Move the REF_DEBUG comment higher in the include list.

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

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2006, 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
 */

/*!
 * \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
        <depend>chan_local</depend>
 ***/

/*!  \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 */
/*
   Uncomment the define below,  if you are having refcount related memory leaks.
   With this uncommented, this module will generate a file, /tmp/refs, which contains
   a history of the ao2_ref() calls. To be useful, all calls to ao2_* functions should
   be modified to ao2_t_* calls, and include a tag describing what is happening with
   enough detail, to make pairing up a reference count increment with its corresponding decrement.
   The refcounter program in utils/ can be invaluable in highlighting objects that are not
   balanced, along with the complete history for that object.
   In normal operation, the macros defined will throw away the tags, so they do not
   affect the speed of the program at all. They can be considered to be documentation.
*/
/* #define  REF_DEBUG 1 */
#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/features.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/netsock.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/event.h"
#include "asterisk/stun.h"
#include "asterisk/cel.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 "sip/include/dialog.h"
#include "sip/include/dialplan_functions.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>
        <function name="SIP_HEADER" language="en_US">
                <synopsis>
                        Gets the specified SIP header.
                </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>
                </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="codecs">
                                                <para>The configured codecs.</para>
                                        </enum>
                                        <enum name="status">
                                                <para>Status (if qualify=yes).</para>
                                        </enum>
                                        <enum name="regexten">
                                                <para>Registration extension.</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 id for peer.</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 />
        </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 URI from the <literal>From:</literal> header.</para>
                                        </enum>
                                        <enum name="uri">
                                                <para>The URI from the <literal>Contact:</literal> header.</para>
                                        </enum>
                                        <enum name="useragent">
                                                <para>The useragent.</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 />
        </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.
                        Peerlist will follow as separate events, followed by a final event called
                        PeerlistComplete.</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>
        </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 RegistrationsComplete.</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 Variable: name=value sequences.</para>
                </description>
        </manager>
 ***/

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 mwi_expiry = DEFAULT_MWI_EXPIRY;

/*! \brief Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
        .flags = 0,
        .max_size = -1,
        .resync_threshold = -1,
        .impl = ""
};
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_UNSUPPORTED_METHOD },
        { SIP_PING,      NO_RTP, "PING",     CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
};

/*! \brief List of well-known SIP options. If we get this in a require,
   we should check the list and answer accordingly. */
static const struct cfsip_options {
        int id;             /*!< Bitmap ID */
        int supported;      /*!< Supported by Asterisk ? */
        char * const text;  /*!< Text id, as in standard */
} sip_options[] = {     /* XXX used in 3 places */
        /* RFC3262: PRACK 100% reliability */
        { SIP_OPT_100REL,       NOT_SUPPORTED,  "100rel" },
        /* RFC3959: SIP Early session support */
        { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED, "early-session" },
        /* SIMPLE events:  RFC4662 */
        { SIP_OPT_EVENTLIST,    NOT_SUPPORTED,  "eventlist" },
        /* RFC 4916- Connected line ID updates */
        { SIP_OPT_FROMCHANGE,   NOT_SUPPORTED,  "from-change" },
        /* GRUU: Globally Routable User Agent URI's */
        { SIP_OPT_GRUU,         NOT_SUPPORTED,  "gruu" },
        /* RFC4244 History info */
        { SIP_OPT_HISTINFO,     NOT_SUPPORTED,  "histinfo" },
        /* RFC3911: SIP Join header support */
        { SIP_OPT_JOIN,         NOT_SUPPORTED,  "join" },
        /* Disable the REFER subscription, RFC 4488 */
        { SIP_OPT_NOREFERSUB,   NOT_SUPPORTED,  "norefersub" },
        /* SIP outbound - the final NAT battle - draft-sip-outbound */
        { SIP_OPT_OUTBOUND,     NOT_SUPPORTED,  "outbound" },
        /* RFC3327: Path support */
        { SIP_OPT_PATH,         NOT_SUPPORTED,  "path" },
        /* RFC3840: Callee preferences */
        { SIP_OPT_PREF,         NOT_SUPPORTED,  "pref" },
        /* RFC3312: Precondition support */
        { SIP_OPT_PRECONDITION, NOT_SUPPORTED,  "precondition" },
        /* RFC3323: Privacy with proxies*/
        { SIP_OPT_PRIVACY,      NOT_SUPPORTED,  "privacy" },
        /* RFC-ietf-sip-uri-list-conferencing-02.txt conference invite lists */
        { SIP_OPT_RECLISTINV,   NOT_SUPPORTED,  "recipient-list-invite" },
        /* RFC-ietf-sip-uri-list-subscribe-02.txt - subscription lists */
        { SIP_OPT_RECLISTSUB,   NOT_SUPPORTED,  "recipient-list-subscribe" },
        /* RFC3891: Replaces: header for transfer */
        { SIP_OPT_REPLACES,     SUPPORTED,      "replaces" },
        /* One version of Polycom firmware has the wrong label */
        { SIP_OPT_REPLACES,     SUPPORTED,      "replace" },
        /* RFC4412 Resource priorities */
        { SIP_OPT_RESPRIORITY,  NOT_SUPPORTED,  "resource-priority" },
        /* RFC3329: Security agreement mechanism */
        { SIP_OPT_SEC_AGREE,    NOT_SUPPORTED,  "sec_agree" },
        /* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
        { SIP_OPT_SDP_ANAT,     NOT_SUPPORTED,  "sdp-anat" },
        /* RFC4028: SIP Session-Timers */
        { SIP_OPT_TIMER,        SUPPORTED,      "timer" },
        /* RFC4538: Target-dialog */
        { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED,  "tdialog" },
};

/*! \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"}
};


/*! \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 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 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 unsigned int default_transports;            /*!< Default Transports (enum sip_transport) that are acceptable */
static unsigned int default_primary_transport;     /*!< Default primary Transport (enum sip_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);    \
        }       \

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_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 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_dynamic_exclude_static = 0; /*!< Exclude static peers from contact registrations */
/*@}*/

/*! \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[2] = {{0}};  /*!< global SIP_ flags */
static int global_t38_maxdatagram;                /*!< global T.38 FaxMaxDatagram override */

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 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

/*! \brief
 * 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  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 Authentication list for realm authentication
 * \todo Move the sip_auth list to AST_LIST */
static struct sip_auth *authl = NULL;

/* --- 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 sockaddr_in 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 sockaddr_in internip;

/*! \brief our external IP address/port for SIP sessions.
 * externip.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 "externip = 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;
 *
 * + with "stunaddr = host[:port]" we run queries every externrefresh seconds
 *   to the specified server, and store the result in externip.
 *
 * Other variables (externhost, externexpire, externrefresh) are used
 * to support the above functions.
 */
static struct sockaddr_in externip;       /*!< External IP address if we are behind NAT */
static struct sockaddr_in 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 struct sockaddr_in stunaddr;     /*!< stun server address */
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 sockaddr_in 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)

/*---------------------------- 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, format_t format, const struct ast_channel *requestor, void *data, int *cause);
static int sip_devicestate(void *data);
static int sip_sendtext(struct ast_channel *ast, const char *text);
static int sip_call(struct ast_channel *ast, 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 sockaddr_in *sin);
static int sip_standard_port(enum sip_transport type, int port);
static int sip_prepare_socket(struct sip_pvt *p);

