chan_sip: Add sendrpid trust options

[asterisk/asterisk.git] / channels / chan_sip.c

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2012, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*!
 * \file
 * \brief Implementation of Session Initiation Protocol
 *
 * \author Mark Spencer <markster@digium.com>
 *
 * See Also:
 * \arg \ref AstCREDITS
 *
 * Implementation of RFC 3261 - without S/MIME, and experimental TCP and TLS support
 * Configuration file \link Config_sip sip.conf \endlink
 *
 * ********** IMPORTANT *
 * \note TCP/TLS support is EXPERIMENTAL and WILL CHANGE. This applies to configuration
 *      settings, dialplan commands and dialplans apps/functions
 * See \ref sip_tcp_tls
 *
 *
 * ******** General TODO:s
 * \todo Better support of forking
 * \todo VIA branch tag transaction checking
 * \todo Transaction support
 *
 * ******** Wishlist: Improvements
 * - Support of SIP domains for devices, so that we match on username\@domain in the From: header
 * - Connect registrations with a specific device on the incoming call. It's not done
 *   automatically in Asterisk
 *
 * \ingroup channel_drivers
 *
 * \par Overview of the handling of SIP sessions
 * The SIP channel handles several types of SIP sessions, or dialogs,
 * not all of them being "telephone calls".
 * - Incoming calls that will be sent to the PBX core
 * - Outgoing calls, generated by the PBX
 * - SIP subscriptions and notifications of states and voicemail messages
 * - SIP registrations, both inbound and outbound
 * - SIP peer management (peerpoke, OPTIONS)
 * - SIP text messages
 *
 * In the SIP channel, there's a list of active SIP dialogs, which includes
 * all of these when they are active. "sip show channels" in the CLI will
 * show most of these, excluding subscriptions which are shown by
 * "sip show subscriptions"
 *
 * \par incoming packets
 * Incoming packets are received in the monitoring thread, then handled by
 * sipsock_read() for udp only. In tcp, packets are read by the tcp_helper thread.
 * sipsock_read() function parses the packet and matches an existing
 * dialog or starts a new SIP dialog.
 *
 * sipsock_read sends the packet to handle_incoming(), that parses a bit more.
 * If it is a response to an outbound request, the packet is sent to handle_response().
 * If it is a request, handle_incoming() sends it to one of a list of functions
 * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
 * sipsock_read locks the ast_channel if it exists (an active call) and
 * unlocks it after we have processed the SIP message.
 *
 * A new INVITE is sent to handle_request_invite(), that will end up
 * starting a new channel in the PBX, the new channel after that executing
 * in a separate channel thread. This is an incoming "call".
 * When the call is answered, either by a bridged channel or the PBX itself
 * the sip_answer() function is called.
 *
 * The actual media - Video or Audio - is mostly handled by the RTP subsystem
 * in rtp.c
 *
 * \par Outbound calls
 * Outbound calls are set up by the PBX through the sip_request_call()
 * function. After that, they are activated by sip_call().
 *
 * \par Hanging up
 * The PBX issues a hangup on both incoming and outgoing calls through
 * the sip_hangup() function
 */

/*! \li \ref chan_sip.c uses configuration files \ref sip.conf and \ref sip_notify.conf
 * \addtogroup configuration_file
 */

/*! \page sip.conf sip.conf
 * \verbinclude sip.conf.sample
 */

/*! \page sip_notify.conf sip_notify.conf
 * \verbinclude sip_notify.conf.sample
 */

/*!
 * \page sip_tcp_tls SIP TCP and TLS support
 *
 * \par tcpfixes TCP implementation changes needed
 * \todo Fix TCP/TLS handling in dialplan, SRV records, transfers and much more
 * \todo Save TCP/TLS sessions in registry
 *      If someone registers a SIPS uri, this forces us to set up a TLS connection back.
 * \todo Add TCP/TLS information to function SIPPEER and SIPCHANINFO
 * \todo If tcpenable=yes, we must open a TCP socket on the same address as the IP for UDP.
 *      The tcpbindaddr config option should only be used to open ADDITIONAL ports
 *      So we should propably go back to
 *              bindaddr= the default address to bind to. If tcpenable=yes, then bind this to both udp and TCP
 *                              if tlsenable=yes, open TLS port (provided we also have cert)
 *              tcpbindaddr = extra address for additional TCP connections
 *              tlsbindaddr = extra address for additional TCP/TLS connections
 *              udpbindaddr = extra address for additional UDP connections
 *                      These three options should take multiple IP/port pairs
 *      Note: Since opening additional listen sockets is a *new* feature we do not have today
 *              the XXXbindaddr options needs to be disabled until we have support for it
 *
 * \todo re-evaluate the transport= setting in sip.conf. This is right now not well
 *      thought of. If a device in sip.conf contacts us via TCP, we should not switch transport,
 *      even if udp is the configured first transport.
 *
 * \todo Be prepared for one outbound and another incoming socket per pvt. This applies
 *       specially to communication with other peers (proxies).
 * \todo We need to test TCP sessions with SIP proxies and in regards
 *       to the SIP outbound specs.
 * \todo ;transport=tls was deprecated in RFC3261 and should not be used at all. See section 26.2.2.
 *
 * \todo If the message is smaller than the given Content-length, the request should get a 400 Bad request
 *       message. If it's a response, it should be dropped. (RFC 3261, Section 18.3)
 * \todo Since we have had multidomain support in Asterisk for quite a while, we need to support
 *       multiple domains in our TLS implementation, meaning one socket and one cert per domain
 * \todo Selection of transport for a request needs to be done after we've parsed all route headers,
 *       also considering outbound proxy options.
 *              First request: Outboundproxy, routes, (reg contact or URI. If URI doesn't have port:  DNS naptr, srv, AAA)
 *              Intermediate requests: Outboundproxy(only when forced), routes, contact/uri
 *      DNS naptr support is crucial. A SIP uri might lead to a TLS connection.
 *      Also note that due to outbound proxy settings, a SIPS uri might have to be sent on UDP (not to recommend though)
 * \todo Default transports are set to UDP, which cause the wrong behaviour when contacting remote
 *      devices directly from the dialplan. UDP is only a fallback if no other method works,
 *      in order to be compatible with RFC2543 (SIP/1.0) devices. For transactions that exceed the
 *      MTU (like INIVTE with video, audio and RTT)  TCP should be preferred.
 *
 *      When dialling unconfigured peers (with no port number)  or devices in external domains
 *      NAPTR records MUST be consulted to find configured transport. If they are not found,
 *      SRV records for both TCP and UDP should be checked. If there's a record for TCP, use that.
 *      If there's no record for TCP, then use UDP as a last resort. If there's no SRV records,
 *      \note this only applies if there's no outbound proxy configured for the session. If an outbound
 *      proxy is configured, these procedures might apply for locating the proxy and determining
 *      the transport to use for communication with the proxy.
 * \par Other bugs to fix ----
 * __set_address_from_contact(const char *fullcontact, struct sockaddr_in *sin, int tcp)
 *      - sets TLS port as default for all TCP connections, unless other port is given in contact.
 * parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
 *      - assumes that the contact the UA registers is using the same transport as the REGISTER request, which is
 *        a bad guess.
 *      - Does not save any information about TCP/TLS connected devices, which is a severe BUG, as discussed on the mailing list.
 * get_destination(struct sip_pvt *p, struct sip_request *oreq)
 *      - Doesn't store the information that we got an incoming SIPS request in the channel, so that
 *        we can require a secure signalling path OUT of Asterisk (on SIP or IAX2). Possibly, the call should
 *        fail on in-secure signalling paths if there's no override in our configuration. At least, provide a
 *        channel variable in the dialplan.
 * get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req)
 *      - As above, if we have a SIPS: uri in the refer-to header
 *      - Does not check transport in refer_to uri.
 */

/*** MODULEINFO
        <use type="module">res_crypto</use>
        <use type="module">res_http_websocket</use>
        <support_level>core</support_level>
 ***/

/*!  \page sip_session_timers SIP Session Timers in Asterisk Chan_sip

        The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
        refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
        request at a negotiated interval. If a session refresh fails then all the entities that support Session-
        Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
        the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
        that do not support Session-Timers).

        The Session-Timers can be configured on a system-wide, per-user, or per-peer basis. The peruser/
        per-peer settings override the global settings. The following new parameters have been
        added to the sip.conf file.
                session-timers=["accept", "originate", "refuse"]
                session-expires=[integer]
                session-minse=[integer]
                session-refresher=["uas", "uac"]

        The session-timers parameter in sip.conf defines the mode of operation of SIP session-timers feature in
        Asterisk. The Asterisk can be configured in one of the following three modes:

        1. Accept :: In the "accept" mode, the Asterisk server honors session-timers requests
                made by remote end-points. A remote end-point can request Asterisk to engage
                session-timers by either sending it an INVITE request with a "Supported: timer"
                header in it or by responding to Asterisk's INVITE with a 200 OK that contains
                Session-Expires: header in it. In this mode, the Asterisk server does not
                request session-timers from remote end-points. This is the default mode.
        2. Originate :: In the "originate" mode, the Asterisk server requests the remote
                end-points to activate session-timers in addition to honoring such requests
                made by the remote end-pints. In order to get as much protection as possible
                against hanging SIP channels due to network or end-point failures, Asterisk
                resends periodic re-INVITEs even if a remote end-point does not support
                the session-timers feature.
        3. Refuse :: In the "refuse" mode, Asterisk acts as if it does not support session-
                timers for inbound or outbound requests. If a remote end-point requests
                session-timers in a dialog, then Asterisk ignores that request unless it's
                noted as a requirement (Require: header), in which case the INVITE is
                rejected with a 420 Bad Extension response.

*/

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include <signal.h>
#include <sys/signal.h>
#include <regex.h>
#include <inttypes.h>

#include "asterisk/network.h"
#include "asterisk/paths.h"     /* need ast_config_AST_SYSTEM_NAME */
#include "asterisk/lock.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/pickup.h"
#include "asterisk/parking.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj2.h"
#include "asterisk/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/monitor.h"
#include "asterisk/netsock2.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"
#include "asterisk/ast_version.h"
#include "asterisk/data.h"
#include "asterisk/aoc.h"
#include "asterisk/message.h"
#include "sip/include/sip.h"
#include "sip/include/globals.h"
#include "sip/include/config_parser.h"
#include "sip/include/reqresp_parser.h"
#include "sip/include/sip_utils.h"
#include "asterisk/sdp_srtp.h"
#include "asterisk/ccss.h"
#include "asterisk/xml.h"
#include "sip/include/dialog.h"
#include "sip/include/dialplan_functions.h"
#include "sip/include/security_events.h"
#include "sip/include/route.h"
#include "asterisk/sip_api.h"
#include "asterisk/app.h"
#include "asterisk/bridge.h"
#include "asterisk/stasis.h"
#include "asterisk/stasis_endpoints.h"
#include "asterisk/stasis_system.h"
#include "asterisk/stasis_channels.h"
#include "asterisk/features_config.h"
#include "asterisk/http_websocket.h"

