MacBrody' Broadmann - * - * \brief Multicast RTP Paging Channel - * - * \ingroup channel_drivers - */ - -/*** MODULEINFO - core - ***/ - -#include "asterisk.h" - -ASTERISK_FILE_VERSION(__FILE__, "$Revision$") - -#include -#include - -#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, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause); -static int multicast_rtp_call(struct ast_channel *ast, const 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 struct ast_channel_tech multicast_rtp_tech = { - .type = "MulticastRTP", - .description = tdesc, - .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_channel_tech_pvt(ast); - - 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, const char *dest, int timeout) -{ - struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast); - - 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_channel_tech_pvt(ast); - - ast_rtp_instance_destroy(instance); - - ast_channel_tech_pvt_set(ast, NULL); - - return 0; -} - -/*! \brief Function called when we should prepare to call the destination */ -static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause) -{ - char *tmp = ast_strdupa(data), *multicast_type = tmp, *destination, *control; - struct ast_rtp_instance *instance; - struct ast_sockaddr control_address; - struct ast_sockaddr destination_address; - struct ast_channel *chan; - struct ast_format_cap *caps = NULL; - struct ast_format *fmt = NULL; - - fmt = ast_format_cap_get_format(cap, 0); - - ast_sockaddr_setnull(&control_address); - - /* 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 ((control = strchr(destination, '/'))) { - *control++ = '\0'; - if (!ast_sockaddr_parse(&control_address, control, - PARSE_PORT_REQUIRE)) { - goto failure; - } - } - - if (!ast_sockaddr_parse(&destination_address, destination, - PARSE_PORT_REQUIRE)) { - goto failure; - } - - caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); - if (!caps) { - 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, "", "", "", "", "", assignedids, requestor, 0, "MulticastRTP/%p", instance))) { - ast_rtp_instance_destroy(instance); - goto failure; - } - ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan)); - ast_rtp_instance_set_remote_address(instance, &destination_address); - - ast_channel_tech_set(chan, &multicast_rtp_tech); - - ast_format_cap_append(caps, fmt, 0); - ast_channel_nativeformats_set(chan, caps); - ast_channel_set_writeformat(chan, fmt); - ast_channel_set_rawwriteformat(chan, fmt); - ast_channel_set_readformat(chan, fmt); - ast_channel_set_rawreadformat(chan, fmt); - - ast_channel_tech_pvt_set(chan, instance); - - ast_channel_unlock(chan); - - ao2_ref(fmt, -1); - ao2_ref(caps, -1); - - return chan; - -failure: - ao2_cleanup(fmt); - ao2_cleanup(caps); - *cause = AST_CAUSE_FAILURE; - return NULL; -} - -/*! \brief Function called when our module is loaded */ -static int load_module(void) -{ - if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) { - return AST_MODULE_LOAD_DECLINE; - } - ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN); - if (ast_channel_register(&multicast_rtp_tech)) { - ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n"); - ao2_ref(multicast_rtp_tech.capabilities, -1); - multicast_rtp_tech.capabilities = NULL; - 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); - ao2_ref(multicast_rtp_tech.capabilities, -1); - multicast_rtp_tech.capabilities = NULL; - - return 0; -} - -AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Paging Channel", - .support_level = AST_MODULE_SUPPORT_CORE, - .load = load_module, - .unload = unload_module, - .load_pri = AST_MODPRI_CHANNEL_DRIVER, -); diff --git a/channels/chan_rtp.c b/channels/chan_rtp.c new file mode 100644 index 0000000000..97fbf9f318 --- /dev/null +++ b/channels/chan_rtp.c @@ -0,0 +1,335 @@ +/* + * Asterisk -- An open source telephony toolkit. + * + * Copyright (C) 2009 - 2014, 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 + * + * \author Joshua Colp + * \author Andreas 'MacBrody' Broadmann + * + * \brief RTP (Multicast and Unicast) Media Channel + * + * \ingroup channel_drivers + */ + +/*** MODULEINFO + core + ***/ + +#include "asterisk.h" + +ASTERISK_FILE_VERSION(__FILE__, "$Revision$") + +#include "asterisk/channel.h" +#include "asterisk/module.h" +#include "asterisk/pbx.h" +#include "asterisk/acl.h" +#include "asterisk/app.h" +#include "asterisk/rtp_engine.h" +#include "asterisk/causes.h" +#include "asterisk/format_cache.h" + +/* Forward declarations */ +static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause); +static struct ast_channel *unicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause); +static int rtp_call(struct ast_channel *ast, const char *dest, int timeout); +static int rtp_hangup(struct ast_channel *ast); +static struct ast_frame *rtp_read(struct ast_channel *ast); +static int rtp_write(struct ast_channel *ast, struct ast_frame *f); + +/* Multicast channel driver declaration */ +static struct ast_channel_tech multicast_rtp_tech = { + .type = "MulticastRTP", + .description = "Multicast RTP Paging Channel Driver", + .requester = multicast_rtp_request, + .call = rtp_call, + .hangup = rtp_hangup, + .read = rtp_read, + .write = rtp_write, +}; + +/* Unicast channel driver declaration */ +static struct ast_channel_tech unicast_rtp_tech = { + .type = "UnicastRTP", + .description = "Unicast RTP Media Channel Driver", + .requester = unicast_rtp_request, + .call = rtp_call, + .hangup = rtp_hangup, + .read = rtp_read, + .write = rtp_write, +}; + +/*! \brief Function called when we should read a frame from the channel */ +static struct ast_frame *rtp_read(struct ast_channel *ast) +{ + struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast); + int fdno = ast_channel_fdno(ast); + + switch (fdno) { + case 0: + return ast_rtp_instance_read(instance, 0); + default: + return &ast_null_frame; + } +} + +/*! \brief Function called when we should write a frame to the channel */ +static int rtp_write(struct ast_channel *ast, struct ast_frame *f) +{ + struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast); + + return ast_rtp_instance_write(instance, f); +} + +/*! \brief Function called when we should actually call the destination */ +static int rtp_call(struct ast_channel *ast, const char *dest, int timeout) +{ + struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast); + + 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 rtp_hangup(struct ast_channel *ast) +{ + struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast); + + ast_rtp_instance_destroy(instance); + + ast_channel_tech_pvt_set(ast, NULL); + + return 0; +} + +/*! \brief Function called when we should prepare to call the multicast destination */ +static struct ast_channel *multicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause) +{ + char *parse; + struct ast_rtp_instance *instance; + struct ast_sockaddr control_address; + struct ast_sockaddr destination_address; + struct ast_channel *chan; + struct ast_format_cap *caps = NULL; + struct ast_format *fmt = NULL; + AST_DECLARE_APP_ARGS(args, + AST_APP_ARG(type); + AST_APP_ARG(destination); + AST_APP_ARG(control); + ); + + if (ast_strlen_zero(data)) { + ast_log(LOG_ERROR, "A multicast type and destination must be given to the 'MulticastRTP' channel\n"); + goto failure; + } + parse = ast_strdupa(data); + AST_NONSTANDARD_APP_ARGS(args, parse, '/'); + + fmt = ast_format_cap_get_format(cap, 0); + + ast_sockaddr_setnull(&control_address); + + if (!ast_strlen_zero(args.control) && + !ast_sockaddr_parse(&control_address, args.control, PARSE_PORT_REQUIRE)) { + ast_log(LOG_ERROR, "Control address '%s' could not be parsed\n", args.control); + goto failure; + } + + if (!ast_sockaddr_parse(&destination_address, args.destination, + PARSE_PORT_REQUIRE)) { + ast_log(LOG_ERROR, "Destination address '%s' could not be parsed\n", args.destination); + goto failure; + } + + caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + if (!caps) { + goto failure; + } + + if (!(instance = ast_rtp_instance_new("multicast", NULL, &control_address, args.type))) { + ast_log(LOG_ERROR, "Could not create RTP instance for sending media to '%s'\n", args.destination); + goto failure; + } + + if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids, requestor, 0, "MulticastRTP/%p", instance))) { + ast_rtp_instance_destroy(instance); + goto failure; + } + ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan)); + ast_rtp_instance_set_remote_address(instance, &destination_address); + + ast_channel_tech_set(chan, &multicast_rtp_tech); + + ast_format_cap_append(caps, fmt, 0); + ast_channel_nativeformats_set(chan, caps); + ast_channel_set_writeformat(chan, fmt); + ast_channel_set_rawwriteformat(chan, fmt); + ast_channel_set_readformat(chan, fmt); + ast_channel_set_rawreadformat(chan, fmt); + + ast_channel_tech_pvt_set(chan, instance); + + ast_channel_unlock(chan); + + ao2_ref(fmt, -1); + ao2_ref(caps, -1); + + return