/*--- 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, int len);
static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, struct ast_str *data, int len, int fatal, int sipmethod);
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 sockaddr_in *sin, 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, int sipmethod, struct sip_request *req, enum xmittype reliable);
static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, enum xmittype reliable, int newbranch);
static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int seqno, enum xmittype reliable, int newbranch);
static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init);
static int transmit_reinvite_with_sdp(struct sip_pvt *p, int t38version, int oldsdp);
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_with_text(struct sip_pvt *p, const char *text);
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_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, int seqno);
static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int 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);
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, const struct ast_event *event, 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 sockaddr_in *sin, const int intended_method);
static void free_old_route(struct sip_route *route);
static void list_route(struct sip_route *route);
static void build_route(struct sip_pvt *p, struct sip_request *req, int backwards);
static enum check_auth_result register_verify(struct sip_pvt *p, struct sockaddr_in *sin,
                                              struct sip_request *req, const char *uri);
static struct sip_pvt *get_sip_pvt_byid_locked(const char *callid, const char *totag, const char *fromtag);
static void check_pendings(struct sip_pvt *p);
static void *sip_park_thread(void *stuff);
static int sip_park(struct ast_channel *chan1, struct ast_channel *chan2, struct sip_request *req, int seqno);
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_hostent *hp);
static int process_sdp_a_sendonly(const char *a, int *sendonly);
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_codec_to_sdp(const struct sip_pvt *p, format_t codec,
                             struct ast_str **m_buf, struct ast_str **a_buf,
                             int debug, int *min_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, int ignore);
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 sockaddr_in *sin, 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 sockaddr_in *sin);

/*--- 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 struct sip_auth *add_realm_authentication(struct sip_auth *authlist, const char *configuration, int lineno);
static int clear_realm_authentication(struct sip_auth *authlist);       /* Clear realm authentication list (at reload) */
static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, const char *realm);

/*--- Misc functions */
static void check_rtp_timeout(struct sip_pvt *dialog, time_t t);
static int sip_do_reload(enum channelreloadreason reason);
static int reload_config(enum channelreloadreason reason);
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 sip_addrcmp(char *name, struct sockaddr_in *sin); Support for peer matching */
static int sip_refer_allocate(struct sip_pvt *p);
static int sip_notify_allocate(struct sip_pvt *p);
static void ast_quiet_chan(struct ast_channel *chan);
static int attempt_transfer(struct sip_dual *transferer, struct sip_dual *target);
static int do_magic_pickup(struct ast_channel *channel, const char *extension, const char *context);

/*--- Device monitoring and Device/extension state/event handling */
static int cb_extensionstate(char *context, char* exten, int state, void *data);
static int sip_devicestate(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(const struct ast_event *, 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 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 const char *dtmfmode2str(int mode) attribute_const;
static int str2dtmfmode(const char *str) attribute_unused;
static const char *insecure2str(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_sipnotify(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);

/*--- 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 sockaddr_in *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 struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime, int forcenamematch, int devstate_only, int transport);
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);

/* Realtime device support */
static void realtime_update_peer(const char *peername, struct sockaddr_in *sin, const char *username, const char *fullcontact, const char *useragent, int expirey, unsigned short deprecated_username, int lastms);
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(struct ast_variable *var, const char *newpeername);
static struct sip_peer *realtime_peer(const char *peername, struct sockaddr_in *sin, int devstate_only);
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(struct in_addr *them, struct sockaddr_in *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 void append_date(struct sip_request *req);       /* Append date to SIP packet */
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_sip_options(struct sip_pvt *pvt, const char *supported);
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 *get_header(const struct sip_request *req, const char *name);
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 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 int lws2sws(char *msgbuf, int len);
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, 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);
static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
static int get_msg_text(char *buf, int len, struct sip_request *req, int addnewline);
static int transmit_state_notify(struct sip_pvt *p, int state, 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 void change_redirecting_information(struct sip_pvt *p, struct sip_request *req, struct ast_party_redirecting *redirecting, int set_call_forward);
static int get_domain(const char *str, char *domain, int len);
static void get_realm(struct sip_pvt *p, const struct sip_request *req);

/*-- TCP connection handling ---*/
static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, 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 int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, int seqno, int newbranch);
static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod);
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 sockaddr_in *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 sockaddr_in *sin, int newdialog);
static char *generate_random_string(char *buf, size_t size);
static void build_callid_pvt(struct sip_pvt *pvt);
static void build_callid_registry(struct sip_registry *reg, struct in_addr ourip, const char *fromdomain);
static void make_our_tag(char *tagbuf, size_t len);
static int add_header(struct sip_request *req, const char *var, const char *value);
static int add_header_contentLength(struct sip_request *req, int len);
static int add_line(struct sip_request *req, const char *line);
static int add_text(struct sip_request *req, const char *text);
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);
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, char *uri);
static void append_date(struct sip_request *req);
static void build_contact(struct sip_pvt *p);

/*------Request handling functions */
static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, 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, int debug, int seqno, struct sockaddr_in *sin, int *recount, const char *e, int *nounlock);
static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int debug, int 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 sockaddr_in *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);
static int handle_request_subscribe(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int 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);
static int handle_invite_replaces(struct sip_pvt *p, struct sip_request *req, int debug, int seqno, struct sockaddr_in *sin, int *nounlock);
static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, struct sockaddr_in *sin, int seqno, const char *e);
static int local_attended_transfer(struct sip_pvt *transferer, struct sip_dual *current, struct sip_request *req, int seqno, int *nounlock);

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

/*------ T38 Support --------- */
static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
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 *strefresher2str(enum st_refresher r);
static int parse_session_expires(const char *p_hdrval, int *const p_interval, enum st_refresher *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 *);
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, format_t codecs, 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 */
const struct ast_channel_tech sip_tech = {
        .type = "SIP",
        .description = "Session Initiation Protocol (SIP)",
        .capabilities = AST_FORMAT_AUDIO_MASK,  /* all audio formats */
        .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,
        .bridge = ast_rtp_instance_bridge,                      /* XXX chan unlocked ? */
        .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;

/*! \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, 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, 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(enum AST_REDIRECTING_REASON code)
{
        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", \
                        get_transport(tmpl->socket.type), peer->name, get_transport_list(peer->transports) \
                        ); \
                ret = 1; \
        } else if (peer->socket.type & SIP_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, get_transport(tmpl->socket.type) \
                ); \
        } else { \
                ast_debug(1, \
                        "peer '%s' has contacted us over %s even though we prefer %s.\n", \
                        peer->name, get_transport(tmpl->socket.type), 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_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 ? SIP_TRANSPORT_TLS: SIP_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;
        }

        ast_mutex_lock(&tcptls_session->lock);

        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);

        ast_mutex_unlock(&tcptls_session->lock);
        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");
        }
        ast_mutex_unlock(&tcptls_session->lock);

        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(NULL, tcptls_session);
}

/*! \brief SIP TCP thread management function
        This function reads from the socket, parses the packet into a request
*/
static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *tcptls_session)
{
        int res, cl;
        struct sip_request req = { 0, } , reqcpy = { 0, };
        struct sip_threadinfo *me = NULL;
        char buf[1024] = "";
        struct pollfd fds[2] = { { 0 }, { 0 }, };
        struct ast_tcptls_session_args *ca = NULL;