/*** DOCUMENTATION
        <application name="SIPDtmfMode" language="en_US">
                <synopsis>
                        Change the dtmfmode for a SIP call.
                </synopsis>
                <syntax>
                        <parameter name="mode" required="true">
                                <enumlist>
                                        <enum name="inband" />
                                        <enum name="info" />
                                        <enum name="rfc2833" />
                                </enumlist>
                        </parameter>
                </syntax>
                <description>
                        <para>Changes the dtmfmode for a SIP call.</para>
                </description>
        </application>
        <application name="SIPAddHeader" language="en_US">
                <synopsis>
                        Add a SIP header to the outbound call.
                </synopsis>
                <syntax argsep=":">
                        <parameter name="Header" required="true" />
                        <parameter name="Content" required="true" />
                </syntax>
                <description>
                        <para>Adds a header to a SIP call placed with DIAL.</para>
                        <para>Remember to use the X-header if you are adding non-standard SIP
                        headers, like <literal>X-Asterisk-Accountcode:</literal>. Use this with care.
                        Adding the wrong headers may jeopardize the SIP dialog.</para>
                        <para>Always returns <literal>0</literal>.</para>
                </description>
        </application>
        <application name="SIPRemoveHeader" language="en_US">
                <synopsis>
                        Remove SIP headers previously added with SIPAddHeader
                </synopsis>
                <syntax>
                        <parameter name="Header" required="false" />
                </syntax>
                <description>
                        <para>SIPRemoveHeader() allows you to remove headers which were previously
                        added with SIPAddHeader(). If no parameter is supplied, all previously added
                        headers will be removed. If a parameter is supplied, only the matching headers
                        will be removed.</para>
                        <para>For example you have added these 2 headers:</para>
                        <para>SIPAddHeader(P-Asserted-Identity: sip:foo@bar);</para>
                        <para>SIPAddHeader(P-Preferred-Identity: sip:bar@foo);</para>
                        <para></para>
                        <para>// remove all headers</para>
                        <para>SIPRemoveHeader();</para>
                        <para>// remove all P- headers</para>
                        <para>SIPRemoveHeader(P-);</para>
                        <para>// remove only the PAI header (note the : at the end)</para>
                        <para>SIPRemoveHeader(P-Asserted-Identity:);</para>
                        <para></para>
                        <para>Always returns <literal>0</literal>.</para>
                </description>
        </application>
        <application name="SIPSendCustomINFO" language="en_US">
                <synopsis>
                        Send a custom INFO frame on specified channels.
                </synopsis>
                <syntax>
                        <parameter name="Data" required="true" />
                        <parameter name="UserAgent" required="false" />
                </syntax>
                <description>
                        <para>SIPSendCustomINFO() allows you to send a custom INFO message on all
                        active SIP channels or on channels with the specified User Agent. This
                        application is only available if TEST_FRAMEWORK is defined.</para>
                </description>
        </application>
        <function name="SIP_HEADER" language="en_US">
                <synopsis>
                        Gets the specified SIP header from an incoming INVITE message.
                </synopsis>
                <syntax>
                        <parameter name="name" required="true" />
                        <parameter name="number">
                                <para>If not specified, defaults to <literal>1</literal>.</para>
                        </parameter>
                </syntax>
                <description>
                        <para>Since there are several headers (such as Via) which can occur multiple
                        times, SIP_HEADER takes an optional second argument to specify which header with
                        that name to retrieve. Headers start at offset <literal>1</literal>.</para>
                        <para>Please observe that contents of the SDP (an attachment to the
                        SIP request) can't be accessed with this function.</para>
                </description>
        </function>
        <function name="SIPPEER" language="en_US">
                <synopsis>
                        Gets SIP peer information.
                </synopsis>
                <syntax>
                        <parameter name="peername" required="true" />
                        <parameter name="item">
                                <enumlist>
                                        <enum name="ip">
                                                <para>(default) The IP address.</para>
                                        </enum>
                                        <enum name="port">
                                                <para>The port number.</para>
                                        </enum>
                                        <enum name="mailbox">
                                                <para>The configured mailbox.</para>
                                        </enum>
                                        <enum name="context">
                                                <para>The configured context.</para>
                                        </enum>
                                        <enum name="expire">
                                                <para>The epoch time of the next expire.</para>
                                        </enum>
                                        <enum name="dynamic">
                                                <para>Is it dynamic? (yes/no).</para>
                                        </enum>
                                        <enum name="callerid_name">
                                                <para>The configured Caller ID name.</para>
                                        </enum>
                                        <enum name="callerid_num">
                                                <para>The configured Caller ID number.</para>
                                        </enum>
                                        <enum name="callgroup">
                                                <para>The configured Callgroup.</para>
                                        </enum>
                                        <enum name="pickupgroup">
                                                <para>The configured Pickupgroup.</para>
                                        </enum>
                                        <enum name="namedcallgroup">
                                                <para>The configured Named Callgroup.</para>
                                        </enum>
                                        <enum name="namedpickupgroup">
                                                <para>The configured Named Pickupgroup.</para>
                                        </enum>
                                        <enum name="codecs">
                                                <para>The configured codecs.</para>
                                        </enum>
                                        <enum name="status">
                                                <para>Status (if qualify=yes).</para>
                                        </enum>
                                        <enum name="regexten">
                                                <para>Extension activated at registration.</para>
                                        </enum>
                                        <enum name="limit">
                                                <para>Call limit (call-limit).</para>
                                        </enum>
                                        <enum name="busylevel">
                                                <para>Configured call level for signalling busy.</para>
                                        </enum>
                                        <enum name="curcalls">
                                                <para>Current amount of calls. Only available if call-limit is set.</para>
                                        </enum>
                                        <enum name="language">
                                                <para>Default language for peer.</para>
                                        </enum>
                                        <enum name="accountcode">
                                                <para>Account code for this peer.</para>
                                        </enum>
                                        <enum name="useragent">
                                                <para>Current user agent header used by peer.</para>
                                        </enum>
                                        <enum name="maxforwards">
                                                <para>The value used for SIP loop prevention in outbound requests</para>
                                        </enum>
                                        <enum name="chanvar[name]">
                                                <para>A channel variable configured with setvar for this peer.</para>
                                        </enum>
                                        <enum name="codec[x]">
                                                <para>Preferred codec index number <replaceable>x</replaceable> (beginning with zero).</para>
                                        </enum>
                                </enumlist>
                        </parameter>
                </syntax>
                <description></description>
        </function>
        <function name="SIPCHANINFO" language="en_US">
                <synopsis>
                        Gets the specified SIP parameter from the current channel.
                </synopsis>
                <syntax>
                        <parameter name="item" required="true">
                                <enumlist>
                                        <enum name="peerip">
                                                <para>The IP address of the peer.</para>
                                        </enum>
                                        <enum name="recvip">
                                                <para>The source IP address of the peer.</para>
                                        </enum>
                                        <enum name="from">
                                                <para>The SIP URI from the <literal>From:</literal> header.</para>
                                        </enum>
                                        <enum name="uri">
                                                <para>The SIP URI from the <literal>Contact:</literal> header.</para>
                                        </enum>
                                        <enum name="useragent">
                                                <para>The Useragent header used by the peer.</para>
                                        </enum>
                                        <enum name="peername">
                                                <para>The name of the peer.</para>
                                        </enum>
                                        <enum name="t38passthrough">
                                                <para><literal>1</literal> if T38 is offered or enabled in this channel,
                                                otherwise <literal>0</literal>.</para>
                                        </enum>
                                </enumlist>
                        </parameter>
                </syntax>
                <description></description>
        </function>
        <function name="CHECKSIPDOMAIN" language="en_US">
                <synopsis>
                        Checks if domain is a local domain.
                </synopsis>
                <syntax>
                        <parameter name="domain" required="true" />
                </syntax>
                <description>
                        <para>This function checks if the <replaceable>domain</replaceable> in the argument is configured
                        as a local SIP domain that this Asterisk server is configured to handle.
                        Returns the domain name if it is locally handled, otherwise an empty string.
                        Check the <literal>domain=</literal> configuration in <filename>sip.conf</filename>.</para>
                </description>
        </function>
        <manager name="SIPpeers" language="en_US">
                <synopsis>
                        List SIP peers (text format).
                </synopsis>
                <syntax>
                        <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
                </syntax>
                <description>
                        <para>Lists SIP peers in text format with details on current status.
                        <literal>Peerlist</literal> will follow as separate events, followed by a final event called
                        <literal>PeerlistComplete</literal>.</para>
                </description>
        </manager>
        <manager name="SIPshowpeer" language="en_US">
                <synopsis>
                        show SIP peer (text format).
                </synopsis>
                <syntax>
                        <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
                        <parameter name="Peer" required="true">
                                <para>The peer name you want to check.</para>
                        </parameter>
                </syntax>
                <description>
                        <para>Show one SIP peer with details on current status.</para>
                </description>
        </manager>
        <manager name="SIPqualifypeer" language="en_US">
                <synopsis>
                        Qualify SIP peers.
                </synopsis>
                <syntax>
                        <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
                        <parameter name="Peer" required="true">
                                <para>The peer name you want to qualify.</para>
                        </parameter>
                </syntax>
                <description>
                        <para>Qualify a SIP peer.</para>
                </description>
                <see-also>
                        <ref type="managerEvent">SIPQualifyPeerDone</ref>
                </see-also>
        </manager>
        <manager name="SIPshowregistry" language="en_US">
                <synopsis>
                        Show SIP registrations (text format).
                </synopsis>
                <syntax>
                        <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
                </syntax>
                <description>
                        <para>Lists all registration requests and status. Registrations will follow as separate
                        events followed by a final event called <literal>RegistrationsComplete</literal>.</para>
                </description>
        </manager>
        <manager name="SIPnotify" language="en_US">
                <synopsis>
                        Send a SIP notify.
                </synopsis>
                <syntax>
                        <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
                        <parameter name="Channel" required="true">
                                <para>Peer to receive the notify.</para>
                        </parameter>
                        <parameter name="Variable" required="true">
                                <para>At least one variable pair must be specified.
                                <replaceable>name</replaceable>=<replaceable>value</replaceable></para>
                        </parameter>
                </syntax>
                <description>
                        <para>Sends a SIP Notify event.</para>
                        <para>All parameters for this event must be specified in the body of this request
                        via multiple <literal>Variable: name=value</literal> sequences.</para>
                </description>
        </manager>
        <manager name="SIPpeerstatus" language="en_US">
                <synopsis>
                        Show the status of one or all of the sip peers.
                </synopsis>
                <syntax>
                        <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
                        <parameter name="Peer" required="false">
                                <para>The peer name you want to check.</para>
                        </parameter>
                </syntax>
                <description>
                        <para>Retrieves the status of one or all of the sip peers.  If no peer name is specified, status
                        for all of the sip peers will be retrieved.</para>
                </description>
        </manager>
        <info name="SIPMessageFromInfo" language="en_US" tech="SIP">
                <para>The <literal>from</literal> parameter can be a configured peer name
                or in the form of "display-name" &lt;URI&gt;.</para>
        </info>
        <info name="SIPMessageToInfo" language="en_US" tech="SIP">
                <para>Specifying a prefix of <literal>sip:</literal> will send the
                message as a SIP MESSAGE request.</para>
        </info>
        <managerEvent language="en_US" name="SIPQualifyPeerDone">
                <managerEventInstance class="EVENT_FLAG_CALL">
                        <synopsis>Raised when SIPQualifyPeer has finished qualifying the specified peer.</synopsis>
                        <syntax>
                                <parameter name="Peer">
                                        <para>The name of the peer.</para>
                                </parameter>
                                <parameter name="ActionID">
                                        <para>This is only included if an ActionID Header was sent with the action request, in which case it will be that ActionID.</para>
                                </parameter>
                        </syntax>
                        <see-also>
                                <ref type="manager">SIPqualifypeer</ref>
                        </see-also>
                </managerEventInstance>
        </managerEvent>
        <managerEvent language="en_US" name="SessionTimeout">
                <managerEventInstance class="EVENT_FLAG_CALL">
                        <synopsis>Raised when a SIP session times out.</synopsis>
                        <syntax>
                                <channel_snapshot/>
                                <parameter name="Source">
                                        <para>The source of the session timeout.</para>
                                        <enumlist>
                                                <enum name="RTPTimeout" />
                                                <enum name="SIPSessionTimer" />
                                        </enumlist>
                                </parameter>
                        </syntax>
                </managerEventInstance>
        </managerEvent>
 ***/

static int min_expiry = DEFAULT_MIN_EXPIRY;        /*!< Minimum accepted registration time */
static int max_expiry = DEFAULT_MAX_EXPIRY;        /*!< Maximum accepted registration time */
static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
static int min_subexpiry = DEFAULT_MIN_EXPIRY;     /*!< Minimum accepted subscription time */
static int max_subexpiry = DEFAULT_MAX_EXPIRY;     /*!< Maximum accepted subscription time */
static int mwi_expiry = DEFAULT_MWI_EXPIRY;

static int unauth_sessions = 0;
static int authlimit = DEFAULT_AUTHLIMIT;
static int authtimeout = DEFAULT_AUTHTIMEOUT;

/*! \brief Global jitterbuffer configuration - by default, jb is disabled
 *  \note Values shown here match the defaults shown in sip.conf.sample */
static struct ast_jb_conf default_jbconf =
{
        .flags = 0,
        .max_size = 200,
        .resync_threshold = 1000,
        .impl = "fixed",
        .target_extra = 40,
};
static struct ast_jb_conf global_jbconf;                /*!< Global jitterbuffer configuration */

static const char config[] = "sip.conf";                /*!< Main configuration file */
static const char notify_config[] = "sip_notify.conf";  /*!< Configuration file for sending Notify with CLI commands to reconfigure or reboot phones */

/*! \brief Readable descriptions of device states.
 *  \note Should be aligned to above table as index */
static const struct invstate2stringtable {
        const enum invitestates state;
        const char *desc;
} invitestate2string[] = {
        {INV_NONE,              "None"  },
        {INV_CALLING,           "Calling (Trying)"},
        {INV_PROCEEDING,        "Proceeding "},
        {INV_EARLY_MEDIA,       "Early media"},
        {INV_COMPLETED,         "Completed (done)"},
        {INV_CONFIRMED,         "Confirmed (up)"},
        {INV_TERMINATED,        "Done"},
        {INV_CANCELLED,         "Cancelled"}
};

/*! \brief Subscription types that we support. We support
 * - dialoginfo updates (really device status, not dialog info as was the original intent of the standard)
 * - SIMPLE presence used for device status
 * - Voicemail notification subscriptions
 */
static const struct cfsubscription_types {
        enum subscriptiontype type;
        const char * const event;
        const char * const mediatype;
        const char * const text;
} subscription_types[] = {
        { NONE,            "-",        "unknown",                    "unknown" },
        /* RFC 4235: SIP Dialog event package */
        { DIALOG_INFO_XML, "dialog",   "application/dialog-info+xml", "dialog-info+xml" },
        { CPIM_PIDF_XML,   "presence", "application/cpim-pidf+xml",   "cpim-pidf+xml" },  /* RFC 3863 */
        { PIDF_XML,        "presence", "application/pidf+xml",        "pidf+xml" },       /* RFC 3863 */
        { XPIDF_XML,       "presence", "application/xpidf+xml",       "xpidf+xml" },       /* Pre-RFC 3863 with MS additions */
        { MWI_NOTIFICATION,     "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
};

/*! \brief The core structure to setup dialogs. We parse incoming messages by using
 *  structure and then route the messages according to the type.
 *
 *  \note Note that sip_methods[i].id == i must hold or the code breaks
 */
static const struct  cfsip_methods {
        enum sipmethod id;
        int need_rtp;           /*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
        char * const text;
        enum can_create_dialog can_create;
} sip_methods[] = {
        { SIP_UNKNOWN,   RTP,    "-UNKNOWN-",CAN_CREATE_DIALOG },
        { SIP_RESPONSE,  NO_RTP, "SIP/2.0",  CAN_NOT_CREATE_DIALOG },
        { SIP_REGISTER,  NO_RTP, "REGISTER", CAN_CREATE_DIALOG },
        { SIP_OPTIONS,   NO_RTP, "OPTIONS",  CAN_CREATE_DIALOG },
        { SIP_NOTIFY,    NO_RTP, "NOTIFY",   CAN_CREATE_DIALOG },
        { SIP_INVITE,    RTP,    "INVITE",   CAN_CREATE_DIALOG },
        { SIP_ACK,       NO_RTP, "ACK",      CAN_NOT_CREATE_DIALOG },
        { SIP_PRACK,     NO_RTP, "PRACK",    CAN_NOT_CREATE_DIALOG },
        { SIP_BYE,       NO_RTP, "BYE",      CAN_NOT_CREATE_DIALOG },
        { SIP_REFER,     NO_RTP, "REFER",    CAN_CREATE_DIALOG },
        { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE",CAN_CREATE_DIALOG },
        { SIP_MESSAGE,   NO_RTP, "MESSAGE",  CAN_CREATE_DIALOG },
        { SIP_UPDATE,    NO_RTP, "UPDATE",   CAN_NOT_CREATE_DIALOG },
        { SIP_INFO,      NO_RTP, "INFO",     CAN_NOT_CREATE_DIALOG },
        { SIP_CANCEL,    NO_RTP, "CANCEL",   CAN_NOT_CREATE_DIALOG },
        { SIP_PUBLISH,   NO_RTP, "PUBLISH",  CAN_CREATE_DIALOG },
        { SIP_PING,      NO_RTP, "PING",     CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
};

/*! \brief Diversion header reasons
 *
 * The core defines a bunch of constants used to define
 * redirecting reasons. This provides a translation table
 * between those and the strings which may be present in
 * a SIP Diversion header
 */
static const struct sip_reasons {
        enum AST_REDIRECTING_REASON code;
        char * const text;
} sip_reason_table[] = {
        { AST_REDIRECTING_REASON_UNKNOWN, "unknown" },
        { AST_REDIRECTING_REASON_USER_BUSY, "user-busy" },
        { AST_REDIRECTING_REASON_NO_ANSWER, "no-answer" },
        { AST_REDIRECTING_REASON_UNAVAILABLE, "unavailable" },
        { AST_REDIRECTING_REASON_UNCONDITIONAL, "unconditional" },
        { AST_REDIRECTING_REASON_TIME_OF_DAY, "time-of-day" },
        { AST_REDIRECTING_REASON_DO_NOT_DISTURB, "do-not-disturb" },
        { AST_REDIRECTING_REASON_DEFLECTION, "deflection" },
        { AST_REDIRECTING_REASON_FOLLOW_ME, "follow-me" },
        { AST_REDIRECTING_REASON_OUT_OF_ORDER, "out-of-service" },
        { AST_REDIRECTING_REASON_AWAY, "away" },
        { AST_REDIRECTING_REASON_CALL_FWD_DTE, "unknown"},
        { AST_REDIRECTING_REASON_SEND_TO_VM, "send_to_vm"},
};


/*! \name DefaultSettings
        Default setttings are used as a channel setting and as a default when
        configuring devices
*/
static char default_language[MAX_LANGUAGE];      /*!< Default language setting for new channels */
static char default_callerid[AST_MAX_EXTENSION]; /*!< Default caller ID for sip messages */
static char default_mwi_from[80];                /*!< Default caller ID for MWI updates */
static char default_fromdomain[AST_MAX_EXTENSION]; /*!< Default domain on outound messages */
static int default_fromdomainport;                 /*!< Default domain port on outbound messages */
static char default_notifymime[AST_MAX_EXTENSION]; /*!< Default MIME media type for MWI notify messages */
static char default_vmexten[AST_MAX_EXTENSION];    /*!< Default From Username on MWI updates */
static int default_qualify;                        /*!< Default Qualify= setting */
static int default_keepalive;                      /*!< Default keepalive= setting */
static char default_mohinterpret[MAX_MUSICCLASS];  /*!< Global setting for moh class to use when put on hold */
static char default_mohsuggest[MAX_MUSICCLASS];    /*!< Global setting for moh class to suggest when putting
                                                    *   a bridged channel on hold */
static char default_parkinglot[AST_MAX_CONTEXT];   /*!< Parkinglot */
static char default_engine[256];                   /*!< Default RTP engine */
static int default_maxcallbitrate;                 /*!< Maximum bitrate for call */
static struct ast_codec_pref default_prefs;        /*!< Default codec prefs */
static char default_zone[MAX_TONEZONE_COUNTRY];        /*!< Default tone zone for channels created from the SIP driver */
static unsigned int default_transports;            /*!< Default Transports (enum ast_transport) that are acceptable */
static unsigned int default_primary_transport;     /*!< Default primary Transport (enum ast_transport) for outbound connections to devices */

static struct sip_settings sip_cfg;             /*!< SIP configuration data.
                                        \note in the future we could have multiple of these (per domain, per device group etc) */

