unicast_rtp_request() could pass an uninitialized 'us' parameter to ast_ouraddrfor(). If ast_ouraddrfor() returns an error then the 'us' parameter may not get initialized. Thus when the code tries to save the 'us' parameter to the local address we could try to copy a ridiculous sized memory buffer and segfault. * Made pass an initialized 'us' parameter to ast_ouraddrfor() and abort the UnicastRTP channel request if it fails. ASTERISK-26672 Change-Id: I1ef7a7c09f4da4f15dcb6de660d2bcac5f2a95c0
419 lines
12 KiB
C
419 lines
12 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 2009 - 2014, Digium, Inc.
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*
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* Joshua Colp <jcolp@digium.com>
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* Andreas 'MacBrody' Brodmann <andreas.brodmann@gmail.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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/*! \file
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*
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* \author Joshua Colp <jcolp@digium.com>
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* \author Andreas 'MacBrody' Broadmann <andreas.brodmann@gmail.com>
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*
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* \brief RTP (Multicast and Unicast) Media Channel
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*
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* \ingroup channel_drivers
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*/
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/*** MODULEINFO
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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#include "asterisk/channel.h"
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#include "asterisk/module.h"
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#include "asterisk/pbx.h"
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#include "asterisk/acl.h"
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#include "asterisk/app.h"
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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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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);
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static int rtp_call(struct ast_channel *ast, const char *dest, int timeout);
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static int rtp_hangup(struct ast_channel *ast);
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static struct ast_frame *rtp_read(struct ast_channel *ast);
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static int rtp_write(struct ast_channel *ast, struct ast_frame *f);
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/* Multicast channel driver declaration */
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static struct ast_channel_tech multicast_rtp_tech = {
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.type = "MulticastRTP",
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.description = "Multicast RTP Paging Channel Driver",
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.requester = multicast_rtp_request,
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.call = rtp_call,
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.hangup = rtp_hangup,
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.read = rtp_read,
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.write = rtp_write,
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};
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/* Unicast channel driver declaration */
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static struct ast_channel_tech unicast_rtp_tech = {
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.type = "UnicastRTP",
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.description = "Unicast RTP Media Channel Driver",
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.requester = unicast_rtp_request,
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.call = rtp_call,
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.hangup = rtp_hangup,
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.read = rtp_read,
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.write = rtp_write,
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};
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/*! \brief Function called when we should read a frame from the channel */
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static struct ast_frame *rtp_read(struct ast_channel *ast)
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{
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struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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int fdno = ast_channel_fdno(ast);
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switch (fdno) {
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case 0:
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return ast_rtp_instance_read(instance, 0);
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default:
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return &ast_null_frame;
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}
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}
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/*! \brief Function called when we should write a frame to the channel */
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static int rtp_write(struct ast_channel *ast, struct ast_frame *f)
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{
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struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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return ast_rtp_instance_write(instance, f);
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}
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/*! \brief Function called when we should actually call the destination */
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static int rtp_call(struct ast_channel *ast, const char *dest, int timeout)
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{
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struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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ast_queue_control(ast, AST_CONTROL_ANSWER);
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return ast_rtp_instance_activate(instance);
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}
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/*! \brief Function called when we should hang the channel up */
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static int rtp_hangup(struct ast_channel *ast)
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{
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struct ast_rtp_instance *instance = ast_channel_tech_pvt(ast);
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ast_rtp_instance_destroy(instance);
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ast_channel_tech_pvt_set(ast, NULL);
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return 0;
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}
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/*! \brief Function called when we should prepare to call the multicast destination */
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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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{
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char *parse;
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struct ast_rtp_instance *instance;
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struct ast_sockaddr control_address;
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struct ast_sockaddr destination_address;
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struct ast_channel *chan;
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struct ast_format_cap *caps = NULL;
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struct ast_format *fmt = NULL;
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AST_DECLARE_APP_ARGS(args,
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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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goto failure;
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}
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parse = ast_strdupa(data);
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AST_NONSTANDARD_APP_ARGS(args, parse, '/');
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if (ast_strlen_zero(args.type)) {
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ast_log(LOG_ERROR, "Type is required for the 'MulticastRTP' channel\n");