        /* If this is a server session, then the connection has already been setup,
         * simply create the threadinfo object so we can access this thread for writing.
         * 
         * if this is a client connection more work must be done.
         * 1. We own the parent session args for a client connection.  This pointer needs
         *    to be held on to so we can decrement it's ref count on thread destruction.
         * 2. The threadinfo object was created before this thread was launched, however
         *    it must be found within the threadt table.
         * 3. Last, the tcptls_session must be started.
         */
        if (!tcptls_session->client) {
                if (!(me = sip_threadinfo_create(tcptls_session, tcptls_session->ssl ? SIP_TRANSPORT_TLS : SIP_TRANSPORT_TCP))) {
                        goto cleanup;
                }
                ao2_t_ref(me, +1, "Adding threadinfo ref for tcp_helper_thread");
        } else {
                struct sip_threadinfo tmp = {
                        .tcptls_session = tcptls_session,
                };

                if ((!(ca = tcptls_session->parent)) ||
                        (!(me = ao2_t_find(threadt, &tmp, OBJ_POINTER, "ao2_find, getting sip_threadinfo in tcp helper thread"))) ||
                        (!(tcptls_session = ast_tcptls_client_start(tcptls_session)))) {
                        goto cleanup;
                }
        }

        me->threadid = pthread_self();
        ast_debug(2, "Starting thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");

        /* set up pollfd to watch for reads on both the socket and the alert_pipe */
        fds[0].fd = tcptls_session->fd;
        fds[1].fd = me->alert_pipe[0];
        fds[0].events = fds[1].events = POLLIN | POLLPRI;

        if (!(req.data = ast_str_create(SIP_MIN_PACKET)))
                goto cleanup;
        if (!(reqcpy.data = ast_str_create(SIP_MIN_PACKET)))
                goto cleanup;

        for (;;) {
                struct ast_str *str_save;

                res = ast_poll(fds, 2, -1); /* polls for both socket and alert_pipe */
                if (res < 0) {
                        ast_debug(2, "SIP %s server :: ast_wait_for_input returned %d\n", tcptls_session->ssl ? "SSL": "TCP", res);
                        goto cleanup;
                }

                /* handle the socket event, check for both reads from the socket fd,
                 * and writes from alert_pipe fd */
                if (fds[0].revents) { /* there is data on the socket to be read */

                        fds[0].revents = 0;

                        /* clear request structure */
                        str_save = req.data;
                        memset(&req, 0, sizeof(req));
                        req.data = str_save;
                        ast_str_reset(req.data);

                        str_save = reqcpy.data;
                        memset(&reqcpy, 0, sizeof(reqcpy));
                        reqcpy.data = str_save;
                        ast_str_reset(reqcpy.data);

                        memset(buf, 0, sizeof(buf));

                        if (tcptls_session->ssl) {
                                set_socket_transport(&req.socket, SIP_TRANSPORT_TLS);
                                req.socket.port = htons(ourport_tls);
                        } else {
                                set_socket_transport(&req.socket, SIP_TRANSPORT_TCP);
                                req.socket.port = htons(ourport_tcp);
                        }
                        req.socket.fd = tcptls_session->fd;

                        /* Read in headers one line at a time */
                        while (req.len < 4 || strncmp(REQ_OFFSET_TO_STR(&req, len - 4), "\r\n\r\n", 4)) {
                                ast_mutex_lock(&tcptls_session->lock);
                                if (!fgets(buf, sizeof(buf), tcptls_session->f)) {
                                        ast_mutex_unlock(&tcptls_session->lock);
                                        goto cleanup;
                                }
                                ast_mutex_unlock(&tcptls_session->lock);
                                if (me->stop)
                                         goto cleanup;
                                ast_str_append(&req.data, 0, "%s", buf);
                                req.len = req.data->used;
                        }
                        copy_request(&reqcpy, &req);
                        parse_request(&reqcpy);
                        /* In order to know how much to read, we need the content-length header */
                        if (sscanf(get_header(&reqcpy, "Content-Length"), "%30d", &cl)) {
                                while (cl > 0) {
                                        size_t bytes_read;
                                        ast_mutex_lock(&tcptls_session->lock);
                                        if (!(bytes_read = fread(buf, 1, MIN(sizeof(buf) - 1, cl), tcptls_session->f))) {
                                                ast_mutex_unlock(&tcptls_session->lock);
                                                goto cleanup;
                                        }
                                        buf[bytes_read] = '\0';
                                        ast_mutex_unlock(&tcptls_session->lock);
                                        if (me->stop)
                                                goto cleanup;
                                        cl -= strlen(buf);
                                        ast_str_append(&req.data, 0, "%s", buf);
                                        req.len = req.data->used;
                                }
                        }
                        /*! \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 */

                        req.socket.tcptls_session = tcptls_session;
                        handle_request_do(&req, &tcptls_session->remote_address);
                }

                if (fds[1].revents) { /* alert_pipe indicates there is data in the send queue to be sent */
                        enum sip_tcptls_alert alert;
                        struct tcptls_packet *packet;

                        fds[1].revents = 0;

                        if (read(me->alert_pipe[0], &alert, sizeof(alert)) == -1) {
                                ast_log(LOG_ERROR, "read() failed: %s\n", strerror(errno));
                                continue;
                        }

                        switch (alert) {
                        case TCPTLS_ALERT_STOP:
                                goto cleanup;
                        case TCPTLS_ALERT_DATA:
                                ao2_lock(me);
                                if (!(packet = AST_LIST_REMOVE_HEAD(&me->packet_q, entry))) {
                                        ast_log(LOG_WARNING, "TCPTLS thread alert_pipe indicated packet should be sent, but frame_q is empty");
                                } else if (ast_tcptls_server_write(tcptls_session, ast_str_buffer(packet->data), packet->len) == -1) {
                                        ast_log(LOG_WARNING, "Failure to write to tcp/tls socket\n");
                                }

                                if (packet) {
                                        ao2_t_ref(packet, -1, "tcptls packet sent, this is no longer needed");
                                }
                                ao2_unlock(me);
                                break;
                        default:
                                ast_log(LOG_ERROR, "Unknown tcptls thread alert '%d'\n", alert);
                        }
                }
        }

        ast_debug(2, "Shutting down thread for %s server\n", tcptls_session->ssl ? "SSL" : "TCP");

cleanup:
        if (me) {
                ao2_t_unlink(threadt, me, "Removing tcptls helper thread, thread is closing");
                ao2_t_ref(me, -1, "Removing tcp_helper_threads threadinfo ref");
        }
        if (reqcpy.data) {
                ast_free(reqcpy.data);
        }

        if (req.data) {
                ast_free(req.data);
                req.data = NULL;
        }

        /* if client, we own the parent session arguments and must decrement ref */
        if (ca) {
                ao2_t_ref(ca, -1, "closing tcptls thread, getting rid of client tcptls_session arguments");
        }

        if (tcptls_session) {
                ast_mutex_lock(&tcptls_session->lock);
                if (tcptls_session->f) {
                        fclose(tcptls_session->f);
                        tcptls_session->f = NULL;
                }
                if (tcptls_session->fd != -1) {
                        close(tcptls_session->fd);
                        tcptls_session->fd = -1;
                }
                tcptls_session->parent = NULL;
                ast_mutex_unlock(&tcptls_session->lock);

                ao2_ref(tcptls_session, -1);
                tcptls_session = NULL;
        }
        return NULL;
}


/*!
 * helper functions to unreference various types of objects.
 * By handling them this way, we don't have to declare the
 * destructor on each call, which removes the chance of errors.
 */
static void *unref_peer(struct sip_peer *peer, char *tag)
{
        ao2_t_ref(peer, -1, tag);
        return NULL;
}

static struct sip_peer *ref_peer(struct sip_peer *peer, char *tag)
{
        ao2_t_ref(peer, 1, tag);
        return peer;
}