/*!< use this macro when ast_uri_decode is dependent on pedantic checking to be on. */
#define SIP_PEDANTIC_DECODE(str)        \
        if (sip_cfg.pedanticsipchecking && !ast_strlen_zero(str)) {     \
                ast_uri_decode(str, ast_uri_sip_user);  \
        }       \

static unsigned int chan_idx;       /*!< used in naming sip channel */
static int global_match_auth_username;    /*!< Match auth username if available instead of From: Default off. */

static int global_relaxdtmf;        /*!< Relax DTMF */
static int global_prematuremediafilter;   /*!< Enable/disable premature frames in a call (causing 183 early media) */
static int global_rtptimeout;       /*!< Time out call if no RTP */
static int global_rtpholdtimeout;   /*!< Time out call if no RTP during hold */
static int global_rtpkeepalive;     /*!< Send RTP keepalives */
static int global_reg_timeout;      /*!< Global time between attempts for outbound registrations */
static int global_regattempts_max;  /*!< Registration attempts before giving up */
static int global_reg_retry_403;    /*!< Treat 403 responses to registrations as 401 responses */
static int global_shrinkcallerid;   /*!< enable or disable shrinking of caller id  */
static int global_callcounter;      /*!< Enable call counters for all devices. This is currently enabled by setting the peer
                                     *   call-limit to INT_MAX. When we remove the call-limit from the code, we can make it
                                     *   with just a boolean flag in the device structure */
static unsigned int global_tos_sip;      /*!< IP type of service for SIP packets */
static unsigned int global_tos_audio;    /*!< IP type of service for audio RTP packets */
static unsigned int global_tos_video;    /*!< IP type of service for video RTP packets */
static unsigned int global_tos_text;     /*!< IP type of service for text RTP packets */
static unsigned int global_cos_sip;      /*!< 802.1p class of service for SIP packets */
static unsigned int global_cos_audio;    /*!< 802.1p class of service for audio RTP packets */
static unsigned int global_cos_video;    /*!< 802.1p class of service for video RTP packets */
static unsigned int global_cos_text;     /*!< 802.1p class of service for text RTP packets */
static unsigned int recordhistory;       /*!< Record SIP history. Off by default */
static unsigned int dumphistory;         /*!< Dump history to verbose before destroying SIP dialog */
static char global_useragent[AST_MAX_EXTENSION];    /*!< Useragent for the SIP channel */
static char global_sdpsession[AST_MAX_EXTENSION];   /*!< SDP session name for the SIP channel */
static char global_sdpowner[AST_MAX_EXTENSION];     /*!< SDP owner name for the SIP channel */
static int global_authfailureevents;     /*!< Whether we send authentication failure manager events or not. Default no. */
static int global_t1;           /*!< T1 time */
static int global_t1min;        /*!< T1 roundtrip time minimum */
static int global_timer_b;      /*!< Timer B - RFC 3261 Section 17.1.1.2 */
static unsigned int global_autoframing; /*!< Turn autoframing on or off. */
static int global_qualifyfreq;          /*!< Qualify frequency */
static int global_qualify_gap;          /*!< Time between our group of peer pokes */
static int global_qualify_peers;        /*!< Number of peers to poke at a given time */

static enum st_mode global_st_mode;           /*!< Mode of operation for Session-Timers           */
static enum st_refresher_param global_st_refresher; /*!< Session-Timer refresher                        */
static int global_min_se;                     /*!< Lowest threshold for session refresh interval  */
static int global_max_se;                     /*!< Highest threshold for session refresh interval */

static int global_store_sip_cause;    /*!< Whether the MASTER_CHANNEL(HASH(SIP_CAUSE,[chan_name])) var should be set */

static int global_dynamic_exclude_static = 0; /*!< Exclude static peers from contact registrations */
static unsigned char global_refer_addheaders; /*!< Add extra headers to outgoing REFER */
/*@}*/

/*!
 * We use libxml2 in order to parse XML that may appear in the body of a SIP message. Currently,
 * the only usage is for parsing PIDF bodies of incoming PUBLISH requests in the call-completion
 * event package. This variable is set at module load time and may be checked at runtime to determine
 * if XML parsing support was found.
 */
static int can_parse_xml;

/*! \name Object counters @{
 *
 * \bug These counters are not handled in a thread-safe way ast_atomic_fetchadd_int()
 * should be used to modify these values.
 */
static int speerobjs = 0;     /*!< Static peers */
static int rpeerobjs = 0;     /*!< Realtime peers */
static int apeerobjs = 0;     /*!< Autocreated peer objects */
static int regobjs = 0;       /*!< Registry objects */
/*! @} */

static struct ast_flags global_flags[3] = {{0}};  /*!< global SIP_ flags */
static int global_t38_maxdatagram;                /*!< global T.38 FaxMaxDatagram override */

static struct stasis_subscription *network_change_sub; /*!< subscription id for network change events */
static struct stasis_subscription *acl_change_sub; /*!< subscription id for named ACL system change events */
static int network_change_sched_id = -1;

static char used_context[AST_MAX_CONTEXT];        /*!< name of automatically created context for unloading */

AST_MUTEX_DEFINE_STATIC(netlock);

/*! \brief Protect the monitoring thread, so only one process can kill or start it, and not
   when it's doing something critical. */
AST_MUTEX_DEFINE_STATIC(monlock);

AST_MUTEX_DEFINE_STATIC(sip_reload_lock);

/*! \brief This is the thread for the monitor which checks for input on the channels
   which are not currently in use.  */
static pthread_t monitor_thread = AST_PTHREADT_NULL;

static int sip_reloading = FALSE;                       /*!< Flag for avoiding multiple reloads at the same time */
static enum channelreloadreason sip_reloadreason;       /*!< Reason for last reload/load of configuration */

struct ast_sched_context *sched;     /*!< The scheduling context */
static struct io_context *io;           /*!< The IO context */
static int *sipsock_read_id;            /*!< ID of IO entry for sipsock FD */
struct sip_pkt;
static AST_LIST_HEAD_STATIC(domain_list, domain);    /*!< The SIP domain list */

AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */

static enum sip_debug_e sipdebug;

/*! \brief extra debugging for 'text' related events.
 *  At the moment this is set together with sip_debug_console.
 *  \note It should either go away or be implemented properly.
 */
static int sipdebug_text;

static const struct _map_x_s referstatusstrings[] = {
        { REFER_IDLE,      "<none>" },
        { REFER_SENT,      "Request sent" },
        { REFER_RECEIVED,  "Request received" },
        { REFER_CONFIRMED, "Confirmed" },
        { REFER_ACCEPTED,  "Accepted" },
        { REFER_RINGING,   "Target ringing" },
        { REFER_200OK,     "Done" },
        { REFER_FAILED,    "Failed" },
        { REFER_NOAUTH,    "Failed - auth failure" },
        { -1,               NULL} /* terminator */
};

/* --- Hash tables of various objects --------*/
#ifdef LOW_MEMORY
static const int HASH_PEER_SIZE = 17;
static const int HASH_DIALOG_SIZE = 17;
#else
static const int HASH_PEER_SIZE = 563;  /*!< Size of peer hash table, prime number preferred! */
static const int HASH_DIALOG_SIZE = 563;
#endif

static const struct {
        enum ast_cc_service_type service;
        const char *service_string;
} sip_cc_service_map [] = {
        [AST_CC_NONE] = { AST_CC_NONE, "" },
        [AST_CC_CCBS] = { AST_CC_CCBS, "BS" },
        [AST_CC_CCNR] = { AST_CC_CCNR, "NR" },
        [AST_CC_CCNL] = { AST_CC_CCNL, "NL" },
};

static const struct {
        enum sip_cc_notify_state state;
        const char *state_string;
} sip_cc_notify_state_map [] = {
        [CC_QUEUED] = {CC_QUEUED, "cc-state: queued"},
        [CC_READY] = {CC_READY, "cc-state: ready"},
};

AST_LIST_HEAD_STATIC(epa_static_data_list, epa_backend);


/*!
 * Used to create new entity IDs by ESCs.
 */
static int esc_etag_counter;
static const int DEFAULT_PUBLISH_EXPIRES = 3600;

#ifdef HAVE_LIBXML2
static int cc_esc_publish_handler(struct sip_pvt *pvt, struct sip_request *req, struct event_state_compositor *esc, struct sip_esc_entry *esc_entry);

static const struct sip_esc_publish_callbacks cc_esc_publish_callbacks = {
        .initial_handler = cc_esc_publish_handler,
        .modify_handler = cc_esc_publish_handler,
};
#endif

/*!
 * \brief The Event State Compositors
 *
 * An Event State Compositor is an entity which
 * accepts PUBLISH requests and acts appropriately
 * based on these requests.
 *
 * The actual event_state_compositor structure is simply
 * an ao2_container of sip_esc_entrys. When an incoming
 * PUBLISH is received, we can match the appropriate sip_esc_entry
 * using the entity ID of the incoming PUBLISH.
 */
static struct event_state_compositor {
        enum subscriptiontype event;
        const char * name;
        const struct sip_esc_publish_callbacks *callbacks;
        struct ao2_container *compositor;
} event_state_compositors [] = {
#ifdef HAVE_LIBXML2
        {CALL_COMPLETION, "call-completion", &cc_esc_publish_callbacks},
#endif
};

struct state_notify_data {
        int state;
        struct ao2_container *device_state_info;
        int presence_state;
        const char *presence_subtype;
        const char *presence_message;
};


static const int ESC_MAX_BUCKETS = 37;

/*!
 * \details
 * Here we implement the container for dialogs which are in the
 * dialog_needdestroy state to iterate only through the dialogs
 * unlink them instead of iterate through all dialogs
 */
struct ao2_container *dialogs_needdestroy;

/*!
 * \details
 * Here we implement the container for dialogs which have rtp
 * traffic and rtptimeout, rtpholdtimeout or rtpkeepalive
 * set. We use this container instead the whole dialog list.
 */
struct ao2_container *dialogs_rtpcheck;

/*!
 * \details
 * Here we implement the container for dialogs (sip_pvt), defining
 * generic wrapper functions to ease the transition from the current
 * implementation (a single linked list) to a different container.
 * In addition to a reference to the container, we need functions to lock/unlock
 * the container and individual items, and functions to add/remove
 * references to the individual items.
 */
static struct ao2_container *dialogs;
#define sip_pvt_lock(x) ao2_lock(x)
#define sip_pvt_trylock(x) ao2_trylock(x)
#define sip_pvt_unlock(x) ao2_unlock(x)

/*! \brief  The table of TCP threads */
static struct ao2_container *threadt;

/*! \brief  The peer list: Users, Peers and Friends */
static struct ao2_container *peers;
static struct ao2_container *peers_by_ip;

/*! \brief  A bogus peer, to be used when authentication should fail */
static struct sip_peer *bogus_peer;
/*! \brief  We can recognise the bogus peer by this invalid MD5 hash */
#define BOGUS_PEER_MD5SECRET "intentionally_invalid_md5_string"

/*! \brief  The register list: Other SIP proxies we register with and receive calls from */
static struct ast_register_list {
        ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
        int recheck;
} regl;

/*! \brief  The MWI subscription list */
static struct ast_subscription_mwi_list {
        ASTOBJ_CONTAINER_COMPONENTS(struct sip_subscription_mwi);
} submwil;
static int temp_pvt_init(void *);
static void temp_pvt_cleanup(void *);

/*! \brief A per-thread temporary pvt structure */
AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup);

/*! \brief A per-thread buffer for transport to string conversion */
AST_THREADSTORAGE(sip_transport_str_buf);

/*! \brief Size of the SIP transport buffer */
#define SIP_TRANSPORT_STR_BUFSIZE 128

/*! \brief Authentication container for realm authentication */
static struct sip_auth_container *authl = NULL;
/*! \brief Global authentication container protection while adjusting the references. */
AST_MUTEX_DEFINE_STATIC(authl_lock);

static struct ast_manager_event_blob *session_timeout_to_ami(struct stasis_message *msg);
STASIS_MESSAGE_TYPE_DEFN_LOCAL(session_timeout_type,
        .to_ami = session_timeout_to_ami,
        );

/* --- Sockets and networking --------------*/

/*! \brief Main socket for UDP SIP communication.
 *
 * sipsock is shared between the SIP manager thread (which handles reload
 * requests), the udp io handler (sipsock_read()) and the user routines that
 * issue udp writes (using __sip_xmit()).
 * The socket is -1 only when opening fails (this is a permanent condition),
 * or when we are handling a reload() that changes its address (this is
 * a transient situation during which we might have a harmless race, see
 * below). Because the conditions for the race to be possible are extremely
 * rare, we don't want to pay the cost of locking on every I/O.
 * Rather, we remember that when the race may occur, communication is
 * bound to fail anyways, so we just live with this event and let
 * the protocol handle this above us.
 */
static int sipsock  = -1;

struct ast_sockaddr bindaddr;   /*!< UDP: The address we bind to */

/*! \brief our (internal) default address/port to put in SIP/SDP messages
 *  internip is initialized picking a suitable address from one of the
 * interfaces, and the same port number we bind to. It is used as the
 * default address/port in SIP messages, and as the default address
 * (but not port) in SDP messages.
 */
static struct ast_sockaddr internip;

/*! \brief our external IP address/port for SIP sessions.
 * externaddr.sin_addr is only set when we know we might be behind
 * a NAT, and this is done using a variety of (mutually exclusive)
 * ways from the config file:
 *
 * + with "externaddr = host[:port]" we specify the address/port explicitly.
 *   The address is looked up only once when (re)loading the config file;
 *
 * + with "externhost = host[:port]" we do a similar thing, but the
 *   hostname is stored in externhost, and the hostname->IP mapping
 *   is refreshed every 'externrefresh' seconds;
 *
 * Other variables (externhost, externexpire, externrefresh) are used
 * to support the above functions.
 */
static struct ast_sockaddr externaddr;      /*!< External IP address if we are behind NAT */
static struct ast_sockaddr media_address; /*!< External RTP IP address if we are behind NAT */

static char externhost[MAXHOSTNAMELEN];   /*!< External host name */
static time_t externexpire;             /*!< Expiration counter for re-resolving external host name in dynamic DNS */
static int externrefresh = 10;          /*!< Refresh timer for DNS-based external address (dyndns) */
static uint16_t externtcpport;          /*!< external tcp port */
static uint16_t externtlsport;          /*!< external tls port */

/*! \brief  List of local networks
 * We store "localnet" addresses from the config file into an access list,
 * marked as 'DENY', so the call to ast_apply_ha() will return
 * AST_SENSE_DENY for 'local' addresses, and AST_SENSE_ALLOW for 'non local'
 * (i.e. presumably public) addresses.
 */
static struct ast_ha *localaddr;    /*!< List of local networks, on the same side of NAT as this Asterisk */

static int ourport_tcp;             /*!< The port used for TCP connections */
static int ourport_tls;             /*!< The port used for TCP/TLS connections */
static struct ast_sockaddr debugaddr;

static struct ast_config *notify_types = NULL;    /*!< The list of manual NOTIFY types we know how to send */

/*! some list management macros. */

#define UNLINK(element, head, prev) do {        \
        if (prev)                               \
                (prev)->next = (element)->next; \
        else                                    \
                (head) = (element)->next;       \
        } while (0)

struct ao2_container *sip_monitor_instances;

struct show_peers_context;