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goto failure;
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}
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if (ast_strlen_zero(args.destination)) {
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ast_log(LOG_ERROR, "Destination is required for the 'MulticastRTP' channel\n");
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goto failure;
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}
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if (!ast_sockaddr_parse(&destination_address, args.destination, PARSE_PORT_REQUIRE)) {
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ast_log(LOG_ERROR, "Destination address '%s' could not be parsed\n",
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args.destination);
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goto failure;
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}
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ast_sockaddr_setnull(&control_address);
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if (!ast_strlen_zero(args.control)
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&& !ast_sockaddr_parse(&control_address, args.control, PARSE_PORT_REQUIRE)) {
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ast_log(LOG_ERROR, "Control address '%s' could not be parsed\n", args.control);
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goto failure;
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}
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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 codec available for sending RTP to '%s'\n",
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args.destination);
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goto failure;
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}
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caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
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if (!caps) {
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goto failure;
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}
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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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args.type, args.destination);
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goto failure;
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}
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chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
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requestor, 0, "MulticastRTP/%p", instance);
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if (!chan) {
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ast_rtp_instance_destroy(instance);
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goto failure;
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}
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ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan));
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ast_rtp_instance_set_remote_address(instance, &destination_address);
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ast_channel_tech_set(chan, &multicast_rtp_tech);
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ast_format_cap_append(caps, fmt, 0);
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ast_channel_nativeformats_set(chan, caps);
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ast_channel_set_writeformat(chan, fmt);
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ast_channel_set_rawwriteformat(chan, fmt);
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ast_channel_set_readformat(chan, fmt);
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ast_channel_set_rawreadformat(chan, fmt);
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ast_channel_tech_pvt_set(chan, instance);
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ast_channel_unlock(chan);
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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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enum {
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OPT_RTP_CODEC = (1 << 0),
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OPT_RTP_ENGINE = (1 << 1),
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};
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enum {
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OPT_ARG_RTP_CODEC,
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OPT_ARG_RTP_ENGINE,
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/* note: this entry _MUST_ be the last one in the enum */
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OPT_ARG_ARRAY_SIZE
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};
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AST_APP_OPTIONS(unicast_rtp_options, BEGIN_OPTIONS
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/*! Set the codec to be used for unicast RTP */
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AST_APP_OPTION_ARG('c', OPT_RTP_CODEC, OPT_ARG_RTP_CODEC),
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/*! Set the RTP engine to use for unicast RTP */
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AST_APP_OPTION_ARG('e', OPT_RTP_ENGINE, OPT_ARG_RTP_ENGINE),
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END_OPTIONS );
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/*! \brief Function called when we should prepare to call the unicast destination */
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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)
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{
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char *parse;
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struct ast_rtp_instance *instance;
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struct ast_sockaddr address;
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struct ast_sockaddr local_address;
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struct ast_channel *chan;
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struct ast_format_cap *caps = NULL;
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struct ast_format *fmt = NULL;
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const char *engine_name;
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AST_DECLARE_APP_ARGS(args,
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AST_APP_ARG(destination);
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AST_APP_ARG(options);
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);
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struct ast_flags opts = { 0, };
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char *opt_args[OPT_ARG_ARRAY_SIZE];
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if (ast_strlen_zero(data)) {
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ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
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goto failure;
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}
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parse = ast_strdupa(data);
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AST_NONSTANDARD_APP_ARGS(args, parse, '/');
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if (ast_strlen_zero(args.destination)) {
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ast_log(LOG_ERROR, "Destination is required for the 'UnicastRTP' channel\n");
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goto failure;
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}
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if (!ast_sockaddr_parse(&address, args.destination, PARSE_PORT_REQUIRE)) {
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ast_log(LOG_ERROR, "Destination '%s' could not be parsed\n", args.destination);
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goto failure;
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}
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if (!ast_strlen_zero(args.options)
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&& ast_app_parse_options(unicast_rtp_options, &opts, opt_args,
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ast_strdupa(args.options))) {
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ast_log(LOG_ERROR, "'UnicastRTP' channel options '%s' parse error\n",
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args.options);
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goto failure;
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}
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if (ast_test_flag(&opts, OPT_RTP_CODEC)
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&& !ast_strlen_zero(opt_args[OPT_ARG_RTP_CODEC])) {
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fmt = ast_format_cache_get(opt_args[OPT_ARG_RTP_CODEC]);