/*! \brief maintain proper refcounts for a sip_pvt's outboundproxy
 *
 * This function sets pvt's outboundproxy pointer to the one referenced
 * by the proxy parameter. Because proxy may be a refcounted object, and
 * because pvt's old outboundproxy may also be a refcounted object, we need
 * to maintain the proper refcounts.
 *
 * \param pvt The sip_pvt for which we wish to set the outboundproxy
 * \param proxy The sip_proxy which we will point pvt towards.
 * \return Returns void
 */
static void ref_proxy(struct sip_pvt *pvt, struct sip_proxy *proxy)
{
        struct sip_proxy *old_obproxy = pvt->outboundproxy;
        /* The sip_cfg.outboundproxy is statically allocated, and so
         * we don't ever need to adjust refcounts for it
         */
        if (proxy && proxy != &sip_cfg.outboundproxy) {
                ao2_ref(proxy, +1);
        }
        pvt->outboundproxy = proxy;
        if (old_obproxy && old_obproxy != &sip_cfg.outboundproxy) {
                ao2_ref(old_obproxy, -1);
        }
}

/*!
 * \brief Unlink a dialog from the dialogs container, as well as any other places
 * that it may be currently stored.
 *
 * \note A reference to the dialog must be held before calling this function, and this
 * function does not release that reference.
 */
void *dialog_unlink_all(struct sip_pvt *dialog, int lockowner, int lockdialoglist)
{
        struct sip_pkt *cp;

        dialog_ref(dialog, "Let's bump the count in the unlink so it doesn't accidentally become dead before we are done");

        ao2_t_unlink(dialogs, dialog, "unlinking dialog via ao2_unlink");

        /* Unlink us from the owner (channel) if we have one */
        if (dialog->owner) {
                if (lockowner)
                        ast_channel_lock(dialog->owner);
                ast_debug(1, "Detaching from channel %s\n", dialog->owner->name);
                dialog->owner->tech_pvt = dialog_unref(dialog->owner->tech_pvt, "resetting channel dialog ptr in unlink_all");
                if (lockowner)
                        ast_channel_unlock(dialog->owner);
        }
        if (dialog->registry) {
                if (dialog->registry->call == dialog)
                        dialog->registry->call = dialog_unref(dialog->registry->call, "nulling out the registry's call dialog field in unlink_all");
                dialog->registry = registry_unref(dialog->registry, "delete dialog->registry");
        }
        if (dialog->stateid > -1) {
                ast_extension_state_del(dialog->stateid, NULL);
                dialog_unref(dialog, "removing extension_state, should unref the associated dialog ptr that was stored there.");
                dialog->stateid = -1; /* shouldn't we 'zero' this out? */
        }
        /* Remove link from peer to subscription of MWI */
        if (dialog->relatedpeer && dialog->relatedpeer->mwipvt == dialog)
                dialog->relatedpeer->mwipvt = dialog_unref(dialog->relatedpeer->mwipvt, "delete ->relatedpeer->mwipvt");
        if (dialog->relatedpeer && dialog->relatedpeer->call == dialog)
                dialog->relatedpeer->call = dialog_unref(dialog->relatedpeer->call, "unset the relatedpeer->call field in tandem with relatedpeer field itself");

        /* remove all current packets in this dialog */
        while((cp = dialog->packets)) {
                dialog->packets = dialog->packets->next;
                AST_SCHED_DEL(sched, cp->retransid);
                dialog_unref(cp->owner, "remove all current packets in this dialog, and the pointer to the dialog too as part of __sip_destroy");
                if (cp->data) {
                        ast_free(cp->data);
                }
                ast_free(cp);
        }

        AST_SCHED_DEL_UNREF(sched, dialog->waitid, dialog_unref(dialog, "when you delete the waitid sched, you should dec the refcount for the stored dialog ptr"));

        AST_SCHED_DEL_UNREF(sched, dialog->initid, dialog_unref(dialog, "when you delete the initid sched, you should dec the refcount for the stored dialog ptr"));
        
        if (dialog->autokillid > -1)
                AST_SCHED_DEL_UNREF(sched, dialog->autokillid, dialog_unref(dialog, "when you delete the autokillid sched, you should dec the refcount for the stored dialog ptr"));

        if (dialog->request_queue_sched_id > -1) {
                AST_SCHED_DEL_UNREF(sched, dialog->request_queue_sched_id, dialog_unref(dialog, "when you delete the request_queue_sched_id sched, you should dec the refcount for the stored dialog ptr"));
        }

        AST_SCHED_DEL_UNREF(sched, dialog->provisional_keepalive_sched_id, dialog_unref(dialog, "when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr"));

        if (dialog->t38id > -1) {
                AST_SCHED_DEL_UNREF(sched, dialog->t38id, dialog_unref(dialog, "when you delete the t38id sched, you should dec the refcount for the stored dialog ptr"));
        }

        dialog_unref(dialog, "Let's unbump the count in the unlink so the poor pvt can disappear if it is time");
        return NULL;
}

void *registry_unref(struct sip_registry *reg, char *tag)
{
        ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1);
        ASTOBJ_UNREF(reg, sip_registry_destroy);
        return NULL;
}

/*! \brief Add object reference to SIP registry */
static struct sip_registry *registry_addref(struct sip_registry *reg, char *tag)
{
        ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount + 1);
        return ASTOBJ_REF(reg); /* Add pointer to registry in packet */
}

/*! \brief Interface structure with callbacks used to connect to UDPTL module*/
static struct ast_udptl_protocol sip_udptl = {
        type: "SIP",
        get_udptl_info: sip_get_udptl_peer,
        set_udptl_peer: sip_set_udptl_peer,
};

static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
        __attribute__((format(printf, 2, 3)));


/*! \brief Convert transfer status to string */
static const char *referstatus2str(enum referstatus rstatus)
{
        return map_x_s(referstatusstrings, rstatus, "");
}

static inline void pvt_set_needdestroy(struct sip_pvt *pvt, const char *reason)
{
        append_history(pvt, "NeedDestroy", "Setting needdestroy because %s", reason);
        pvt->needdestroy = 1;
}

/*! \brief Initialize the initital request packet in the pvt structure.
        This packet is used for creating replies and future requests in
        a dialog */
static void initialize_initreq(struct sip_pvt *p, struct sip_request *req)
{
        if (p->initreq.headers)
                ast_debug(1, "Initializing already initialized SIP dialog %s (presumably reinvite)\n", p->callid);
        else
                ast_debug(1, "Initializing initreq for method %s - callid %s\n", sip_methods[req->method].text, p->callid);
        /* Use this as the basis */
        copy_request(&p->initreq, req);
        parse_request(&p->initreq);
        if (req->debug)
                ast_verbose("Initreq: %d headers, %d lines\n", p->initreq.headers, p->initreq.lines);
}

/*! \brief Encapsulate setting of SIP_ALREADYGONE to be able to trace it with debugging */
static void sip_alreadygone(struct sip_pvt *dialog)
{
        ast_debug(3, "Setting SIP_ALREADYGONE on dialog %s\n", dialog->callid);
        dialog->alreadygone = 1;
}

/*! Resolve DNS srv name or host name in a sip_proxy structure */
static int proxy_update(struct sip_proxy *proxy)
{
        /* if it's actually an IP address and not a name,
           there's no need for a managed lookup */
        if (!inet_aton(proxy->name, &proxy->ip.sin_addr)) {
                /* Ok, not an IP address, then let's check if it's a domain or host */
                /* XXX Todo - if we have proxy port, don't do SRV */
                if (ast_get_ip_or_srv(&proxy->ip, proxy->name, sip_cfg.srvlookup ? "_sip._udp" : NULL) < 0) {
                        ast_log(LOG_WARNING, "Unable to locate host '%s'\n", proxy->name);
                        return FALSE;
                }
        }
        proxy->last_dnsupdate = time(NULL);
        return TRUE;
}