/*---------------------------- Forward declarations of functions in chan_sip.c */
/* Note: This is added to help splitting up chan_sip.c into several files
        in coming releases. */

/*--- PBX interface functions */
static struct ast_channel *sip_request_call(const char *type, struct ast_format_cap *cap, const struct ast_assigned_ids *assignedids, const struct ast_channel *requestor, const char *dest, int *cause);
static int sip_devicestate(const char *data);
static int sip_sendtext(struct ast_channel *ast, const char *text);
static int sip_call(struct ast_channel *ast, const char *dest, int timeout);
static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data, int datalen);
static int sip_hangup(struct ast_channel *ast);
static int sip_answer(struct ast_channel *ast);
static struct ast_frame *sip_read(struct ast_channel *ast);
static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
static int sip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
static int sip_transfer(struct ast_channel *ast, const char *dest);
static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static int sip_senddigit_begin(struct ast_channel *ast, char digit);
static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int duration);
static int sip_setoption(struct ast_channel *chan, int option, void *data, int datalen);
static int sip_queryoption(struct ast_channel *chan, int option, void *data, int *datalen);
static const char *sip_get_callid(struct ast_channel *chan);

static int handle_request_do(struct sip_request *req, struct ast_sockaddr *addr);
static int sip_standard_port(enum ast_transport type, int port);
static int sip_prepare_socket(struct sip_pvt *p);
static int get_address_family_filter(unsigned int transport);

/*--- Transmitting responses and requests */
static int sipsock_read(int *id, int fd, short events, void *ignore);
static int __sip_xmit(struct sip_pvt *p, struct ast_str *data);
static int __sip_reliable_xmit(struct sip_pvt *p, uint32_t seqno, int resp, struct ast_str *data, int fatal, int sipmethod);
static void add_cc_call_info_to_response(struct sip_pvt *p, struct sip_request *resp);
static int __transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
static int retrans_pkt(const void *data);
static int transmit_response_using_temp(ast_string_field callid, struct ast_sockaddr *addr, int useglobal_nat, const int intended_method, const struct sip_request *req, const char *msg);
static int transmit_response(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_reliable(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_with_date(struct sip_pvt *p, const char *msg, const struct sip_request *req);
static int transmit_response_with_sdp(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable, int oldsdp, int rpid);
static int transmit_response_with_unsupported(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *unsupported);
static int transmit_response_with_auth(struct sip_pvt *p, const char *msg, const struct sip_request *req, const char *rand, enum xmittype reliable, const char *header, int stale);
static int transmit_provisional_response(struct sip_pvt *p, const char *msg, const struct sip_request *req, int with_sdp);
static int transmit_response_with_allow(struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable);
static void transmit_fake_auth_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable);
static int transmit_request(struct sip_pvt *p, int sipmethod, uint32_t seqno, enum xmittype reliable, int newbranch);
static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, uint32_t seqno, enum xmittype reliable, int newbranch);
static int transmit_publish(struct sip_epa_entry *epa_entry, enum sip_publish_type publish_type, const char * const explicit_uri);
static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init, const char * const explicit_uri);
static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int oldsdp);
static int transmit_info_with_aoc(struct sip_pvt *p, struct ast_aoc_decoded *decoded);
static int transmit_info_with_digit(struct sip_pvt *p, const char digit, unsigned int duration);
static int transmit_info_with_vidupdate(struct sip_pvt *p);
static int transmit_message(struct sip_pvt *p, int init, int auth);
static int transmit_refer(struct sip_pvt *p, const char *dest);
static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs, const char *vmexten);
static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate);
static int transmit_cc_notify(struct ast_cc_agent *agent, struct sip_pvt *subscription, enum sip_cc_notify_state state);
static int transmit_register(struct sip_registry *r, int sipmethod, const char *auth, const char *authheader);
static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno);
static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, uint32_t seqno);
static void copy_request(struct sip_request *dst, const struct sip_request *src);
static void receive_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req, char **name, char **number, int set_call_forward);
static int sip_send_mwi_to_peer(struct sip_peer *peer, int cache_only);

/* Misc dialog routines */
static int __sip_autodestruct(const void *data);
static void *registry_unref(struct sip_registry *reg, char *tag);
static int update_call_counter(struct sip_pvt *fup, int event);
static int auto_congest(const void *arg);
static struct sip_pvt *find_call(struct sip_request *req, struct ast_sockaddr *addr, const int intended_method);
static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards, int resp);
static int build_path(struct sip_pvt *p, struct sip_peer *peer, struct sip_request *req, const char *pathbuf);
static enum check_auth_result register_verify(struct sip_pvt *p, struct ast_sockaddr *addr,
                                              struct sip_request *req, const char *uri);
static int get_sip_pvt_from_replaces(const char *callid, const char *totag, const char *fromtag,
                struct sip_pvt **out_pvt, struct ast_channel **out_chan);
static void check_pendings(struct sip_pvt *p);
static void sip_set_owner(struct sip_pvt *p, struct ast_channel *chan);

static void *sip_pickup_thread(void *stuff);
static int sip_pickup(struct ast_channel *chan);

static int sip_sipredirect(struct sip_pvt *p, const char *dest);
static int is_method_allowed(unsigned int *allowed_methods, enum sipmethod method);

/*--- Codec handling / SDP */
static void try_suggested_sip_codec(struct sip_pvt *p);
static const char *get_sdp_iterate(int* start, struct sip_request *req, const char *name);
static char get_sdp_line(int *start, int stop, struct sip_request *req, const char **value);
static int find_sdp(struct sip_request *req);
static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action);
static int process_sdp_o(const char *o, struct sip_pvt *p);
static int process_sdp_c(const char *c, struct ast_sockaddr *addr);
static int process_sdp_a_sendonly(const char *a, int *sendonly);
static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_instance *instance);
static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec);
static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec);
static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec);
static int process_sdp_a_image(const char *a, struct sip_pvt *p);
static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf);
static void add_dtls_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a_buf);
static void start_ice(struct ast_rtp_instance *instance);
static void add_codec_to_sdp(const struct sip_pvt *p, struct ast_format *codec,
                             struct ast_str **m_buf, struct ast_str **a_buf,
                             int debug, int *min_packet_size, int *max_packet_size);
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
                                struct ast_str **m_buf, struct ast_str **a_buf,
                                int debug);
static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38);
static void do_setnat(struct sip_pvt *p);
static void stop_media_flows(struct sip_pvt *p);

/*--- Authentication stuff */
static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
static enum check_auth_result check_auth(struct sip_pvt *p, struct sip_request *req, const char *username,
                                         const char *secret, const char *md5secret, int sipmethod,
                                         const char *uri, enum xmittype reliable);
static enum check_auth_result check_user_full(struct sip_pvt *p, struct sip_request *req,
                                              int sipmethod, const char *uri, enum xmittype reliable,
                                              struct ast_sockaddr *addr, struct sip_peer **authpeer);
static int check_user(struct sip_pvt *p, struct sip_request *req, int sipmethod, const char *uri, enum xmittype reliable, struct ast_sockaddr *addr);

/*--- Domain handling */
static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
static int add_sip_domain(const char *domain, const enum domain_mode mode, const char *context);
static void clear_sip_domains(void);

/*--- SIP realm authentication */
static void add_realm_authentication(struct sip_auth_container **credentials, const char *configuration, int lineno);
static struct sip_auth *find_realm_authentication(struct sip_auth_container *credentials, const char *realm);

/*--- Misc functions */
static int check_rtp_timeout(struct sip_pvt *dialog, time_t t);
static int reload_config(enum channelreloadreason reason);
static void add_diversion(struct sip_request *req, struct sip_pvt *pvt);
static int expire_register(const void *data);
static void *do_monitor(void *data);
static int restart_monitor(void);
static void peer_mailboxes_to_str(struct ast_str **mailbox_str, struct sip_peer *peer);
static struct ast_variable *copy_vars(struct ast_variable *src);
static int dialog_find_multiple(void *obj, void *arg, int flags);
static struct ast_channel *sip_pvt_lock_full(struct sip_pvt *pvt);
/* static int sip_addrcmp(char *name, struct sockaddr_in *sin); Support for peer matching */
static int sip_refer_alloc(struct sip_pvt *p);
static int sip_notify_alloc(struct sip_pvt *p);
static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);
static void set_peer_nat(const struct sip_pvt *p, struct sip_peer *peer);
static void check_for_nat(const struct ast_sockaddr *them, struct sip_pvt *p);

/*--- Device monitoring and Device/extension state/event handling */
static int extensionstate_update(const char *context, const char *exten, struct state_notify_data *data, struct sip_pvt *p, int force);
static int cb_extensionstate(char *context, char *exten, struct ast_state_cb_info *info, void *data);
static int sip_poke_noanswer(const void *data);
static int sip_poke_peer(struct sip_peer *peer, int force);
static void sip_poke_all_peers(void);
static void sip_peer_hold(struct sip_pvt *p, int hold);
static void mwi_event_cb(void *, struct stasis_subscription *, struct stasis_message *);
static void network_change_stasis_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message);
static void acl_change_stasis_cb(void *data, struct stasis_subscription *sub, struct stasis_message *message);
static void sip_keepalive_all_peers(void);

/*--- Applications, functions, CLI and manager command helpers */
static const char *sip_nat_mode(const struct sip_pvt *p);
static char *sip_show_inuse(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *transfermode2str(enum transfermodes mode) attribute_const;
static int peer_status(struct sip_peer *peer, char *status, int statuslen);
static char *sip_show_sched(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char * _sip_show_peers(int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static struct sip_peer *_sip_show_peers_one(int fd, struct mansession *s, struct show_peers_context *cont, struct sip_peer *peer);
static char *sip_show_peers(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_objects(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static void  print_group(int fd, ast_group_t group, int crlf);
static void  print_named_groups(int fd, struct ast_namedgroups *groups, int crlf);
static const char *dtmfmode2str(int mode) attribute_const;
static int str2dtmfmode(const char *str) attribute_unused;
static const char *insecure2str(int mode) attribute_const;
static const char *allowoverlap2str(int mode) attribute_const;
static void cleanup_stale_contexts(char *new, char *old);
static void print_codec_to_cli(int fd, struct ast_codec_pref *pref);
static const char *domain_mode_to_text(const enum domain_mode mode);
static char *sip_show_domains(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_show_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *_sip_qualify_peer(int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[]);
static char *sip_qualify_peer(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_registry(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_unregister(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_mwi(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static const char *subscription_type2str(enum subscriptiontype subtype) attribute_pure;
static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
static char *complete_sip_peer(const char *word, int state, int flags2);
static char *complete_sip_registered_peer(const char *word, int state, int flags2);
static char *complete_sip_show_history(const char *line, const char *word, int pos, int state);
static char *complete_sip_show_peer(const char *line, const char *word, int pos, int state);
static char *complete_sip_unregister(const char *line, const char *word, int pos, int state);
static char *complete_sip_notify(const char *line, const char *word, int pos, int state);
static char *sip_show_channel(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_channelstats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_show_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_do_debug_ip(int fd, const char *arg);
static char *sip_do_debug_peer(int fd, const char *arg);
static char *sip_do_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_cli_notify(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static char *sip_set_history(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static int sip_dtmfmode(struct ast_channel *chan, const char *data);
static int sip_addheader(struct ast_channel *chan, const char *data);
static int sip_do_reload(enum channelreloadreason reason);
static char *sip_reload(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);
static int ast_sockaddr_resolve_first_af(struct ast_sockaddr *addr,
                                      const char *name, int flag, int family);
static int ast_sockaddr_resolve_first(struct ast_sockaddr *addr,
                                      const char *name, int flag);
static int ast_sockaddr_resolve_first_transport(struct ast_sockaddr *addr,
                                                const char *name, int flag, unsigned int transport);

/*--- Debugging
        Functions for enabling debug per IP or fully, or enabling history logging for
        a SIP dialog
*/
static void sip_dump_history(struct sip_pvt *dialog);   /* Dump history to debuglog at end of dialog, before destroying data */
static inline int sip_debug_test_addr(const struct ast_sockaddr *addr);
static inline int sip_debug_test_pvt(struct sip_pvt *p);
static void append_history_full(struct sip_pvt *p, const char *fmt, ...);
static void sip_dump_history(struct sip_pvt *dialog);

/*--- Device object handling */
static struct sip_peer *build_peer(const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime, int devstate_only);
static int update_call_counter(struct sip_pvt *fup, int event);
static void sip_destroy_peer(struct sip_peer *peer);
static void sip_destroy_peer_fn(void *peer);
static void set_peer_defaults(struct sip_peer *peer);
static struct sip_peer *temp_peer(const char *name);
static void register_peer_exten(struct sip_peer *peer, int onoff);
static int sip_poke_peer_s(const void *data);
static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *p, struct sip_request *req);
static void reg_source_db(struct sip_peer *peer);
static void destroy_association(struct sip_peer *peer);
static void set_insecure_flags(struct ast_flags *flags, const char *value, int lineno);
static int handle_common_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v);
static void set_socket_transport(struct sip_socket *socket, int transport);
static int peer_ipcmp_cb_full(void *obj, void *arg, void *data, int flags);

/* Realtime device support */
static void realtime_update_peer(const char *peername, struct ast_sockaddr *addr, const char *username, const char *fullcontact, const char *useragent, int expirey, unsigned short deprecated_username, int lastms, const char *path);
static void update_peer(struct sip_peer *p, int expire);
static struct ast_variable *get_insecure_variable_from_config(struct ast_config *config);
static const char *get_name_from_variable(const struct ast_variable *var);
static struct sip_peer *realtime_peer(const char *peername, struct ast_sockaddr *sin, char *callbackexten, int devstate_only, int which_objects);
static char *sip_prune_realtime(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a);

/*--- Internal UA client handling (outbound registrations) */
static void ast_sip_ouraddrfor(const struct ast_sockaddr *them, struct ast_sockaddr *us, struct sip_pvt *p);
static void sip_registry_destroy(struct sip_registry *reg);
static int sip_register(const char *value, int lineno);
static const char *regstate2str(enum sipregistrystate regstate) attribute_const;
static int sip_reregister(const void *data);
static int __sip_do_register(struct sip_registry *r);
static int sip_reg_timeout(const void *data);
static void sip_send_all_registers(void);
static int sip_reinvite_retry(const void *data);

/*--- Parsing SIP requests and responses */
static int determine_firstline_parts(struct sip_request *req);
static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
static const char *gettag(const struct sip_request *req, const char *header, char *tagbuf, int tagbufsize);
static int find_sip_method(const char *msg);
static unsigned int parse_allowed_methods(struct sip_request *req);
static unsigned int set_pvt_allowed_methods(struct sip_pvt *pvt, struct sip_request *req);
static int parse_request(struct sip_request *req);
static const char *referstatus2str(enum referstatus rstatus) attribute_pure;
static int method_match(enum sipmethod id, const char *name);
static void parse_copy(struct sip_request *dst, const struct sip_request *src);
static void parse_oli(struct sip_request *req, struct ast_channel *chan);
static const char *find_alias(const char *name, const char *_default);
static const char *__get_header(const struct sip_request *req, const char *name, int *start);
static void lws2sws(struct ast_str *msgbuf);
static void extract_uri(struct sip_pvt *p, struct sip_request *req);
static char *remove_uri_parameters(char *uri);
static int get_refer_info(struct sip_pvt *transferer, struct sip_request *outgoing_req);
static int get_also_info(struct sip_pvt *p, struct sip_request *oreq);
static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req);
static int set_address_from_contact(struct sip_pvt *pvt);
static void check_via(struct sip_pvt *p, const struct sip_request *req);
static int get_rpid(struct sip_pvt *p, struct sip_request *oreq);
static int get_rdnis(struct sip_pvt *p, struct sip_request *oreq, char **name, char **number, int *reason, char **reason_str);
static enum sip_get_dest_result get_destination(struct sip_pvt *p, struct sip_request *oreq, int *cc_recall_core_id);
static int transmit_state_notify(struct sip_pvt *p, struct state_notify_data *data, int full, int timeout);
static void update_connectedline(struct sip_pvt *p, const void *data, size_t datalen);
static void update_redirecting(struct sip_pvt *p, const void *data, size_t datalen);
static int get_domain(const char *str, char *domain, int len);
static void get_realm(struct sip_pvt *p, const struct sip_request *req);
static char *get_content(struct sip_request *req);