chan; + +failure: + ao2_cleanup(fmt); + ao2_cleanup(caps); + *cause = AST_CAUSE_FAILURE; + return NULL; +} + +/*! \brief Function called when we should prepare to call the unicast destination */ +static struct ast_channel *unicast_rtp_request(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *data, int *cause) +{ + char *parse; + struct ast_rtp_instance *instance; + struct ast_sockaddr address; + struct ast_sockaddr local_address; + struct ast_channel *chan; + struct ast_format_cap *caps = NULL; + struct ast_format *fmt = NULL; + AST_DECLARE_APP_ARGS(args, + AST_APP_ARG(destination); + AST_APP_ARG(engine); + AST_APP_ARG(format); + ); + + if (ast_strlen_zero(data)) { + goto failure; + } + parse = ast_strdupa(data); + AST_NONSTANDARD_APP_ARGS(args, parse, '/'); + + if (!ast_strlen_zero(args.format)) { + fmt = ast_format_cache_get(args.format); + } else { + fmt = ast_format_cap_get_format(cap, 0); + } + + if (!fmt) { + ast_log(LOG_ERROR, "No format specified for sending RTP to '%s'\n", args.destination); + goto failure; + } + + if (!ast_sockaddr_parse(&address, args.destination, + PARSE_PORT_REQUIRE)) { + ast_log(LOG_ERROR, "Destination '%s' could not be parsed\n", args.destination); + goto failure; + } + + caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT); + if (!caps) { + goto failure; + } + + ast_ouraddrfor(&address, &local_address); + if (!(instance = ast_rtp_instance_new(args.engine, NULL, &local_address, NULL))) { + ast_log(LOG_ERROR, "Could not create RTP instance for sending media to '%s'\n", args.destination); + goto failure; + } + + if (!(chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids, requestor, 0, "UnicastRTP/%s-%p", args.destination, instance))) { + ast_rtp_instance_destroy(instance); + goto failure; + } + ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan)); + ast_rtp_instance_set_remote_address(instance, &address); + ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(instance, 0)); + + ast_channel_tech_set(chan, &unicast_rtp_tech); + + ast_format_cap_append(caps, fmt, 0); + ast_channel_nativeformats_set(chan, caps); + ast_channel_set_writeformat(chan, fmt); + ast_channel_set_rawwriteformat(chan, fmt); + ast_channel_set_readformat(chan, fmt); + ast_channel_set_rawreadformat(chan, fmt); + + ast_channel_tech_pvt_set(chan, instance); + + pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS", ast_sockaddr_stringify_addr(&local_address)); + ast_rtp_instance_get_local_address(instance, &local_address); + pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT", ast_sockaddr_stringify_port(&local_address)); + + ast_channel_unlock(chan); + + ao2_ref(fmt, -1); + ao2_ref(caps, -1); + + return chan; + +failure: + ao2_cleanup(fmt); + ao2_cleanup(caps); + *cause = AST_CAUSE_FAILURE; + return NULL; +} + +/*! \brief Function called when our module is unloaded */ +static int unload_module(void) +{ + ast_channel_unregister(&multicast_rtp_tech); + ao2_cleanup(multicast_rtp_tech.capabilities); + multicast_rtp_tech.capabilities = NULL; + + ast_channel_unregister(&unicast_rtp_tech); + ao2_cleanup(unicast_rtp_tech.capabilities); + unicast_rtp_tech.capabilities = NULL; + + return 0; +} + +/*! \brief Function called when our module is loaded */ +static int load_module(void) +{ + if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) { + return AST_MODULE_LOAD_DECLINE; + } + ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN); + if (ast_channel_register(&multicast_rtp_tech)) { + ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n"); + unload_module(); + return AST_MODULE_LOAD_DECLINE; + } + + if (!(unicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) { + unload_module(); + return AST_MODULE_LOAD_DECLINE; + } + ast_format_cap_append_by_type(unicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN); + if (ast_channel_register(&unicast_rtp_tech)) { + ast_log(LOG_ERROR, "Unable to register channel class 'UnicastRTP'\n"); + unload_module(); + return AST_MODULE_LOAD_DECLINE; + } + + return AST_MODULE_LOAD_SUCCESS; +} + +AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "RTP Media Channel", + .support_level = AST_MODULE_SUPPORT_CORE, + .load = load_module, + .unload = unload_module, + .load_pri = AST_MODPRI_CHANNEL_DRIVER, +);