/*! \brief converts ascii port to int representation. If no
 *  pt buffer is provided or the pt has errors when being converted
 *  to an int value, the port provided as the standard is used.
 */
unsigned int port_str2int(const char *pt, unsigned int standard)
{
        int port = standard;
        if (ast_strlen_zero(pt) || (sscanf(pt, "%30d", &port) != 1) || (port < 1) || (port > 65535)) {
                port = standard;
        }

        return port;
}

/*! \brief Allocate and initialize sip proxy */
static struct sip_proxy *proxy_allocate(char *name, char *port, int force)
{
        struct sip_proxy *proxy;

        if (ast_strlen_zero(name)) {
                return NULL;
        }

        proxy = ao2_alloc(sizeof(*proxy), NULL);
        if (!proxy)
                return NULL;
        proxy->force = force;
        ast_copy_string(proxy->name, name, sizeof(proxy->name));
        proxy->ip.sin_port = htons(port_str2int(port, STANDARD_SIP_PORT));
        proxy_update(proxy);
        return proxy;
}

/*! \brief Get default outbound proxy or global proxy */
static struct sip_proxy *obproxy_get(struct sip_pvt *dialog, struct sip_peer *peer)
{
        if (peer && peer->outboundproxy) {
                if (sipdebug)
                        ast_debug(1, "OBPROXY: Applying peer OBproxy to this call\n");
                append_history(dialog, "OBproxy", "Using peer obproxy %s", peer->outboundproxy->name);
                return peer->outboundproxy;
        }
        if (sip_cfg.outboundproxy.name[0]) {
                if (sipdebug)
                        ast_debug(1, "OBPROXY: Applying global OBproxy to this call\n");
                append_history(dialog, "OBproxy", "Using global obproxy %s", sip_cfg.outboundproxy.name);
                return &sip_cfg.outboundproxy;
        }
        if (sipdebug)
                ast_debug(1, "OBPROXY: Not applying OBproxy to this call\n");
        return NULL;
}

/*! \brief returns true if 'name' (with optional trailing whitespace)
 * matches the sip method 'id'.
 * Strictly speaking, SIP methods are case SENSITIVE, but we do
 * a case-insensitive comparison to be more tolerant.
 * following Jon Postel's rule: Be gentle in what you accept, strict with what you send
 */
static int method_match(enum sipmethod id, const char *name)
{
        int len = strlen(sip_methods[id].text);
        int l_name = name ? strlen(name) : 0;
        /* true if the string is long enough, and ends with whitespace, and matches */
        return (l_name >= len && name[len] < 33 &&
                !strncasecmp(sip_methods[id].text, name, len));
}

/*! \brief  find_sip_method: Find SIP method from header */
static int find_sip_method(const char *msg)
{
        int i, res = 0;
        
        if (ast_strlen_zero(msg))
                return 0;
        for (i = 1; i < ARRAY_LEN(sip_methods) && !res; i++) {
                if (method_match(i, msg))
                        res = sip_methods[i].id;
        }
        return res;
}

/*! \brief Parse supported header in incoming packet */
static unsigned int parse_sip_options(struct sip_pvt *pvt, const char *supported)
{
        char *next, *sep;
        char *temp;
        unsigned int profile = 0;
        int i, found;

        if (ast_strlen_zero(supported) )
                return 0;
        temp = ast_strdupa(supported);

        if (sipdebug)
                ast_debug(3, "Begin: parsing SIP \"Supported: %s\"\n", supported);

        for (next = temp; next; next = sep) {
                found = FALSE;
                if ( (sep = strchr(next, ',')) != NULL)
                        *sep++ = '\0';
                next = ast_skip_blanks(next);
                if (sipdebug)
                        ast_debug(3, "Found SIP option: -%s-\n", next);
                for (i = 0; i < ARRAY_LEN(sip_options); i++) {
                        if (!strcasecmp(next, sip_options[i].text)) {
                                profile |= sip_options[i].id;
                                found = TRUE;
                                if (sipdebug)
                                        ast_debug(3, "Matched SIP option: %s\n", next);
                                break;
                        }
                }

                /* This function is used to parse both Suported: and Require: headers.
                Let the caller of this function know that an unknown option tag was
                encountered, so that if the UAC requires it then the request can be
                rejected with a 420 response. */
                if (!found)
                        profile |= SIP_OPT_UNKNOWN;

                if (!found && sipdebug) {
                        if (!strncasecmp(next, "x-", 2))
                                ast_debug(3, "Found private SIP option, not supported: %s\n", next);
                        else
                                ast_debug(3, "Found no match for SIP option: %s (Please file bug report!)\n", next);
                }
        }

        if (pvt)
                pvt->sipoptions = profile;
        return profile;
}

/*! \brief See if we pass debug IP filter */
static inline int sip_debug_test_addr(const struct sockaddr_in *addr)
{
        if (!sipdebug)
                return 0;
        if (debugaddr.sin_addr.s_addr) {
                if (((ntohs(debugaddr.sin_port) != 0)
                        && (debugaddr.sin_port != addr->sin_port))
                        || (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
                        return 0;
        }
        return 1;
}

/*! \brief The real destination address for a write */
static const struct sockaddr_in *sip_real_dst(const struct sip_pvt *p)
{
        if (p->outboundproxy)
                return &p->outboundproxy->ip;

        return ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT) || ast_test_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT) ? &p->recv : &p->sa;
}

/*! \brief Display SIP nat mode */
static const char *sip_nat_mode(const struct sip_pvt *p)
{
        return ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT) ? "NAT" : "no NAT";
}

/*! \brief Test PVT for debugging output */
static inline int sip_debug_test_pvt(struct sip_pvt *p)
{
        if (!sipdebug)
                return 0;
        return sip_debug_test_addr(sip_real_dst(p));
}

/*! \brief Return int representing a bit field of transport types found in const char *transport */
static int get_transport_str2enum(const char *transport)
{
        int res = 0;

        if (ast_strlen_zero(transport)) {
                return res;
        }

        if (!strcasecmp(transport, "udp")) {
                res |= SIP_TRANSPORT_UDP;
        }
        if (!strcasecmp(transport, "tcp")) {
                res |= SIP_TRANSPORT_TCP;
        }
        if (!strcasecmp(transport, "tls")) {
                res |= SIP_TRANSPORT_TLS;
        }

        return res;
}

/*! \brief Return configuration of transports for a device */
static inline const char *get_transport_list(unsigned int transports) {
        switch (transports) {
                case SIP_TRANSPORT_UDP:
                        return "UDP";
                case SIP_TRANSPORT_TCP:
                        return "TCP";
                case SIP_TRANSPORT_TLS:
                        return "TLS";
                case SIP_TRANSPORT_UDP | SIP_TRANSPORT_TCP:
                        return "TCP,UDP";
                case SIP_TRANSPORT_UDP | SIP_TRANSPORT_TLS:
                        return "TLS,UDP";
                case SIP_TRANSPORT_TCP | SIP_TRANSPORT_TLS:
                        return "TLS,TCP";
                default:
                        return transports ?
                                "TLS,TCP,UDP" : "UNKNOWN";      
        }
}