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if (!fmt) {
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ast_log(LOG_ERROR, "Codec '%s' not found for sending RTP to '%s'\n",
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opt_args[OPT_ARG_RTP_CODEC], args.destination);
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goto failure;
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}
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} else {
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fmt = ast_format_cap_get_format(cap, 0);
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if (!fmt) {
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ast_log(LOG_ERROR, "No codec available for sending RTP to '%s'\n",
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args.destination);
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goto failure;
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}
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}
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caps = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
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if (!caps) {
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goto failure;
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}
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engine_name = S_COR(ast_test_flag(&opts, OPT_RTP_ENGINE),
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opt_args[OPT_ARG_RTP_ENGINE], "asterisk");
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ast_sockaddr_copy(&local_address, &address);
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if (ast_ouraddrfor(&address, &local_address)) {
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ast_log(LOG_ERROR, "Could not get our address for sending media to '%s'\n",
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args.destination);
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goto failure;
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}
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instance = ast_rtp_instance_new(engine_name, NULL, &local_address, NULL);
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if (!instance) {
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ast_log(LOG_ERROR,
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"Could not create %s RTP instance for sending media to '%s'\n",
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S_OR(engine_name, "default"), args.destination);
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goto failure;
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}
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chan = ast_channel_alloc(1, AST_STATE_DOWN, "", "", "", "", "", assignedids,
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requestor, 0, "UnicastRTP/%s-%p", args.destination, instance);
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if (!chan) {
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ast_rtp_instance_destroy(instance);
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goto failure;
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}
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ast_rtp_instance_set_channel_id(instance, ast_channel_uniqueid(chan));
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ast_rtp_instance_set_remote_address(instance, &address);
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ast_channel_set_fd(chan, 0, ast_rtp_instance_fd(instance, 0));
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ast_channel_tech_set(chan, &unicast_rtp_tech);
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ast_format_cap_append(caps, fmt, 0);
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ast_channel_nativeformats_set(chan, caps);
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ast_channel_set_writeformat(chan, fmt);
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ast_channel_set_rawwriteformat(chan, fmt);
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ast_channel_set_readformat(chan, fmt);
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ast_channel_set_rawreadformat(chan, fmt);
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ast_channel_tech_pvt_set(chan, instance);
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pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_ADDRESS",
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ast_sockaddr_stringify_addr(&local_address));
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ast_rtp_instance_get_local_address(instance, &local_address);
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pbx_builtin_setvar_helper(chan, "UNICASTRTP_LOCAL_PORT",
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ast_sockaddr_stringify_port(&local_address));
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ast_channel_unlock(chan);
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ao2_ref(fmt, -1);
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ao2_ref(caps, -1);
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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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*cause = AST_CAUSE_FAILURE;
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return NULL;
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}
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/*! \brief Function called when our module is unloaded */
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static int unload_module(void)
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{
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ast_channel_unregister(&multicast_rtp_tech);
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ao2_cleanup(multicast_rtp_tech.capabilities);
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multicast_rtp_tech.capabilities = NULL;
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ast_channel_unregister(&unicast_rtp_tech);
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ao2_cleanup(unicast_rtp_tech.capabilities);
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unicast_rtp_tech.capabilities = NULL;
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return 0;
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}
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/*! \brief Function called when our module is loaded */
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static int load_module(void)
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{
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if (!(multicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
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return AST_MODULE_LOAD_DECLINE;
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}
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ast_format_cap_append_by_type(multicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
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if (ast_channel_register(&multicast_rtp_tech)) {
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ast_log(LOG_ERROR, "Unable to register channel class 'MulticastRTP'\n");
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unload_module();
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return AST_MODULE_LOAD_DECLINE;
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}
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if (!(unicast_rtp_tech.capabilities = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT))) {
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unload_module();
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return AST_MODULE_LOAD_DECLINE;
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}
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ast_format_cap_append_by_type(unicast_rtp_tech.capabilities, AST_MEDIA_TYPE_UNKNOWN);
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if (ast_channel_register(&unicast_rtp_tech)) {
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ast_log(LOG_ERROR, "Unable to register channel class 'UnicastRTP'\n");
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unload_module();
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return AST_MODULE_LOAD_DECLINE;
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}
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return AST_MODULE_LOAD_SUCCESS;
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}
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AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "RTP Media Channel",
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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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.load_pri = AST_MODPRI_CHANNEL_DRIVER,
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);
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