/*-- TCP connection handling ---*/
static void *_sip_tcp_helper_thread(struct ast_tcptls_session_instance *tcptls_session);
static void *sip_tcp_worker_fn(void *);

/*--- Constructing requests and responses */
static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
static int init_req(struct sip_request *req, int sipmethod, const char *recip);
static void deinit_req(struct sip_request *req);
static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, uint32_t seqno, int newbranch);
static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, const char * const explicit_uri);
static int init_resp(struct sip_request *resp, const char *msg);
static inline int resp_needs_contact(const char *msg, enum sipmethod method);
static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg, const struct sip_request *req);
static const struct ast_sockaddr *sip_real_dst(const struct sip_pvt *p);
static void build_via(struct sip_pvt *p);
static int create_addr_from_peer(struct sip_pvt *r, struct sip_peer *peer);
static int create_addr(struct sip_pvt *dialog, const char *opeer, struct ast_sockaddr *addr, int newdialog);
static char *generate_random_string(char *buf, size_t size);
static void build_callid_pvt(struct sip_pvt *pvt);
static void change_callid_pvt(struct sip_pvt *pvt, const char *callid);
static void build_callid_registry(struct sip_registry *reg, const struct ast_sockaddr *ourip, const char *fromdomain);
static void build_localtag_registry(struct sip_registry *reg);
static void make_our_tag(struct sip_pvt *pvt);
static int add_header(struct sip_request *req, const char *var, const char *value);
static int add_max_forwards(struct sip_pvt *dialog, struct sip_request *req);
static int add_content(struct sip_request *req, const char *line);
static int finalize_content(struct sip_request *req);
static void destroy_msg_headers(struct sip_pvt *pvt);
static int add_text(struct sip_request *req, struct sip_pvt *p);
static int add_digit(struct sip_request *req, char digit, unsigned int duration, int mode);
static int add_rpid(struct sip_request *req, struct sip_pvt *p);
static int add_vidupdate(struct sip_request *req);
static void add_route(struct sip_request *req, struct sip_route *route, int skip);
static int copy_header(struct sip_request *req, const struct sip_request *orig, const char *field);
static int copy_all_header(struct sip_request *req, const struct sip_request *orig, const char *field);
static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field);
static void set_destination(struct sip_pvt *p, const char *uri);
static void add_date(struct sip_request *req);
static void add_expires(struct sip_request *req, int expires);
static void build_contact(struct sip_pvt *p);

/*------Request handling functions */
static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, int *recount, int *nounlock);
static int handle_request_update(struct sip_pvt *p, struct sip_request *req);
static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, int *recount, const char *e, int *nounlock);
static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, uint32_t seqno, int *nounlock);
static int handle_request_bye(struct sip_pvt *p, struct sip_request *req);
static int handle_request_register(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *sin, const char *e);
static int handle_request_cancel(struct sip_pvt *p, struct sip_request *req);
static int handle_request_message(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e);
static void handle_request_info(struct sip_pvt *p, struct sip_request *req);
static int handle_request_options(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e);
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req,
                int *nounlock, struct sip_pvt *replaces_pvt, struct ast_channel *replaces_chan);
static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, uint32_t seqno, const char *e);
static int local_attended_transfer(struct sip_pvt *transferer, struct ast_channel *transferer_chan, uint32_t seqno, int *nounlock);

/*------Response handling functions */
static void handle_response_publish(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
static void handle_response_invite(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
static void handle_response_notify(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
static void handle_response_refer(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
static void handle_response_subscribe(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
static int handle_response_register(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);
static void handle_response(struct sip_pvt *p, int resp, const char *rest, struct sip_request *req, uint32_t seqno);

/*------ SRTP Support -------- */
static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struct ast_sdp_srtp **srtp, const char *a);

/*------ T38 Support --------- */
static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
static void change_t38_state(struct sip_pvt *p, int state);

/*------ Session-Timers functions --------- */
static void proc_422_rsp(struct sip_pvt *p, struct sip_request *rsp);
static int  proc_session_timer(const void *vp);
static void stop_session_timer(struct sip_pvt *p);
static void start_session_timer(struct sip_pvt *p);
static void restart_session_timer(struct sip_pvt *p);
static const char *strefresherparam2str(enum st_refresher r);
static int parse_session_expires(const char *p_hdrval, int *const p_interval, enum st_refresher_param *const p_ref);
static int parse_minse(const char *p_hdrval, int *const p_interval);
static int st_get_se(struct sip_pvt *, int max);
static enum st_refresher st_get_refresher(struct sip_pvt *);
static enum st_mode st_get_mode(struct sip_pvt *, int no_cached);
static struct sip_st_dlg* sip_st_alloc(struct sip_pvt *const p);

/*------- RTP Glue functions -------- */
static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance *instance, struct ast_rtp_instance *vinstance, struct ast_rtp_instance *tinstance, const struct ast_format_cap *cap, int nat_active);

/*!--- SIP MWI Subscription support */
static int sip_subscribe_mwi(const char *value, int lineno);
static void sip_subscribe_mwi_destroy(struct sip_subscription_mwi *mwi);
static void sip_send_all_mwi_subscriptions(void);
static int sip_subscribe_mwi_do(const void *data);
static int __sip_subscribe_mwi_do(struct sip_subscription_mwi *mwi);

/*! \brief Definition of this channel for PBX channel registration */
struct ast_channel_tech sip_tech = {
        .type = "SIP",
        .description = "Session Initiation Protocol (SIP)",
        .properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
        .requester = sip_request_call,                  /* called with chan unlocked */
        .devicestate = sip_devicestate,                 /* called with chan unlocked (not chan-specific) */
        .call = sip_call,                       /* called with chan locked */
        .send_html = sip_sendhtml,
        .hangup = sip_hangup,                   /* called with chan locked */
        .answer = sip_answer,                   /* called with chan locked */
        .read = sip_read,                       /* called with chan locked */
        .write = sip_write,                     /* called with chan locked */
        .write_video = sip_write,               /* called with chan locked */
        .write_text = sip_write,
        .indicate = sip_indicate,               /* called with chan locked */
        .transfer = sip_transfer,               /* called with chan locked */
        .fixup = sip_fixup,                     /* called with chan locked */
        .send_digit_begin = sip_senddigit_begin,        /* called with chan unlocked */
        .send_digit_end = sip_senddigit_end,
        .early_bridge = ast_rtp_instance_early_bridge,
        .send_text = sip_sendtext,              /* called with chan locked */
        .func_channel_read = sip_acf_channel_read,
        .setoption = sip_setoption,
        .queryoption = sip_queryoption,
        .get_pvt_uniqueid = sip_get_callid,
};

/*! \brief This version of the sip channel tech has no send_digit_begin
 * callback so that the core knows that the channel does not want
 * DTMF BEGIN frames.
 * The struct is initialized just before registering the channel driver,
 * and is for use with channels using SIP INFO DTMF.
 */
struct ast_channel_tech sip_tech_info;

/*------- CC Support -------- */
static int sip_cc_agent_init(struct ast_cc_agent *agent, struct ast_channel *chan);
static int sip_cc_agent_start_offer_timer(struct ast_cc_agent *agent);
static int sip_cc_agent_stop_offer_timer(struct ast_cc_agent *agent);
static void sip_cc_agent_respond(struct ast_cc_agent *agent, enum ast_cc_agent_response_reason reason);
static int sip_cc_agent_status_request(struct ast_cc_agent *agent);
static int sip_cc_agent_start_monitoring(struct ast_cc_agent *agent);
static int sip_cc_agent_recall(struct ast_cc_agent *agent);
static void sip_cc_agent_destructor(struct ast_cc_agent *agent);

static struct ast_cc_agent_callbacks sip_cc_agent_callbacks = {
        .type = "SIP",
        .init = sip_cc_agent_init,
        .start_offer_timer = sip_cc_agent_start_offer_timer,
        .stop_offer_timer = sip_cc_agent_stop_offer_timer,
        .respond = sip_cc_agent_respond,
        .status_request = sip_cc_agent_status_request,
        .start_monitoring = sip_cc_agent_start_monitoring,
        .callee_available = sip_cc_agent_recall,
        .destructor = sip_cc_agent_destructor,
};

/* -------- End of declarations of structures, constants and forward declarations of functions
   Below starts actual code
   ------------------------
*/

static int sip_epa_register(const struct epa_static_data *static_data)
{
        struct epa_backend *backend = ast_calloc(1, sizeof(*backend));

        if (!backend) {
                return -1;
        }

        backend->static_data = static_data;

        AST_LIST_LOCK(&epa_static_data_list);
        AST_LIST_INSERT_TAIL(&epa_static_data_list, backend, next);
        AST_LIST_UNLOCK(&epa_static_data_list);
        return 0;
}

static void sip_epa_unregister_all(void)
{
        struct epa_backend *backend;

        AST_LIST_LOCK(&epa_static_data_list);
        while ((backend = AST_LIST_REMOVE_HEAD(&epa_static_data_list, next))) {
                ast_free(backend);
        }
        AST_LIST_UNLOCK(&epa_static_data_list);
}

static void cc_handle_publish_error(struct sip_pvt *pvt, const int resp, struct sip_request *req, struct sip_epa_entry *epa_entry);

static void cc_epa_destructor(void *data)
{
        struct sip_epa_entry *epa_entry = data;
        struct cc_epa_entry *cc_entry = epa_entry->instance_data;
        ast_free(cc_entry);
}

static const struct epa_static_data cc_epa_static_data  = {
        .event = CALL_COMPLETION,
        .name = "call-completion",
        .handle_error = cc_handle_publish_error,
        .destructor = cc_epa_destructor,
};

static const struct epa_static_data *find_static_data(const char * const event_package)
{
        const struct epa_backend *backend = NULL;

        AST_LIST_LOCK(&epa_static_data_list);
        AST_LIST_TRAVERSE(&epa_static_data_list, backend, next) {
                if (!strcmp(backend->static_data->name, event_package)) {
                        break;
                }
        }
        AST_LIST_UNLOCK(&epa_static_data_list);
        return backend ? backend->static_data : NULL;
}

static struct sip_epa_entry *create_epa_entry (const char * const event_package, const char * const destination)
{
        struct sip_epa_entry *epa_entry;
        const struct epa_static_data *static_data;

        if (!(static_data = find_static_data(event_package))) {
                return NULL;
        }

        if (!(epa_entry = ao2_t_alloc(sizeof(*epa_entry), static_data->destructor, "Allocate new EPA entry"))) {
                return NULL;
        }

        epa_entry->static_data = static_data;
        ast_copy_string(epa_entry->destination, destination, sizeof(epa_entry->destination));
        return epa_entry;
}
static enum ast_cc_service_type service_string_to_service_type(const char * const service_string)
{
        enum ast_cc_service_type service;
        for (service = AST_CC_CCBS; service <= AST_CC_CCNL; ++service) {
                if (!strcasecmp(service_string, sip_cc_service_map[service].service_string)) {
                        return service;
                }
        }
        return AST_CC_NONE;
}

/* Even state compositors code */
static void esc_entry_destructor(void *obj)
{
        struct sip_esc_entry *esc_entry = obj;
        if (esc_entry->sched_id > -1) {
                AST_SCHED_DEL(sched, esc_entry->sched_id);
        }
}

static int esc_hash_fn(const void *obj, const int flags)
{
        const struct sip_esc_entry *entry = obj;
        return ast_str_hash(entry->entity_tag);
}

static int esc_cmp_fn(void *obj, void *arg, int flags)
{
        struct sip_esc_entry *entry1 = obj;
        struct sip_esc_entry *entry2 = arg;

        return (!strcmp(entry1->entity_tag, entry2->entity_tag)) ? (CMP_MATCH | CMP_STOP) : 0;
}

static struct event_state_compositor *get_esc(const char * const event_package) {
        int i;
        for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
                if (!strcasecmp(event_package, event_state_compositors[i].name)) {
                        return &event_state_compositors[i];
                }
        }
        return NULL;
}

static struct sip_esc_entry *get_esc_entry(const char * entity_tag, struct event_state_compositor *esc) {
        struct sip_esc_entry *entry;
        struct sip_esc_entry finder;

        ast_copy_string(finder.entity_tag, entity_tag, sizeof(finder.entity_tag));

        entry = ao2_find(esc->compositor, &finder, OBJ_POINTER);

        return entry;
}

static int publish_expire(const void *data)
{
        struct sip_esc_entry *esc_entry = (struct sip_esc_entry *) data;
        struct event_state_compositor *esc = get_esc(esc_entry->event);

        ast_assert(esc != NULL);

        ao2_unlink(esc->compositor, esc_entry);
        ao2_ref(esc_entry, -1);
        return 0;
}

static void create_new_sip_etag(struct sip_esc_entry *esc_entry, int is_linked)
{
        int new_etag = ast_atomic_fetchadd_int(&esc_etag_counter, +1);
        struct event_state_compositor *esc = get_esc(esc_entry->event);

        ast_assert(esc != NULL);
        if (is_linked) {
                ao2_unlink(esc->compositor, esc_entry);
        }
        snprintf(esc_entry->entity_tag, sizeof(esc_entry->entity_tag), "%d", new_etag);
        ao2_link(esc->compositor, esc_entry);
}

static struct sip_esc_entry *create_esc_entry(struct event_state_compositor *esc, struct sip_request *req, const int expires)
{
        struct sip_esc_entry *esc_entry;
        int expires_ms;

        if (!(esc_entry = ao2_alloc(sizeof(*esc_entry), esc_entry_destructor))) {
                return NULL;
        }

        esc_entry->event = esc->name;

        expires_ms = expires * 1000;
        /* Bump refcount for scheduler */
        ao2_ref(esc_entry, +1);
        esc_entry->sched_id = ast_sched_add(sched, expires_ms, publish_expire, esc_entry);