/*! \brief Return transport as string */
static inline const char *get_transport(enum sip_transport t)
{
        switch (t) {
        case SIP_TRANSPORT_UDP:
                return "UDP";
        case SIP_TRANSPORT_TCP:
                return "TCP";
        case SIP_TRANSPORT_TLS:
                return "TLS";
        }

        return "UNKNOWN";
}

/*! \brief Return transport of dialog.
        \note this is based on a false assumption. We don't always use the
        outbound proxy for all requests in a dialog. It depends on the
        "force" parameter. The FIRST request is always sent to the ob proxy.
        \todo Fix this function to work correctly
*/
static inline const char *get_transport_pvt(struct sip_pvt *p)
{
        if (p->outboundproxy && p->outboundproxy->transport) {
                set_socket_transport(&p->socket, p->outboundproxy->transport);
        }

        return get_transport(p->socket.type);
}

/*! \brief Transmit SIP message
        Sends a SIP request or response on a given socket (in the pvt)
        Called by retrans_pkt, send_request, send_response and
        __sip_reliable_xmit
        \return length of transmitted message, XMIT_ERROR on known network failures -1 on other failures.
*/
static int __sip_xmit(struct sip_pvt *p, struct ast_str *data, int len)
{
        int res = 0;
        const struct sockaddr_in *dst = sip_real_dst(p);

        ast_debug(2, "Trying to put '%.11s' onto %s socket destined for %s:%d\n", data->str, get_transport_pvt(p), ast_inet_ntoa(dst->sin_addr), htons(dst->sin_port));

        if (sip_prepare_socket(p) < 0)
                return XMIT_ERROR;

        if (p->socket.type == SIP_TRANSPORT_UDP) {
                res = sendto(p->socket.fd, data->str, len, 0, (const struct sockaddr *)dst, sizeof(struct sockaddr_in));
        } else if (p->socket.tcptls_session) {
                res = sip_tcptls_write(p->socket.tcptls_session, data->str, len);
        } else {
                ast_debug(2, "Socket type is TCP but no tcptls_session is present to write to\n");
                return XMIT_ERROR;
        }

        if (res == -1) {
                switch (errno) {
                case EBADF:             /* Bad file descriptor - seems like this is generated when the host exist, but doesn't accept the UDP packet */
                case EHOSTUNREACH:      /* Host can't be reached */
                case ENETDOWN:          /* Interface down */
                case ENETUNREACH:       /* Network failure */
                case ECONNREFUSED:      /* ICMP port unreachable */
                        res = XMIT_ERROR;       /* Don't bother with trying to transmit again */
                }
        }
        if (res != len)
                ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s:%d returned %d: %s\n", data, len, ast_inet_ntoa(dst->sin_addr), ntohs(dst->sin_port), res, strerror(errno));

        return res;
}

/*! \brief Build a Via header for a request */
static void build_via(struct sip_pvt *p)
{
        /* Work around buggy UNIDEN UIP200 firmware */
        const char *rport = (ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT) || ast_test_flag(&p->flags[0], SIP_NAT_RPORT_PRESENT)) ? ";rport" : "";

        /* z9hG4bK is a magic cookie.  See RFC 3261 section 8.1.1.7 */
        snprintf(p->via, sizeof(p->via), "SIP/2.0/%s %s:%d;branch=z9hG4bK%08x%s",
                 get_transport_pvt(p),
                 ast_inet_ntoa(p->ourip.sin_addr),
                 ntohs(p->ourip.sin_port), (int) p->branch, rport);
}

/*! \brief NAT fix - decide which IP address to use for Asterisk server?
 *
 * Using the localaddr structure built up with localnet statements in sip.conf
 * apply it to their address to see if we need to substitute our
 * externip or can get away with our internal bindaddr
 * 'us' is always overwritten.
 */
static void ast_sip_ouraddrfor(struct in_addr *them, struct sockaddr_in *us, struct sip_pvt *p)
{
        struct sockaddr_in theirs;
        /* Set want_remap to non-zero if we want to remap 'us' to an externally
         * reachable IP address and port. This is done if:
         * 1. we have a localaddr list (containing 'internal' addresses marked
         *    as 'deny', so ast_apply_ha() will return AST_SENSE_DENY on them,
         *    and AST_SENSE_ALLOW on 'external' ones);
         * 2. either stunaddr or externip is set, so we know what to use as the
         *    externally visible address;
         * 3. the remote address, 'them', is external;
         * 4. the address returned by ast_ouraddrfor() is 'internal' (AST_SENSE_DENY
         *    when passed to ast_apply_ha() so it does need to be remapped.
         *    This fourth condition is checked later.
         */
        int want_remap;

        *us = internip;         /* starting guess for the internal address */
        /* now ask the system what would it use to talk to 'them' */
        ast_ouraddrfor(them, &us->sin_addr);
        theirs.sin_addr = *them;

        want_remap = localaddr &&
                (externip.sin_addr.s_addr || stunaddr.sin_addr.s_addr) &&
                ast_apply_ha(localaddr, &theirs) == AST_SENSE_ALLOW ;

        if (want_remap &&
            (!sip_cfg.matchexterniplocally || !ast_apply_ha(localaddr, us)) ) {
                /* if we used externhost or stun, see if it is time to refresh the info */
                if (externexpire && time(NULL) >= externexpire) {
                        if (stunaddr.sin_addr.s_addr) {
                                ast_stun_request(sipsock, &stunaddr, NULL, &externip);
                        } else {
                                if (ast_parse_arg(externhost, PARSE_INADDR, &externip))
                                        ast_log(LOG_NOTICE, "Warning: Re-lookup of '%s' failed!\n", externhost);
                        }
                        externexpire = time(NULL) + externrefresh;
                }
                if (externip.sin_addr.s_addr) {
                        *us = externip;
                        switch (p->socket.type) {
                        case SIP_TRANSPORT_TCP:
                                us->sin_port = htons(externtcpport);
                                break;
                        case SIP_TRANSPORT_TLS:
                                us->sin_port = htons(externtlsport);
                                break;
                        case SIP_TRANSPORT_UDP:
                                break; /* fall through */
                        default:
                                us->sin_port = htons(STANDARD_SIP_PORT); /* we should never get here */
                        }
                }
                else
                        ast_log(LOG_WARNING, "stun failed\n");
                ast_debug(1, "Target address %s is not local, substituting externip\n",
                        ast_inet_ntoa(*(struct in_addr *)&them->s_addr));
        } else if (p) {
                /* no remapping, but we bind to a specific address, so use it. */
                switch (p->socket.type) {
                case SIP_TRANSPORT_TCP:
                        if (sip_tcp_desc.local_address.sin_addr.s_addr) {
                                *us = sip_tcp_desc.local_address;
                        } else {
                                us->sin_port = sip_tcp_desc.local_address.sin_port;
                        }
                        break;
                case SIP_TRANSPORT_TLS:
                        if (sip_tls_desc.local_address.sin_addr.s_addr) {
                                *us = sip_tls_desc.local_address;
                        } else {
                                us->sin_port = sip_tls_desc.local_address.sin_port;
                        }
                                break;
                case SIP_TRANSPORT_UDP:
                        /* fall through on purpose */
                default:
                        if (bindaddr.sin_addr.s_addr) {
                                *us = bindaddr;
                        }
                }
        } else if (bindaddr.sin_addr.s_addr) {
                *us = bindaddr;
        }
        ast_debug(3, "Setting SIP_TRANSPORT_%s with address %s:%d\n", get_transport(p->socket.type), ast_inet_ntoa(us->sin_addr), ntohs(us->sin_port));
}

/*! \brief Append to SIP dialog history with arg list  */
static __attribute__((format(printf, 2, 0))) void append_history_va(struct sip_pvt *p, const char *fmt, va_list ap)
{
        char buf[80], *c = buf; /* max history length */
        struct sip_history *hist;
        int l;

        vsnprintf(buf, sizeof(buf), fmt, ap);
        strsep(&c, "\r\n"); /* Trim up everything after \r or \n */
        l = strlen(buf) + 1;
        if (!(hist = ast_calloc(1, sizeof(*hist) + l)))
                return;
        if (!p->history && !(p->history = ast_calloc(1, sizeof(*p->history)))) {
                ast_free(hist);
                return;
        }
        memcpy(hist->event, buf, l);
        if (p->history_entries == MAX_HISTORY_ENTRIES) {
                struct sip_history *oldest;
                oldest = AST_LIST_REMOVE_HEAD(p->history, list);
                p->history_entries--;
                ast_free(oldest);
        }
        AST_LIST_INSERT_TAIL(p->history, hist, list);
        p->history_entries++;
}

