multicast RTP: Add dialing options
This adds a new parameter to the end of a multicast RTP dialing string. This parameter defines the following options: * i: Set the interface from which multicast RTP is sent * l: Set whether multicast packets are looped back to the sender * t: Set the TTL for multicast packets * c: Set the codec to use for RTP ASTERISK-26068 #close Reported by Mark Michelson Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219
This commit is contained in:
parent
a6b16d7029
commit
bb0f4a6310
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@ -43,6 +43,7 @@ ASTERISK_REGISTER_FILE()
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#include "asterisk/rtp_engine.h"
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#include "asterisk/causes.h"
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#include "asterisk/format_cache.h"
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#include "asterisk/multicast_rtp.h"
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/* Forward declarations */
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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);
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@ -132,7 +133,9 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
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AST_APP_ARG(type);
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AST_APP_ARG(destination);
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AST_APP_ARG(control);
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AST_APP_ARG(options);
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);
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struct ast_multicast_rtp_options *mcast_options = NULL;
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if (ast_strlen_zero(data)) {
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ast_log(LOG_ERROR, "A multicast type and destination must be given to the 'MulticastRTP' channel\n");
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@ -163,7 +166,15 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
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goto failure;
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}
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fmt = ast_format_cap_get_format(cap, 0);
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mcast_options = ast_multicast_rtp_create_options(args.type, args.options);
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if (!mcast_options) {
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goto failure;
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}
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fmt = ast_multicast_rtp_options_get_format(mcast_options);
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if (!fmt) {
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fmt = ast_format_cap_get_format(cap, 0);
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}
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if (!fmt) {
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ast_log(LOG_ERROR, "No format available for sending RTP to '%s'\n",
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args.destination);
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@ -175,7 +186,7 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
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goto failure;
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}
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instance = ast_rtp_instance_new("multicast", NULL, &control_address, args.type);
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instance = ast_rtp_instance_new("multicast", NULL, &control_address, mcast_options);
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if (!instance) {
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ast_log(LOG_ERROR,
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"Could not create '%s' multicast RTP instance for sending media to '%s'\n",
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@ -207,12 +218,14 @@ static struct ast_channel *multicast_rtp_request(const char *type, struct ast_fo
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ao2_ref(fmt, -1);
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ao2_ref(caps, -1);
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ast_multicast_rtp_free_options(mcast_options);
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return chan;
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failure:
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ao2_cleanup(fmt);
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ao2_cleanup(caps);
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ast_multicast_rtp_free_options(mcast_options);
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*cause = AST_CAUSE_FAILURE;
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return NULL;
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}
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@ -0,0 +1,58 @@
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/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2016, Digium, Inc.
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*
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* Mark Michelson <mmichelson@digium.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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#ifndef MULTICAST_RTP_H_
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#define MULTICAST_RTP_H_
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struct ast_multicast_rtp_options;
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/*!
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* \brief Create multicast RTP options.
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*
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* These are passed to the multicast RTP engine on its creation.
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*
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* \param type The type of multicast RTP, either "basic" or "linksys"
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* \param options Miscellaneous options
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* \retval NULL Failure
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* \retval non-NULL success
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*/
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struct ast_multicast_rtp_options *ast_multicast_rtp_create_options(const char *type,
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const char *options);
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/*!
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* \brief Free multicast RTP options
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*
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* This function is NULL-tolerant
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*
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* \param mcast_options Options to free
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*/
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void ast_multicast_rtp_free_options(struct ast_multicast_rtp_options *mcast_options);
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/*!
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* \brief Get format specified in multicast options
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*
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* Multicast options allow for a format to be selected.
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* This function accesses the selected format and creates
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* an ast_format structure for it.
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*
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* \param mcast_options The options where a codec was specified
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* \retval NULL No format specified in the options
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* \revval non-NULL The format to use for communication
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*/
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struct ast_format *ast_multicast_rtp_options_get_format(struct ast_multicast_rtp_options *mcast_options);
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#endif /* MULTICAST_RTP_H_ */
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@ -54,6 +54,8 @@ ASTERISK_REGISTER_FILE()
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#include "asterisk/module.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/format_cache.h"
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#include "asterisk/multicast_rtp.h"
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#include "asterisk/app.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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@ -63,8 +65,10 @@ ASTERISK_REGISTER_FILE()
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/*! \brief Type of paging to do */
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enum multicast_type {
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/*! Type has not been set yet */
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MULTICAST_TYPE_UNSPECIFIED = 0,
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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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MULTICAST_TYPE_BASIC,
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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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@ -95,6 +99,91 @@ struct multicast_rtp {
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struct timeval txcore;
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};
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enum {
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OPT_CODEC = (1 << 0),
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OPT_LOOP = (1 << 1),
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OPT_TTL = (1 << 2),
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OPT_IF = (1 << 3),
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};
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enum {
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OPT_ARG_CODEC = 0,
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OPT_ARG_LOOP,
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OPT_ARG_TTL,
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OPT_ARG_IF,
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OPT_ARG_ARRAY_SIZE,
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};
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AST_APP_OPTIONS(multicast_rtp_options, BEGIN_OPTIONS
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/*! Set the codec to be used for multicast RTP */
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AST_APP_OPTION_ARG('c', OPT_CODEC, OPT_ARG_CODEC),
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/*! Set whether multicast RTP is looped back to the sender */
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AST_APP_OPTION_ARG('l', OPT_LOOP, OPT_ARG_LOOP),
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/*! Set the hop count for multicast RTP */
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AST_APP_OPTION_ARG('t', OPT_TTL, OPT_ARG_TTL),
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/*! Set the interface from which multicast RTP is sent */