        /* Note: This links the esc_entry into the ESC properly */
        create_new_sip_etag(esc_entry, 0);

        return esc_entry;
}

static int initialize_escs(void)
{
        int i, res = 0;
        for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
                if (!((event_state_compositors[i].compositor) =
                                        ao2_container_alloc(ESC_MAX_BUCKETS, esc_hash_fn, esc_cmp_fn))) {
                        res = -1;
                }
        }
        return res;
}

static void destroy_escs(void)
{
        int i;
        for (i = 0; i < ARRAY_LEN(event_state_compositors); i++) {
                ao2_ref(event_state_compositors[i].compositor, -1);
        }
}


static int find_by_notify_uri_helper(void *obj, void *arg, int flags)
{
        struct ast_cc_agent *agent = obj;
        struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
        const char *uri = arg;

        return !sip_uri_cmp(agent_pvt->notify_uri, uri) ? CMP_MATCH | CMP_STOP : 0;
}

static struct ast_cc_agent *find_sip_cc_agent_by_notify_uri(const char * const uri)
{
        struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_notify_uri_helper, (char *)uri, "SIP");
        return agent;
}

static int find_by_subscribe_uri_helper(void *obj, void *arg, int flags)
{
        struct ast_cc_agent *agent = obj;
        struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
        const char *uri = arg;

        return !sip_uri_cmp(agent_pvt->subscribe_uri, uri) ? CMP_MATCH | CMP_STOP : 0;
}

static struct ast_cc_agent *find_sip_cc_agent_by_subscribe_uri(const char * const uri)
{
        struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_subscribe_uri_helper, (char *)uri, "SIP");
        return agent;
}

static int find_by_callid_helper(void *obj, void *arg, int flags)
{
        struct ast_cc_agent *agent = obj;
        struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
        struct sip_pvt *call_pvt = arg;

        return !strcmp(agent_pvt->original_callid, call_pvt->callid) ? CMP_MATCH | CMP_STOP : 0;
}

static struct ast_cc_agent *find_sip_cc_agent_by_original_callid(struct sip_pvt *pvt)
{
        struct ast_cc_agent *agent = ast_cc_agent_callback(0, find_by_callid_helper, pvt, "SIP");
        return agent;
}

static int sip_cc_agent_init(struct ast_cc_agent *agent, struct ast_channel *chan)
{
        struct sip_cc_agent_pvt *agent_pvt = ast_calloc(1, sizeof(*agent_pvt));
        struct sip_pvt *call_pvt = ast_channel_tech_pvt(chan);

        if (!agent_pvt) {
                return -1;
        }

        ast_assert(!strcmp(ast_channel_tech(chan)->type, "SIP"));

        ast_copy_string(agent_pvt->original_callid, call_pvt->callid, sizeof(agent_pvt->original_callid));
        ast_copy_string(agent_pvt->original_exten, call_pvt->exten, sizeof(agent_pvt->original_exten));
        agent_pvt->offer_timer_id = -1;
        agent->private_data = agent_pvt;
        sip_pvt_lock(call_pvt);
        ast_set_flag(&call_pvt->flags[0], SIP_OFFER_CC);
        sip_pvt_unlock(call_pvt);
        return 0;
}

static int sip_offer_timer_expire(const void *data)
{
        struct ast_cc_agent *agent = (struct ast_cc_agent *) data;
        struct sip_cc_agent_pvt *agent_pvt = agent->private_data;

        agent_pvt->offer_timer_id = -1;

        return ast_cc_failed(agent->core_id, "SIP agent %s's offer timer expired", agent->device_name);
}

static int sip_cc_agent_start_offer_timer(struct ast_cc_agent *agent)
{
        struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
        int when;

        when = ast_get_cc_offer_timer(agent->cc_params) * 1000;
        agent_pvt->offer_timer_id = ast_sched_add(sched, when, sip_offer_timer_expire, agent);
        return 0;
}

static int sip_cc_agent_stop_offer_timer(struct ast_cc_agent *agent)
{
        struct sip_cc_agent_pvt *agent_pvt = agent->private_data;

        AST_SCHED_DEL(sched, agent_pvt->offer_timer_id);
        return 0;
}

static void sip_cc_agent_respond(struct ast_cc_agent *agent, enum ast_cc_agent_response_reason reason)
{
        struct sip_cc_agent_pvt *agent_pvt = agent->private_data;

        sip_pvt_lock(agent_pvt->subscribe_pvt);
        ast_set_flag(&agent_pvt->subscribe_pvt->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED);
        if (reason == AST_CC_AGENT_RESPONSE_SUCCESS || !ast_strlen_zero(agent_pvt->notify_uri)) {
                /* The second half of this if statement may be a bit hard to grasp,
                 * so here's an explanation. When a subscription comes into
                 * chan_sip, as long as it is not malformed, it will be passed
                 * to the CC core. If the core senses an out-of-order state transition,
                 * then the core will call this callback with the "reason" set to a
                 * failure condition.
                 * However, an out-of-order state transition will occur during a resubscription
                 * for CC. In such a case, we can see that we have already generated a notify_uri
                 * and so we can detect that this isn't a *real* failure. Rather, it is just
                 * something the core doesn't recognize as a legitimate SIP state transition.
                 * Thus we respond with happiness and flowers.
                 */
                transmit_response(agent_pvt->subscribe_pvt, "200 OK", &agent_pvt->subscribe_pvt->initreq);
                transmit_cc_notify(agent, agent_pvt->subscribe_pvt, CC_QUEUED);
        } else {
                transmit_response(agent_pvt->subscribe_pvt, "500 Internal Error", &agent_pvt->subscribe_pvt->initreq);
        }
        sip_pvt_unlock(agent_pvt->subscribe_pvt);
        agent_pvt->is_available = TRUE;
}

static int sip_cc_agent_status_request(struct ast_cc_agent *agent)
{
        struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
        enum ast_device_state state = agent_pvt->is_available ? AST_DEVICE_NOT_INUSE : AST_DEVICE_INUSE;
        return ast_cc_agent_status_response(agent->core_id, state);
}

static int sip_cc_agent_start_monitoring(struct ast_cc_agent *agent)
{
        /* To start monitoring just means to wait for an incoming PUBLISH
         * to tell us that the caller has become available again. No special
         * action is needed
         */
        return 0;
}

static int sip_cc_agent_recall(struct ast_cc_agent *agent)
{
        struct sip_cc_agent_pvt *agent_pvt = agent->private_data;
        /* If we have received a PUBLISH beforehand stating that the caller in question
         * is not available, we can save ourself a bit of effort here and just report
         * the caller as busy
         */
        if (!agent_pvt->is_available) {
                return ast_cc_agent_caller_busy(agent->core_id, "Caller %s is busy, reporting to the core",
                                agent->device_name);
        }
        /* Otherwise, we transmit a NOTIFY to the caller and await either
         * a PUBLISH or an INVITE
         */
        sip_pvt_lock(agent_pvt->subscribe_pvt);
        transmit_cc_notify(agent, agent_pvt->subscribe_pvt, CC_READY);
        sip_pvt_unlock(agent_pvt->subscribe_pvt);
        return 0;
}

static void sip_cc_agent_destructor(struct ast_cc_agent *agent)
{
        struct sip_cc_agent_pvt *agent_pvt = agent->private_data;

        if (!agent_pvt) {
                /* The agent constructor probably failed. */
                return;
        }

        sip_cc_agent_stop_offer_timer(agent);
        if (agent_pvt->subscribe_pvt) {
                sip_pvt_lock(agent_pvt->subscribe_pvt);
                if (!ast_test_flag(&agent_pvt->subscribe_pvt->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED)) {
                        /* If we haven't sent a 200 OK for the SUBSCRIBE dialog yet, then we need to send a response letting
                         * the subscriber know something went wrong
                         */
                        transmit_response(agent_pvt->subscribe_pvt, "500 Internal Server Error", &agent_pvt->subscribe_pvt->initreq);
                }
                sip_pvt_unlock(agent_pvt->subscribe_pvt);
                agent_pvt->subscribe_pvt = dialog_unref(agent_pvt->subscribe_pvt, "SIP CC agent destructor: Remove ref to subscription");
        }
        ast_free(agent_pvt);
}


static int sip_monitor_instance_hash_fn(const void *obj, const int flags)
{
        const struct sip_monitor_instance *monitor_instance = obj;
        return monitor_instance->core_id;
}

static int sip_monitor_instance_cmp_fn(void *obj, void *arg, int flags)
{
        struct sip_monitor_instance *monitor_instance1 = obj;
        struct sip_monitor_instance *monitor_instance2 = arg;

        return monitor_instance1->core_id == monitor_instance2->core_id ? CMP_MATCH | CMP_STOP : 0;
}

static void sip_monitor_instance_destructor(void *data)
{
        struct sip_monitor_instance *monitor_instance = data;
        if (monitor_instance->subscription_pvt) {
                sip_pvt_lock(monitor_instance->subscription_pvt);
                monitor_instance->subscription_pvt->expiry = 0;
                transmit_invite(monitor_instance->subscription_pvt, SIP_SUBSCRIBE, FALSE, 0, monitor_instance->subscribe_uri);
                sip_pvt_unlock(monitor_instance->subscription_pvt);
                dialog_unref(monitor_instance->subscription_pvt, "Unref monitor instance ref of subscription pvt");
        }
        if (monitor_instance->suspension_entry) {
                monitor_instance->suspension_entry->body[0] = '\0';
                transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_REMOVE ,monitor_instance->notify_uri);
                ao2_t_ref(monitor_instance->suspension_entry, -1, "Decrementing suspension entry refcount in sip_monitor_instance_destructor");
        }
        ast_string_field_free_memory(monitor_instance);
}

static struct sip_monitor_instance *sip_monitor_instance_init(int core_id, const char * const subscribe_uri, const char * const peername, const char * const device_name)
{
        struct sip_monitor_instance *monitor_instance = ao2_alloc(sizeof(*monitor_instance), sip_monitor_instance_destructor);

        if (!monitor_instance) {
                return NULL;
        }

        if (ast_string_field_init(monitor_instance, 256)) {
                ao2_ref(monitor_instance, -1);
                return NULL;
        }

        ast_string_field_set(monitor_instance, subscribe_uri, subscribe_uri);
        ast_string_field_set(monitor_instance, peername, peername);
        ast_string_field_set(monitor_instance, device_name, device_name);
        monitor_instance->core_id = core_id;
        ao2_link(sip_monitor_instances, monitor_instance);
        return monitor_instance;
}

static int find_sip_monitor_instance_by_subscription_pvt(void *obj, void *arg, int flags)
{
        struct sip_monitor_instance *monitor_instance = obj;
        return monitor_instance->subscription_pvt == arg ? CMP_MATCH | CMP_STOP : 0;
}

static int find_sip_monitor_instance_by_suspension_entry(void *obj, void *arg, int flags)
{
        struct sip_monitor_instance *monitor_instance = obj;
        return monitor_instance->suspension_entry == arg ? CMP_MATCH | CMP_STOP : 0;
}

static int sip_cc_monitor_request_cc(struct ast_cc_monitor *monitor, int *available_timer_id);
static int sip_cc_monitor_suspend(struct ast_cc_monitor *monitor);
static int sip_cc_monitor_unsuspend(struct ast_cc_monitor *monitor);
static int sip_cc_monitor_cancel_available_timer(struct ast_cc_monitor *monitor, int *sched_id);
static void sip_cc_monitor_destructor(void *private_data);

static struct ast_cc_monitor_callbacks sip_cc_monitor_callbacks = {
        .type = "SIP",
        .request_cc = sip_cc_monitor_request_cc,
        .suspend = sip_cc_monitor_suspend,
        .unsuspend = sip_cc_monitor_unsuspend,
        .cancel_available_timer = sip_cc_monitor_cancel_available_timer,
        .destructor = sip_cc_monitor_destructor,
};

static int sip_cc_monitor_request_cc(struct ast_cc_monitor *monitor, int *available_timer_id)
{
        struct sip_monitor_instance *monitor_instance = monitor->private_data;
        enum ast_cc_service_type service = monitor->service_offered;
        int when;

        if (!monitor_instance) {
                return -1;
        }

        if (!(monitor_instance->subscription_pvt = sip_alloc(NULL, NULL, 0, SIP_SUBSCRIBE, NULL, NULL))) {
                return -1;
        }

        when = service == AST_CC_CCBS ? ast_get_ccbs_available_timer(monitor->interface->config_params) :
                ast_get_ccnr_available_timer(monitor->interface->config_params);

        sip_pvt_lock(monitor_instance->subscription_pvt);
        ast_set_flag(&monitor_instance->subscription_pvt->flags[0], SIP_OUTGOING);
        create_addr(monitor_instance->subscription_pvt, monitor_instance->peername, 0, 1);
        ast_sip_ouraddrfor(&monitor_instance->subscription_pvt->sa, &monitor_instance->subscription_pvt->ourip, monitor_instance->subscription_pvt);
        monitor_instance->subscription_pvt->subscribed = CALL_COMPLETION;
        monitor_instance->subscription_pvt->expiry = when;

        transmit_invite(monitor_instance->subscription_pvt, SIP_SUBSCRIBE, FALSE, 2, monitor_instance->subscribe_uri);
        sip_pvt_unlock(monitor_instance->subscription_pvt);

        ao2_t_ref(monitor, +1, "Adding a ref to the monitor for the scheduler");
        *available_timer_id = ast_sched_add(sched, when * 1000, ast_cc_available_timer_expire, monitor);
        return 0;
}

static int construct_pidf_body(enum sip_cc_publish_state state, char *pidf_body, size_t size, const char *presentity)
{
        struct ast_str *body = ast_str_alloca(size);
        char tuple_id[32];

        generate_random_string(tuple_id, sizeof(tuple_id));

        /* We'll make this a bare-bones pidf body. In state_notify_build_xml, the PIDF
         * body gets a lot more extra junk that isn't necessary, so we'll leave it out here.
         */
        ast_str_append(&body, 0, "<?xml version=\"1.0\" encoding=\"UTF-8\"?>\n");
        /* XXX The entity attribute is currently set to the peer name associated with the
         * dialog. This is because we currently only call this function for call-completion
         * PUBLISH bodies. In such cases, the entity is completely disregarded. For other
         * event packages, it may be crucial to have a proper URI as the presentity so this
         * should be revisited as support is expanded.
         */
        ast_str_append(&body, 0, "<presence xmlns=\"urn:ietf:params:xml:ns:pidf\" entity=\"%s\">\n", presentity);
        ast_str_append(&body, 0, "<tuple id=\"%s\">\n", tuple_id);
        ast_str_append(&body, 0, "<status><basic>%s</basic></status>\n", state == CC_OPEN ? "open" : "closed");
        ast_str_append(&body, 0, "</tuple>\n");
        ast_str_append(&body, 0, "</presence>\n");
        ast_copy_string(pidf_body, ast_str_buffer(body), size);
        return 0;
}

static int sip_cc_monitor_suspend(struct ast_cc_monitor *monitor)
{
        struct sip_monitor_instance *monitor_instance = monitor->private_data;
        enum sip_publish_type publish_type;
        struct cc_epa_entry *cc_entry;

        if (!monitor_instance) {
                return -1;
        }

        if (!monitor_instance->suspension_entry) {
                /* We haven't yet allocated the suspension entry, so let's give it a shot */
                if (!(monitor_instance->suspension_entry = create_epa_entry("call-completion", monitor_instance->peername))) {
                        ast_log(LOG_WARNING, "Unable to allocate sip EPA entry for call-completion\n");
                        ao2_ref(monitor_instance, -1);
                        return -1;
                }
                if (!(cc_entry = ast_calloc(1, sizeof(*cc_entry)))) {
                        ast_log(LOG_WARNING, "Unable to allocate space for instance data of EPA entry for call-completion\n");
                        ao2_ref(monitor_instance, -1);
                        return -1;
                }
                cc_entry->core_id = monitor->core_id;
                monitor_instance->suspension_entry->instance_data = cc_entry;
                publish_type = SIP_PUBLISH_INITIAL;
        } else {
                publish_type = SIP_PUBLISH_MODIFY;
                cc_entry = monitor_instance->suspension_entry->instance_data;
        }

        cc_entry->current_state = CC_CLOSED;

        if (ast_strlen_zero(monitor_instance->notify_uri)) {
                /* If we have no set notify_uri, then what this means is that we have
                 * not received a NOTIFY from this destination stating that he is
                 * currently available.
                 *
                 * This situation can arise when the core calls the suspend callbacks
                 * of multiple destinations. If one of the other destinations aside
                 * from this one notified Asterisk that he is available, then there
                 * is no reason to take any suspension action on this device. Rather,
                 * we should return now and if we receive a NOTIFY while monitoring
                 * is still "suspended" then we can immediately respond with the
                 * proper PUBLISH to let this endpoint know what is going on.
                 */
                return 0;
        }
        construct_pidf_body(CC_CLOSED, monitor_instance->suspension_entry->body, sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
        return transmit_publish(monitor_instance->suspension_entry, publish_type, monitor_instance->notify_uri);
}