/*! \brief Append to SIP dialog history with arg list  */
static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
{
        va_list ap;

        if (!p)
                return;

        if (!p->do_history && !recordhistory && !dumphistory)
                return;

        va_start(ap, fmt);
        append_history_va(p, fmt, ap);
        va_end(ap);

        return;
}

/*! \brief Retransmit SIP message if no answer (Called from scheduler) */
static int retrans_pkt(const void *data)
{
        struct sip_pkt *pkt = (struct sip_pkt *)data, *prev, *cur = NULL;
        int reschedule = DEFAULT_RETRANS;
        int xmitres = 0;
        
        /* Lock channel PVT */
        sip_pvt_lock(pkt->owner);

        if (pkt->retrans < MAX_RETRANS) {
                pkt->retrans++;
                if (!pkt->timer_t1) {   /* Re-schedule using timer_a and timer_t1 */
                        if (sipdebug)
                                ast_debug(4, "SIP TIMER: Not rescheduling id #%d:%s (Method %d) (No timer T1)\n", pkt->retransid, sip_methods[pkt->method].text, pkt->method);
                } else {
                        int siptimer_a;

                        if (sipdebug)
                                ast_debug(4, "SIP TIMER: Rescheduling retransmission #%d (%d) %s - %d\n", pkt->retransid, pkt->retrans, sip_methods[pkt->method].text, pkt->method);
                        if (!pkt->timer_a)
                                pkt->timer_a = 2 ;
                        else
                                pkt->timer_a = 2 * pkt->timer_a;

                        /* For non-invites, a maximum of 4 secs */
                        siptimer_a = pkt->timer_t1 * pkt->timer_a;      /* Double each time */
                        if (pkt->method != SIP_INVITE && siptimer_a > 4000)
                                siptimer_a = 4000;
                
                        /* Reschedule re-transmit */
                        reschedule = siptimer_a;
                        ast_debug(4, "** SIP timers: Rescheduling retransmission %d to %d ms (t1 %d ms (Retrans id #%d)) \n", pkt->retrans +1, siptimer_a, pkt->timer_t1, pkt->retransid);
                }

                if (sip_debug_test_pvt(pkt->owner)) {
                        const struct sockaddr_in *dst = sip_real_dst(pkt->owner);
                        ast_verbose("Retransmitting #%d (%s) to %s:%d:\n%s\n---\n",
                                pkt->retrans, sip_nat_mode(pkt->owner),
                                ast_inet_ntoa(dst->sin_addr),
                                ntohs(dst->sin_port), pkt->data->str);
                }

                append_history(pkt->owner, "ReTx", "%d %s", reschedule, pkt->data->str);
                xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
                sip_pvt_unlock(pkt->owner);
                if (xmitres == XMIT_ERROR)
                        ast_log(LOG_WARNING, "Network error on retransmit in dialog %s\n", pkt->owner->callid);
                else
                        return  reschedule;
        }
        /* Too many retries */
        if (pkt->owner && pkt->method != SIP_OPTIONS && xmitres == 0) {
                if (pkt->is_fatal || sipdebug)  /* Tell us if it's critical or if we're debugging */
                        ast_log(LOG_WARNING, "Maximum retries exceeded on transmission %s for seqno %d (%s %s) -- See doc/sip-retransmit.txt.\n",
                                pkt->owner->callid, pkt->seqno,
                                pkt->is_fatal ? "Critical" : "Non-critical", pkt->is_resp ? "Response" : "Request");
        } else if (pkt->method == SIP_OPTIONS && sipdebug) {
                        ast_log(LOG_WARNING, "Cancelling retransmit of OPTIONs (call id %s)  -- See doc/sip-retransmit.txt.\n", pkt->owner->callid);

        }
        if (xmitres == XMIT_ERROR) {
                ast_log(LOG_WARNING, "Transmit error :: Cancelling transmission on Call ID %s\n", pkt->owner->callid);
                append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
        } else
                append_history(pkt->owner, "MaxRetries", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
                
        pkt->retransid = -1;

        if (pkt->is_fatal) {
                while(pkt->owner->owner && ast_channel_trylock(pkt->owner->owner)) {
                        sip_pvt_unlock(pkt->owner);     /* SIP_PVT, not channel */
                        usleep(1);
                        sip_pvt_lock(pkt->owner);
                }

                if (pkt->owner->owner && !pkt->owner->owner->hangupcause)
                        pkt->owner->owner->hangupcause = AST_CAUSE_NO_USER_RESPONSE;
                
                if (pkt->owner->owner) {
                        sip_alreadygone(pkt->owner);
                        ast_log(LOG_WARNING, "Hanging up call %s - no reply to our critical packet (see doc/sip-retransmit.txt).\n", pkt->owner->callid);
                        ast_queue_hangup_with_cause(pkt->owner->owner, AST_CAUSE_PROTOCOL_ERROR);
                        ast_channel_unlock(pkt->owner->owner);
                } else {
                        /* If no channel owner, destroy now */

                        /* Let the peerpoke system expire packets when the timer expires for poke_noanswer */
                        if (pkt->method != SIP_OPTIONS && pkt->method != SIP_REGISTER) {
                                pvt_set_needdestroy(pkt->owner, "no response to critical packet");
                                sip_alreadygone(pkt->owner);
                                append_history(pkt->owner, "DialogKill", "Killing this failed dialog immediately");
                        }
                }
        }

        if (pkt->method == SIP_BYE) {
                /* We're not getting answers on SIP BYE's.  Tear down the call anyway. */
                if (pkt->owner->owner)
                        ast_channel_unlock(pkt->owner->owner);
                append_history(pkt->owner, "ByeFailure", "Remote peer doesn't respond to bye. Destroying call anyway.");
                pvt_set_needdestroy(pkt->owner, "no response to BYE");
        }

        /* Remove the packet */
        for (prev = NULL, cur = pkt->owner->packets; cur; prev = cur, cur = cur->next) {
                if (cur == pkt) {
                        UNLINK(cur, pkt->owner->packets, prev);
                        sip_pvt_unlock(pkt->owner);
                        if (pkt->owner)
                                pkt->owner = dialog_unref(pkt->owner,"pkt is being freed, its dialog ref is dead now");
                        if (pkt->data)
                                ast_free(pkt->data);
                        pkt->data = NULL;
                        ast_free(pkt);
                        return 0;
                }
        }
        /* error case */
        ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
        sip_pvt_unlock(pkt->owner);
        return 0;
}

/*! \brief Transmit packet with retransmits
        \return 0 on success, -1 on failure to allocate packet
*/
static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, struct ast_str *data, int len, int fatal, int sipmethod)
{
        struct sip_pkt *pkt = NULL;
        int siptimer_a = DEFAULT_RETRANS;
        int xmitres = 0;
        int respid;

        if (sipmethod == SIP_INVITE) {
                /* Note this is a pending invite */
                p->pendinginvite = seqno;
        }