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AST_APP_OPTION_ARG('i', OPT_IF, OPT_ARG_IF),
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END_OPTIONS );
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struct ast_multicast_rtp_options {
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char *type;
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char *options;
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struct ast_format *fmt;
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struct ast_flags opts;
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char *opt_args[OPT_ARG_ARRAY_SIZE];
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/*! The type and options are stored in this buffer */
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char buf[0];
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};
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struct ast_multicast_rtp_options *ast_multicast_rtp_create_options(const char *type,
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const char *options)
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{
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struct ast_multicast_rtp_options *mcast_options;
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char *pos;
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mcast_options = ast_calloc(1, sizeof(*mcast_options)
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+ strlen(type)
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+ strlen(options) + 2);
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if (!mcast_options) {
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return NULL;
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}
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pos = mcast_options->buf;
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/* Safe */
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strcpy(pos, type);
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mcast_options->type = pos;
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pos += strlen(type) + 1;
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/* Safe */
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strcpy(pos, options);
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mcast_options->options = pos;
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if (ast_app_parse_options(multicast_rtp_options, &mcast_options->opts,
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mcast_options->opt_args, mcast_options->options)) {
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ast_log(LOG_WARNING, "Error parsing multicast RTP options\n");
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ast_multicast_rtp_free_options(mcast_options);
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return NULL;
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}
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return mcast_options;
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}
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void ast_multicast_rtp_free_options(struct ast_multicast_rtp_options *mcast_options)
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{
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ast_free(mcast_options);
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}
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struct ast_format *ast_multicast_rtp_options_get_format(struct ast_multicast_rtp_options *mcast_options)
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{
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if (ast_test_flag(&mcast_options->opts, OPT_CODEC)
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&& !ast_strlen_zero(mcast_options->opt_args[OPT_ARG_CODEC])) {
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return ast_format_cache_get(mcast_options->opt_args[OPT_ARG_CODEC]);
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}
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return NULL;
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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 ast_sched_context *sched, struct ast_sockaddr *addr, void *data);
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static int multicast_rtp_activate(struct ast_rtp_instance *instance);
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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 ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
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static int set_type(struct multicast_rtp *multicast, const char *type)
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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_log(LOG_WARNING, "Unrecognized multicast type '%s' specified.\n", type);
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return -1;
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}
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return 0;
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}
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static void set_ttl(int sock, const char *ttl_str)
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{
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int ttl;
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if (ast_strlen_zero(ttl_str)) {
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return;
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}
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ast_debug(3, "Setting multicast TTL to %s\n", ttl_str);
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if (sscanf(ttl_str, "%30d", &ttl) < 1) {
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ast_log(LOG_WARNING, "Inavlid multicast ttl option '%s'\n", ttl_str);
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return;
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}
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if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_TTL, &ttl, sizeof(ttl)) < 0) {
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ast_log(LOG_WARNING, "Could not set multicast ttl to '%s': %s\n",
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ttl_str, strerror(errno));
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}
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}
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static void set_loop(int sock, const char *loop_str)
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{
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unsigned char loop;
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if (ast_strlen_zero(loop_str)) {
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return;
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}
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ast_debug(3, "Setting multicast loop to %s\n", loop_str);
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if (sscanf(loop_str, "%30hhu", &loop) < 1) {
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ast_log(LOG_WARNING, "Invalid multicast loop option '%s'\n", loop_str);
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return;
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}
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if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_LOOP, &loop, sizeof(loop)) < 0) {
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ast_log(LOG_WARNING, "Could not set multicast loop to '%s': %s\n",
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loop_str, strerror(errno));
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}
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}
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static void set_if(int sock, const char *if_str)
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{
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struct in_addr iface;
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if (ast_strlen_zero(if_str)) {
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return;
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}
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ast_debug(3, "Setting multicast if to %s\n", if_str);
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if (!inet_aton(if_str, &iface)) {
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ast_log(LOG_WARNING, "Cannot parse if option '%s'\n", if_str);
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}
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if (setsockopt(sock, IPPROTO_IP, IP_MULTICAST_IF, &iface, sizeof(iface)) < 0) {
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ast_log(LOG_WARNING, "Could not set multicast if to '%s': %s\n",
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if_str, strerror(errno));
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}
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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 ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
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{
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struct multicast_rtp *multicast;
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struct ast_multicast_rtp_options *mcast_options = 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 (set_type(multicast, mcast_options->type)) {
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ast_free(multicast);
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return -1;
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}
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return -1;
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}
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if (ast_test_flag(&mcast_options->opts, OPT_LOOP)) {
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set_loop(multicast->socket, mcast_options->opt_args[OPT_ARG_LOOP]);
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}
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if (ast_test_flag(&mcast_options->opts, OPT_TTL)) {
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set_ttl(multicast->socket, mcast_options->opt_args[OPT_ARG_TTL]);
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}
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if (ast_test_flag(&mcast_options->opts, OPT_IF)) {
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set_if(multicast->socket, mcast_options->opt_args[OPT_ARG_IF]);
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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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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Multicast RTP Engine",
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "Multicast RTP Engine",
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.support_level = AST_MODULE_SUPPORT_CORE,
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.load = load_module,
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.unload = unload_module,
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@ -0,0 +1,6 @@
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{
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global:
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LINKER_SYMBOL_PREFIXast_multicast_rtp*;
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local:
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*;
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};
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