static int sip_cc_monitor_unsuspend(struct ast_cc_monitor *monitor)
{
        struct sip_monitor_instance *monitor_instance = monitor->private_data;
        struct cc_epa_entry *cc_entry;

        if (!monitor_instance) {
                return -1;
        }

        ast_assert(monitor_instance->suspension_entry != NULL);

        cc_entry = monitor_instance->suspension_entry->instance_data;
        cc_entry->current_state = CC_OPEN;
        if (ast_strlen_zero(monitor_instance->notify_uri)) {
                /* This means we are being asked to unsuspend a call leg we never
                 * sent a PUBLISH on. As such, there is no reason to send another
                 * PUBLISH at this point either. We can just return instead.
                 */
                return 0;
        }
        construct_pidf_body(CC_OPEN, monitor_instance->suspension_entry->body, sizeof(monitor_instance->suspension_entry->body), monitor_instance->peername);
        return transmit_publish(monitor_instance->suspension_entry, SIP_PUBLISH_MODIFY, monitor_instance->notify_uri);
}

static int sip_cc_monitor_cancel_available_timer(struct ast_cc_monitor *monitor, int *sched_id)
{
        if (*sched_id != -1) {
                AST_SCHED_DEL(sched, *sched_id);
                ao2_t_ref(monitor, -1, "Removing scheduler's reference to the monitor");
        }
        return 0;
}

static void sip_cc_monitor_destructor(void *private_data)
{
        struct sip_monitor_instance *monitor_instance = private_data;
        ao2_unlink(sip_monitor_instances, monitor_instance);
        ast_module_unref(ast_module_info->self);
}

static int sip_get_cc_information(struct sip_request *req, char *subscribe_uri, size_t size, enum ast_cc_service_type *service)
{
        char *call_info = ast_strdupa(sip_get_header(req, "Call-Info"));
        char *uri;
        char *purpose;
        char *service_str;
        static const char cc_purpose[] = "purpose=call-completion";
        static const int cc_purpose_len = sizeof(cc_purpose) - 1;

        if (ast_strlen_zero(call_info)) {
                /* No Call-Info present. Definitely no CC offer */
                return -1;
        }

        uri = strsep(&call_info, ";");

        while ((purpose = strsep(&call_info, ";"))) {
                if (!strncmp(purpose, cc_purpose, cc_purpose_len)) {
                        break;
                }
        }
        if (!purpose) {
                /* We didn't find the appropriate purpose= parameter. Oh well */
                return -1;
        }

        /* Okay, call-completion has been offered. Let's figure out what type of service this is */
        while ((service_str = strsep(&call_info, ";"))) {
                if (!strncmp(service_str, "m=", 2)) {
                        break;
                }
        }
        if (!service_str) {
                /* So they didn't offer a particular service, We'll just go with CCBS since it really
                 * doesn't matter anyway
                 */
                service_str = "BS";
        } else {
                /* We already determined that there is an "m=" so no need to check
                 * the result of this strsep
                 */
                strsep(&service_str, "=");
        }

        if ((*service = service_string_to_service_type(service_str)) == AST_CC_NONE) {
                /* Invalid service offered */
                return -1;
        }

        ast_copy_string(subscribe_uri, get_in_brackets(uri), size);

        return 0;
}

/*
 * \brief Determine what, if any, CC has been offered and queue a CC frame if possible
 *
 * After taking care of some formalities to be sure that this call is eligible for CC,
 * we first try to see if we can make use of native CC. We grab the information from
 * the passed-in sip_request (which is always a response to an INVITE). If we can
 * use native CC monitoring for the call, then so be it.
 *
 * If native cc monitoring is not possible or not supported, then we will instead attempt
 * to use generic monitoring. Falling back to generic from a failed attempt at using native
 * monitoring will only work if the monitor policy of the endpoint is "always"
 *
 * \param pvt The current dialog. Contains CC parameters for the endpoint
 * \param req The response to the INVITE we want to inspect
 * \param service The service to use if generic monitoring is to be used. For native
 * monitoring, we get the service from the SIP response itself
 */
static void sip_handle_cc(struct sip_pvt *pvt, struct sip_request *req, enum ast_cc_service_type service)
{
        enum ast_cc_monitor_policies monitor_policy = ast_get_cc_monitor_policy(pvt->cc_params);
        int core_id;
        char interface_name[AST_CHANNEL_NAME];

        if (monitor_policy == AST_CC_MONITOR_NEVER) {
                /* Don't bother, just return */
                return;
        }

        if ((core_id = ast_cc_get_current_core_id(pvt->owner)) == -1) {
                /* For some reason, CC is invalid, so don't try it! */
                return;
        }

        ast_channel_get_device_name(pvt->owner, interface_name, sizeof(interface_name));

        if (monitor_policy == AST_CC_MONITOR_ALWAYS || monitor_policy == AST_CC_MONITOR_NATIVE) {
                char subscribe_uri[SIPBUFSIZE];
                char device_name[AST_CHANNEL_NAME];
                enum ast_cc_service_type offered_service;
                struct sip_monitor_instance *monitor_instance;
                if (sip_get_cc_information(req, subscribe_uri, sizeof(subscribe_uri), &offered_service)) {
                        /* If CC isn't being offered to us, or for some reason the CC offer is
                         * not formatted correctly, then it may still be possible to use generic
                         * call completion since the monitor policy may be "always"
                         */
                        goto generic;
                }
                ast_channel_get_device_name(pvt->owner, device_name, sizeof(device_name));
                if (!(monitor_instance = sip_monitor_instance_init(core_id, subscribe_uri, pvt->peername, device_name))) {
                        /* Same deal. We can try using generic still */
                        goto generic;
                }
                /* We bump the refcount of chan_sip because once we queue this frame, the CC core
                 * will have a reference to callbacks in this module. We decrement the module
                 * refcount once the monitor destructor is called
                 */
                ast_module_ref(ast_module_info->self);
                ast_queue_cc_frame(pvt->owner, "SIP", pvt->dialstring, offered_service, monitor_instance);
                ao2_ref(monitor_instance, -1);
                return;
        }

generic:
        if (monitor_policy == AST_CC_MONITOR_GENERIC || monitor_policy == AST_CC_MONITOR_ALWAYS) {
                ast_queue_cc_frame(pvt->owner, AST_CC_GENERIC_MONITOR_TYPE, interface_name, service, NULL);
        }
}

/*! \brief Working TLS connection configuration */
static struct ast_tls_config sip_tls_cfg;

/*! \brief Default TLS connection configuration */
static struct ast_tls_config default_tls_cfg;

/*! \brief The TCP server definition */
static struct ast_tcptls_session_args sip_tcp_desc = {
        .accept_fd = -1,
        .master = AST_PTHREADT_NULL,
        .tls_cfg = NULL,
        .poll_timeout = -1,
        .name = "SIP TCP server",
        .accept_fn = ast_tcptls_server_root,
        .worker_fn = sip_tcp_worker_fn,
};

/*! \brief The TCP/TLS server definition */
static struct ast_tcptls_session_args sip_tls_desc = {
        .accept_fd = -1,
        .master = AST_PTHREADT_NULL,
        .tls_cfg = &sip_tls_cfg,
        .poll_timeout = -1,
        .name = "SIP TLS server",
        .accept_fn = ast_tcptls_server_root,
        .worker_fn = sip_tcp_worker_fn,
};

/*! \brief Append to SIP dialog history
        \return Always returns 0 */
#define append_history(p, event, fmt , args... )        append_history_full(p, "%-15s " fmt, event, ## args)

struct sip_pvt *dialog_ref_debug(struct sip_pvt *p, const char *tag, char *file, int line, const char *func)
{
        if (p)
#ifdef REF_DEBUG
                __ao2_ref_debug(p, 1, tag, file, line, func);
#else
                ao2_ref(p, 1);
#endif
        else
                ast_log(LOG_ERROR, "Attempt to Ref a null pointer\n");
        return p;
}

struct sip_pvt *dialog_unref_debug(struct sip_pvt *p, const char *tag, char *file, int line, const char *func)
{
        if (p)
#ifdef REF_DEBUG
                __ao2_ref_debug(p, -1, tag, file, line, func);
#else
                ao2_ref(p, -1);
#endif
        return NULL;
}

/*! \brief map from an integer value to a string.
 * If no match is found, return errorstring
 */
static const char *map_x_s(const struct _map_x_s *table, int x, const char *errorstring)
{
        const struct _map_x_s *cur;

        for (cur = table; cur->s; cur++) {
                if (cur->x == x) {
                        return cur->s;
                }
        }
        return errorstring;
}

/*! \brief map from a string to an integer value, case insensitive.
 * If no match is found, return errorvalue.
 */
static int map_s_x(const struct _map_x_s *table, const char *s, int errorvalue)
{
        const struct _map_x_s *cur;

        for (cur = table; cur->s; cur++) {
                if (!strcasecmp(cur->s, s)) {
                        return cur->x;
                }
        }
        return errorvalue;
}

static enum AST_REDIRECTING_REASON sip_reason_str_to_code(const char *text)
{
        enum AST_REDIRECTING_REASON ast = AST_REDIRECTING_REASON_UNKNOWN;
        int i;

        for (i = 0; i < ARRAY_LEN(sip_reason_table); ++i) {
                if (!strcasecmp(text, sip_reason_table[i].text)) {
                        ast = sip_reason_table[i].code;
                        break;
                }
        }

        return ast;
}

static const char *sip_reason_code_to_str(struct ast_party_redirecting_reason *reason, int *table_lookup)
{
        int code = reason->code;

        /* If there's a specific string set, then we just
         * use it.
         */
        if (!ast_strlen_zero(reason->str)) {
                /* If we care about whether this can be found in
                 * the table, then we need to check about that.
                 */
                if (table_lookup) {
                        /* If the string is literally "unknown" then don't bother with the lookup
                         * because it can lead to a false negative.
                         */
                        if (!strcasecmp(reason->str, "unknown") ||
                                        sip_reason_str_to_code(reason->str) != AST_REDIRECTING_REASON_UNKNOWN) {
                                *table_lookup = TRUE;
                        } else {
                                *table_lookup = FALSE;
                        }
                }
                return reason->str;
        }

        if (table_lookup) {
                *table_lookup = TRUE;
        }

        if (code >= 0 && code < ARRAY_LEN(sip_reason_table)) {
                return sip_reason_table[code].text;
        }

        return "unknown";
}

/*!
 * \brief generic function for determining if a correct transport is being
 * used to contact a peer
 *
 * this is done as a macro so that the "tmpl" var can be passed either a
 * sip_request or a sip_peer
 */
#define check_request_transport(peer, tmpl) ({ \
        int ret = 0; \
        if (peer->socket.type == tmpl->socket.type) \
                ; \
        else if (!(peer->transports & tmpl->socket.type)) {\
                ast_log(LOG_ERROR, \
                        "'%s' is not a valid transport for '%s'. we only use '%s'! ending call.\n", \
                        sip_get_transport(tmpl->socket.type), peer->name, get_transport_list(peer->transports) \
                        ); \
                ret = 1; \
        } else if (peer->socket.type & AST_TRANSPORT_TLS) { \
                ast_log(LOG_WARNING, \
                        "peer '%s' HAS NOT USED (OR SWITCHED TO) TLS in favor of '%s' (but this was allowed in sip.conf)!\n", \
                        peer->name, sip_get_transport(tmpl->socket.type) \
                ); \
        } else { \
                ast_debug(1, \
                        "peer '%s' has contacted us over %s even though we prefer %s.\n", \
                        peer->name, sip_get_transport(tmpl->socket.type), sip_get_transport(peer->socket.type) \
                ); \
        }\
        (ret); \
})

/*! \brief
 * duplicate a list of channel variables, \return the copy.
 */
static struct ast_variable *copy_vars(struct ast_variable *src)
{
        struct ast_variable *res = NULL, *tmp, *v = NULL;

        for (v = src ; v ; v = v->next) {
                if ((tmp = ast_variable_new(v->name, v->value, v->file))) {
                        tmp->next = res;
                        res = tmp;
                }
        }
        return res;
}

static void tcptls_packet_destructor(void *obj)
{
        struct tcptls_packet *packet = obj;

        ast_free(packet->data);
}

static void sip_tcptls_client_args_destructor(void *obj)
{
        struct ast_tcptls_session_args *args = obj;
        if (args->tls_cfg) {
                ast_free(args->tls_cfg->certfile);
                ast_free(args->tls_cfg->pvtfile);
                ast_free(args->tls_cfg->cipher);
                ast_free(args->tls_cfg->cafile);
                ast_free(args->tls_cfg->capath);

                ast_ssl_teardown(args->tls_cfg);
        }
        ast_free(args->tls_cfg);
        ast_free((char *) args->name);
}

static void sip_threadinfo_destructor(void *obj)
{
        struct sip_threadinfo *th = obj;
        struct tcptls_packet *packet;

        if (th->alert_pipe[1] > -1) {
                close(th->alert_pipe[0]);
        }
        if (th->alert_pipe[1] > -1) {
                close(th->alert_pipe[1]);
        }
        th->alert_pipe[0] = th->alert_pipe[1] = -1;

        while ((packet = AST_LIST_REMOVE_HEAD(&th->packet_q, entry))) {
                ao2_t_ref(packet, -1, "thread destruction, removing packet from frame queue");
        }

        if (th->tcptls_session) {
                ao2_t_ref(th->tcptls_session, -1, "remove tcptls_session for sip_threadinfo object");
        }
}

/*! \brief creates a sip_threadinfo object and links it into the threadt table. */
static struct sip_threadinfo *sip_threadinfo_create(struct ast_tcptls_session_instance *tcptls_session, int transport)
{
        struct sip_threadinfo *th;

        if (!tcptls_session || !(th = ao2_alloc(sizeof(*th), sip_threadinfo_destructor))) {
                return NULL;
        }

        th->alert_pipe[0] = th->alert_pipe[1] = -1;

        if (pipe(th->alert_pipe) == -1) {
                ao2_t_ref(th, -1, "Failed to open alert pipe on sip_threadinfo");
                ast_log(LOG_ERROR, "Could not create sip alert pipe in tcptls thread, error %s\n", strerror(errno));
                return NULL;
        }
        ao2_t_ref(tcptls_session, +1, "tcptls_session ref for sip_threadinfo object");
        th->tcptls_session = tcptls_session;
        th->type = transport ? transport : (tcptls_session->ssl ? AST_TRANSPORT_TLS: AST_TRANSPORT_TCP);
        ao2_t_link(threadt, th, "Adding new tcptls helper thread");
        ao2_t_ref(th, -1, "Decrementing threadinfo ref from alloc, only table ref remains");
        return th;
}

/*! \brief used to indicate to a tcptls thread that data is ready to be written */
static int sip_tcptls_write(struct ast_tcptls_session_instance *tcptls_session, const void *buf, size_t len)
{
        int res = len;
        struct sip_threadinfo *th = NULL;
        struct tcptls_packet *packet = NULL;
        struct sip_threadinfo tmp = {
                .tcptls_session = tcptls_session,
        };
        enum sip_tcptls_alert alert = TCPTLS_ALERT_DATA;

        if (!tcptls_session) {
                return XMIT_ERROR;
        }

        ao2_lock(tcptls_session);

        if ((tcptls_session->fd == -1) ||
                !(th = ao2_t_find(threadt, &tmp, OBJ_POINTER, "ao2_find, getting sip_threadinfo in tcp helper thread")) ||
                !(packet = ao2_alloc(sizeof(*packet), tcptls_packet_destructor)) ||
                !(packet->data = ast_str_create(len))) {
                goto tcptls_write_setup_error;
        }

        /* goto tcptls_write_error should _NOT_ be used beyond this point */
        ast_str_set(&packet->data, 0, "%s", (char *) buf);
        packet->len = len;

        /* alert tcptls thread handler that there is a packet to be sent.
         * must lock the thread info object to guarantee control of the
         * packet queue */
        ao2_lock(th);
        if (write(th->alert_pipe[1], &alert, sizeof(alert)) == -1) {
                ast_log(LOG_ERROR, "write() to alert pipe failed: %s\n", strerror(errno));
                ao2_t_ref(packet, -1, "could not write to alert pipe, remove packet");
                packet = NULL;
                res = XMIT_ERROR;
        } else { /* it is safe to queue the frame after issuing the alert when we hold the threadinfo lock */
                AST_LIST_INSERT_TAIL(&th->packet_q, packet, entry);
        }
        ao2_unlock(th);

        ao2_unlock(tcptls_session);
        ao2_t_ref(th, -1, "In sip_tcptls_write, unref threadinfo object after finding it");
        return res;

tcptls_write_setup_error:
        if (th) {
                ao2_t_ref(th, -1, "In sip_tcptls_write, unref threadinfo obj, could not create packet");
        }
        if (packet) {
                ao2_t_ref(packet, -1, "could not allocate packet's data");
        }
        ao2_unlock(tcptls_session);

        return XMIT_ERROR;
}

/*! \brief SIP TCP connection handler */
static void *sip_tcp_worker_fn(void *data)
{
        struct ast_tcptls_session_instance *tcptls_session = data;

        return _sip_tcp_helper_thread(tcptls_session);
}

/*! \brief SIP WebSocket connection handler */
static void sip_websocket_callback(struct ast_websocket *session, struct ast_variable *parameters, struct ast_variable *headers)
{
        int res;

        if (ast_websocket_set_nonblock(session)) {
                goto end;
        }

        while ((res = ast_wait_for_input(ast_websocket_fd(session), -1)) > 0) {
                char *payload;
                uint64_t payload_len;
                enum ast_websocket_opcode opcode;
                int fragmented;

                if (ast_websocket_read(session, &payload, &payload_len, &opcode, &fragmented)) {
                        /* We err on the side of caution and terminate the session if any error occurs */
                        break;
                }

                if (opcode == AST_WEBSOCKET_OPCODE_TEXT || opcode == AST_WEBSOCKET_OPCODE_BINARY) {
                        struct sip_request req = { 0, };

                        if (!(req.data = ast_str_create(payload_len + 1))) {
                                goto end;
                        }

                        if (ast_str_set(&req.data, -1, "%s", payload) == AST_DYNSTR_BUILD_FAILED) {
                                deinit_req(&req);
                                goto end;
                        }

                        req.socket.fd = ast_websocket_fd(session);
                        set_socket_transport(&req.socket, ast_websocket_is_secure(session) ? AST_TRANSPORT_WSS : AST_TRANSPORT_WS);
                        req.socket.ws_session = session;

                        handle_request_do(&req, ast_websocket_remote_address(session));
                        deinit_req(&req);

                } else if (opcode == AST_WEBSOCKET_OPCODE_CLOSE) {
                        break;
                }
        }

end:
        ast_websocket_unref(session);
}

/*! \brief Check if the authtimeout has expired.
 * \param start the time when the session started
 *
 * \retval 0 the timeout has expired
 * \retval -1 error
 * \return the number of milliseconds until the timeout will expire
 */
static int sip_check_authtimeout(time_t start)
{
        int timeout;
        time_t now;
        if(time(&now) == -1) {
                ast_log(LOG_ERROR, "error executing time(): %s\n", strerror(errno));
                return -1;
        }

        timeout = (authtimeout - (now - start)) * 1000;
        if (timeout < 0) {
                /* we have timed out */
                return 0;
        }

        return timeout;
}

/*!
 * \brief Read a SIP request or response from a TLS connection
 *
 * Because TLS operations are hidden from view via a FILE handle, the
 * logic for reading data is a bit complex, and we have to make periodic
 * checks to be sure we aren't taking too long to perform the necessary
 * action.
 *
 * \todo XXX This should be altered in the future not to use a FILE pointer
 *
 * \param req The request structure to fill in
 * \param tcptls_session The TLS connection on which the data is being received
 * \param authenticated A flag indicating whether authentication has occurred yet.
 *        This is only relevant in a server role.
 * \param start The time at which we started attempting to read data. Used in
 *        determining if there has been a timeout.
 * \param me Thread info. Used as a means of determining if the session needs to be stoppped.
 * \retval -1 Failed to read data
 * \retval 0 Succeeded in reading data
 */
static int sip_tls_read(struct sip_request *req, struct sip_request *reqcpy, struct ast_tcptls_session_instance *tcptls_session,
                        int authenticated, time_t start, struct sip_threadinfo *me)
{
        int res, content_length, after_poll = 1, need_poll = 1;
        size_t datalen = ast_str_strlen(req->data);
        char buf[1024] = "";
        int timeout = -1;

        /* Read in headers one line at a time */
        while (datalen < 4 || strncmp(REQ_OFFSET_TO_STR(req, data->used - 4), "\r\n\r\n", 4)) {
                if (!tcptls_session->client && !authenticated) {
                        if ((timeout = sip_check_authtimeout(start)) < 0) {
                                ast_debug(2, "SIP TLS server failed to determine authentication timeout\n");
                                return -1;
                        }

                        if (timeout == 0) {
                                ast_debug(2, "SIP TLS server timed out\n");
                                return -1;
                        }
                } else {
                        timeout = -1;
                }

                /* special polling behavior is required for TLS
                 * sockets because of the buffering done in the
                 * TLS layer */
                if (need_poll) {
                        need_poll = 0;
                        after_poll = 1;
                        res = ast_wait_for_input(tcptls_session->fd, timeout);
                        if (res < 0) {
                                ast_debug(2, "SIP TLS server :: ast_wait_for_input returned %d\n", res);
                                return -1;
                        } else if (res == 0) {
                                /* timeout */
                                ast_debug(2, "SIP TLS server timed out\n");
                                return -1;
                        }
                }

                ao2_lock(tcptls_session);
                if (!fgets(buf, sizeof(buf), tcptls_session->f)) {
                        ao2_unlock(tcptls_session);
                        if (after_poll) {
                                return -1;
                        } else {
                                need_poll = 1;
                                continue;
                        }
                }
                ao2_unlock(tcptls_session);
                after_poll = 0;
                if (me->stop) {
                        return -1;
                }
                ast_str_append(&req->data, 0, "%s", buf);

                datalen = ast_str_strlen(req->data);
                if (datalen > SIP_MAX_PACKET_SIZE) {
                        ast_log(LOG_WARNING, "Rejecting TLS packet from '%s' because way too large: %zu\n",
                                ast_sockaddr_stringify(&tcptls_session->remote_address), datalen);
                        return -1;
                }
        }
        copy_request(reqcpy, req);
        parse_request(reqcpy);
        /* In order to know how much to read, we need the content-length header */
        if (sscanf(sip_get_header(reqcpy, "Content-Length"), "%30d", &content_length)) {
                while (content_length > 0) {
                        size_t bytes_read;
                        if (!tcptls_session->client && !authenticated) {
                                if ((timeout = sip_check_authtimeout(start)) < 0) {
                                        return -1;
                                }

                                if (timeout == 0) {
                                        ast_debug(2, "SIP TLS server timed out\n");
                                        return -1;
                                }
                        } else {
                                timeout = -1;
                        }

                        if (need_poll) {
                                need_poll = 0;
                                after_poll = 1;
                                res = ast_wait_for_input(tcptls_session->fd, timeout);
                                if (res < 0) {
                                        ast_debug(2, "SIP TLS server :: ast_wait_for_input returned %d\n", res);
                                        return -1;
                                } else if (res == 0) {
                                        /* timeout */
                                        ast_debug(2, "SIP TLS server timed out\n");
                                        return -1;
                                }
                        }

                        ao2_lock(tcptls_session);
                        if (!(bytes_read = fread(buf, 1, MIN(sizeof(buf) - 1, content_length), tcptls_session->f))) {
                                ao2_unlock(tcptls_session);
                                if (after_poll) {
                                        return -1;
                                } else {
                                        need_poll = 1;
                                        continue;
                                }
                        }
                        buf[bytes_read] = '\0';
                        ao2_unlock(tcptls_session);
                        after_poll = 0;
                        if (me->stop) {
                                return -1;
                        }
                        content_length -= strlen(buf);
                        ast_str_append(&req->data, 0, "%s", buf);

                        datalen = ast_str_strlen(req->data);
                        if (datalen > SIP_MAX_PACKET_SIZE) {
                                ast_log(LOG_WARNING, "Rejecting TLS packet from '%s' because way too large: %zu\n",
                                        ast_sockaddr_stringify(&tcptls_session->remote_address), datalen);
                                return -1;
                        }
                }
        }
        /*! \todo XXX If there's no Content-Length or if the content-length and what
                                        we receive is not the same - we should generate an error */
        return 0;
}

/*!
 * \brief Indication of a TCP message's integrity
 */
enum message_integrity {
        /*!
         * The message has an error in it with
         * regards to its Content-Length header
         */
        MESSAGE_INVALID,
        /*!
         * The message is incomplete
         */
        MESSAGE_FRAGMENT,
        /*!
         * The data contains a complete message
         * plus a fragment of another.
         */
        MESSAGE_FRAGMENT_COMPLETE,
        /*!
         * The message is complete
         */
        MESSAGE_COMPLETE,
};

/*!
 * \brief
 * Get the content length from an unparsed SIP message
 *
 * \param message The unparsed SIP message headers
 * \return The value of the Content-Length header or -1 if message is invalid
 */
static int read_raw_content_length(const char *message)
{
        char *content_length_str;
        int content_length = -1;

        struct ast_str *msg_copy;
        char *msg;

        /* Using a ast_str because lws2sws takes one of those */
        if (!(msg_copy = ast_str_create(strlen(message) + 1))) {
                return -1;
        }
        ast_str_set(&msg_copy, 0, "%s", message);

        if (sip_cfg.pedanticsipchecking) {
                lws2sws(msg_copy);
        }

        msg = ast_str_buffer(msg_copy);

        /* Let's find a Content-Length header */
        if ((content_length_str = strcasestr(msg, "\nContent-Length:"))) {
                content_length_str += sizeof("\nContent-Length:") - 1;
        } else if ((content_length_str = strcasestr(msg, "\nl:"))) {
                content_length_str += sizeof("\nl:") - 1;
        } else {
                /* RFC 3261 18.3
                 * "In the case of stream-oriented transports such as TCP, the Content-
                 *  Length header field indicates the size of the body.  The Content-
                 *  Length header field MUST be used with stream oriented transports."
                 */
                goto done;
        }

        /* Double-check that this is a complete header */
        if (!strchr(content_length_str, '\n')) {
                goto done;
        }

        if (sscanf(content_length_str, "%30d", &content_length) != 1) {
                content_length = -1;
        }

done:
        ast_free(msg_copy);
        return content_length;
}

/*!
 * \brief Check that a message received over TCP is a full message
 *
 * This will take the information read in and then determine if
 * 1) The message is a full SIP request
 * 2) The message is a partial SIP request
 * 3) The message contains a full SIP request along with another partial request
 * \param data The unparsed incoming SIP message.
 * \param request The resulting request with extra fragments removed.
 * \param overflow If the message contains more than a full request, this is the remainder of the message
 * \return The resulting integrity of the message
 */
static enum message_integrity check_message_integrity(struct ast_str **request, struct ast_str **overflow)
{
        char *message = ast_str_buffer(*request);
        char *body;
        int content_length;
        int message_len = ast_str_strlen(*request);
        int body_len;

        /* Important pieces to search for in a SIP request are \r\n\r\n. This
         * marks either
         * 1) The division between the headers and body
         * 2) The end of the SIP request
         */
        body = strstr(message, "\r\n\r\n");
        if (!body) {
                /* This is clearly a partial message since we haven't reached an end
                 * yet.
                 */
                return MESSAGE_FRAGMENT;
        }
        body += sizeof("\r\n\r\n") - 1;
        body_len = message_len - (body - message);

        body[-1] = '\0';
        content_length = read_raw_content_length(message);
        body[-1] = '\n';

        if (content_length < 0) {
                return MESSAGE_INVALID;
        } else if (content_length == 0) {
                /* We've definitely received an entire message. We need
                 * to check if there's also a fragment of another message
                 * in addition.
                 */
                if (body_len == 0) {
                        return MESSAGE_COMPLETE;
                } else {
                        ast_str_append(overflow, 0, "%s", body);
                        ast_str_truncate(*request, message_len - body_len);
                        return MESSAGE_FRAGMENT_COMPLETE;
                }
        }
        /* Positive content length. Let's see what sort of
         * message body we're dealing with.
         */
        if (body_len < content_length) {
                /* We don't have the full message body yet */
                return MESSAGE_FRAGMENT;
        } else if (body_len > content_length) {
                /* We have the full message plus a fragment of a further
                 * message
                 */
                ast_str_append(overflow, 0, "%s", body + content_length);
                ast_str_truncate(*request, message_len - (body_len - content_length));
                return MESSAGE_FRAGMENT_COMPLETE;
        } else {
                /* Yay! Full message with no extra content */
                return MESSAGE_COMPLETE;
        }
}

/*!
 * \brief Read SIP request or response from a TCP connection
 *
 * \param req The request structure to be filled in
 * \param tcptls_session The TCP connection from which to read
 * \retval -1 Failed to read data
 * \retval 0 Successfully read data
 */
static int sip_tcp_read(struct sip_request *req, struct ast_tcptls_session_instance *tcptls_session,
                int authenticated, time_t start)
{
        enum message_integrity message_integrity = MESSAGE_FRAGMENT;

        while (message_integrity == MESSAGE_FRAGMENT) {
                size_t datalen;

                if (ast_str_strlen(tcptls_session->overflow_buf) == 0) {
                        char readbuf[4097];
                        int timeout;
                        int res;
                        if (!tcptls_session->client && !authenticated) {
                                if ((timeout = sip_check_authtimeout(start)) < 0) {
                                        return -1;
                                }

                                if (timeout == 0) {
                                        ast_debug(2, "SIP TCP server timed out\n");
                                        return -1;
                                }
                        } else {
                                timeout = -1;
                        }
                        res = ast_wait_for_input(tcptls_session->fd, timeout);
                        if (res < 0) {
                                ast_debug(2, "SIP TCP server :: ast_wait_for_input returned %d\n", res);
                                return -1;
                        } else if (res == 0) {
                                ast_debug(2, "SIP TCP server timed out\n");
                                return -1;
                        }

                        res = recv(tcptls_session->fd, readbuf, sizeof(readbuf) - 1, 0);
                        if (res < 0) {
                                ast_debug(2, "SIP TCP server error when receiving data\n");
                                return -1;
                        } else if (res == 0) {
                                ast_debug(2, "SIP TCP server has shut down\n");
                                return -1;
                        }
                        readbuf[res] = '\0';
                        ast_str_append(&req->data, 0, "%s", readbuf);
                } else {
                        ast_str_append(&req->data, 0, "%s", ast_str_buffer(tcptls_session->overflow_buf));
                        ast_str_reset(tcptls_session->overflow_buf);
                }

                datalen = ast_str_strlen(req->data);
                if (datalen > SIP_MAX_PACKET_SIZE) {
                        ast_log(LOG_WARNING, "Rejecting TCP packet from '%s' because way too large: %zu\n",
                                ast_sockaddr_stringify(&tcptls_session->remote_address), datalen);
                        return -1;