        /* If the transport is something reliable (TCP or TLS) then don't really send this reliably */
        /* I removed the code from retrans_pkt that does the same thing so it doesn't get loaded into the scheduler */
        /*! \todo According to the RFC some packets need to be retransmitted even if its TCP, so this needs to get revisited */
        if (!(p->socket.type & SIP_TRANSPORT_UDP)) {
                xmitres = __sip_xmit(p, data, len);     /* Send packet */
                if (xmitres == XMIT_ERROR) {    /* Serious network trouble, no need to try again */
                        append_history(p, "XmitErr", "%s", fatal ? "(Critical)" : "(Non-critical)");
                        return AST_FAILURE;
                } else {
                        return AST_SUCCESS;
                }
        }

        if (!(pkt = ast_calloc(1, sizeof(*pkt) + len + 1)))
                return AST_FAILURE;
        /* copy data, add a terminator and save length */
        if (!(pkt->data = ast_str_create(len))) {
                ast_free(pkt);
                return AST_FAILURE;
        }
        ast_str_set(&pkt->data, 0, "%s%s", data->str, "\0");
        pkt->packetlen = len;
        /* copy other parameters from the caller */
        pkt->method = sipmethod;
        pkt->seqno = seqno;
        pkt->is_resp = resp;
        pkt->is_fatal = fatal;
        pkt->owner = dialog_ref(p, "__sip_reliable_xmit: setting pkt->owner");
        pkt->next = p->packets;
        p->packets = pkt;       /* Add it to the queue */
        if (resp) {
                /* Parse out the response code */
                if (sscanf(ast_str_buffer(pkt->data), "SIP/2.0 %30u", &respid) == 1) {
                        pkt->response_code = respid;
                }
        }
        pkt->timer_t1 = p->timer_t1;    /* Set SIP timer T1 */
        pkt->retransid = -1;
        if (pkt->timer_t1)
                siptimer_a = pkt->timer_t1 * 2;

        /* Schedule retransmission */
        AST_SCHED_REPLACE_VARIABLE(pkt->retransid, sched, siptimer_a, retrans_pkt, pkt, 1);
        if (sipdebug)
                ast_debug(4, "*** SIP TIMER: Initializing retransmit timer on packet: Id  #%d\n", pkt->retransid);

        xmitres = __sip_xmit(pkt->owner, pkt->data, pkt->packetlen);    /* Send packet */

        if (xmitres == XMIT_ERROR) {    /* Serious network trouble, no need to try again */
                append_history(pkt->owner, "XmitErr", "%s", pkt->is_fatal ? "(Critical)" : "(Non-critical)");
                ast_log(LOG_ERROR, "Serious Network Trouble; __sip_xmit returns error for pkt data\n");
                AST_SCHED_DEL(sched, pkt->retransid);
                p->packets = pkt->next;
                pkt->owner = dialog_unref(pkt->owner,"pkt is being freed, its dialog ref is dead now");
                ast_free(pkt->data);
                ast_free(pkt);
                return AST_FAILURE;
        } else {
                return AST_SUCCESS;
        }
}

/*! \brief Kill a SIP dialog (called only by the scheduler)
 * The scheduler has a reference to this dialog when p->autokillid != -1,
 * and we are called using that reference. So if the event is not
 * rescheduled, we need to call dialog_unref().
 */
static int __sip_autodestruct(const void *data)
{
        struct sip_pvt *p = (struct sip_pvt *)data;

        /* If this is a subscription, tell the phone that we got a timeout */
        if (p->subscribed) {
                transmit_state_notify(p, AST_EXTENSION_DEACTIVATED, 1, TRUE);   /* Send last notification */
                p->subscribed = NONE;
                append_history(p, "Subscribestatus", "timeout");
                ast_debug(3, "Re-scheduled destruction of SIP subscription %s\n", p->callid ? p->callid : "<unknown>");
                return 10000;   /* Reschedule this destruction so that we know that it's gone */
        }

        /* If there are packets still waiting for delivery, delay the destruction */
        if (p->packets) {
                if (!p->needdestroy) {
                        char method_str[31];
                        ast_debug(3, "Re-scheduled destruction of SIP call %s\n", p->callid ? p->callid : "<unknown>");
                        append_history(p, "ReliableXmit", "timeout");
                        if (sscanf(p->lastmsg, "Tx: %30s", method_str) == 1 || sscanf(p->lastmsg, "Rx: %30s", method_str) == 1) {
                                if (method_match(SIP_CANCEL, method_str) || method_match(SIP_BYE, method_str)) {
                                        pvt_set_needdestroy(p, "autodestruct");
                                }
                        }
                        return 10000;
                } else {
                        /* They've had their chance to respond. Time to bail */
                        __sip_pretend_ack(p);
                }
        }

        if (p->subscribed == MWI_NOTIFICATION) {
                if (p->relatedpeer) {
                        p->relatedpeer = unref_peer(p->relatedpeer, "__sip_autodestruct: unref peer p->relatedpeer");   /* Remove link to peer. If it's realtime, make sure it's gone from memory) */
                }
        }

        /* Reset schedule ID */
        p->autokillid = -1;

        if (p->owner) {
                ast_log(LOG_WARNING, "Autodestruct on dialog '%s' with owner in place (Method: %s)\n", p->callid, sip_methods[p->method].text);
                ast_queue_hangup_with_cause(p->owner, AST_CAUSE_PROTOCOL_ERROR);
        } else if (p->refer && !p->alreadygone) {
                ast_debug(3, "Finally hanging up channel after transfer: %s\n", p->callid);
                transmit_request_with_auth(p, SIP_BYE, 0, XMIT_RELIABLE, 1);
                append_history(p, "ReferBYE", "Sending BYE on transferer call leg %s", p->callid);
                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
        } else {
                append_history(p, "AutoDestroy", "%s", p->callid);
                ast_debug(3, "Auto destroying SIP dialog '%s'\n", p->callid);
                dialog_unlink_all(p, TRUE, TRUE); /* once it's unlinked and unrefd everywhere, it'll be freed automagically */
                /* dialog_unref(p, "unref dialog-- no other matching conditions"); -- unlink all now should finish off the dialog's references and free it. */
                /* sip_destroy(p); */           /* Go ahead and destroy dialog. All attempts to recover is done */
                /* sip_destroy also absorbs the reference */
        }
        dialog_unref(p, "The ref to a dialog passed to this sched callback is going out of scope; unref it.");
        return 0;
}

/*! \brief Schedule destruction of SIP dialog */
void sip_scheddestroy(struct sip_pvt *p, int ms)
{
        if (ms < 0) {
                if (p->timer_t1 == 0) {
                        p->timer_t1 = global_t1;        /* Set timer T1 if not set (RFC 3261) */
                }
                if (p->timer_b == 0) {
                        p->timer_b = global_timer_b;  /* Set timer B if not set (RFC 3261) */
                }
                ms = p->timer_t1 * 64;
        }
        if (sip_debug_test_pvt(p))
                ast_verbose("Scheduling destruction of SIP dialog '%s' in %d ms (Method: %s)\n", p->callid, ms, sip_methods[p->method].text);
        if (sip_cancel_destroy(p))
                ast_log(LOG_WARNING, "Unable to cancel SIP destruction.  Expect bad things.\n");

        if (p->do_history)
                append_history(p, "SchedDestroy", "%d ms", ms);
        p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, dialog_ref(p, "setting ref as passing into ast_sched_add for __sip_autodestruct"));

        if (p->stimer && p->stimer->st_active == TRUE && p->stimer->st_schedid > 0)
                stop_session_timer(p);
}

/*! \brief Cancel destruction of SIP dialog.
 * Be careful as this also absorbs the reference - if you call it
 * from within the scheduler, this might be the last reference.
 */
int sip_cancel_destroy(struct sip_pvt *p)
{
        int res = 0;
        if (p->autokillid > -1) {
                int res3;

                if (!(res3 = ast_sched_del(sched, p->autokillid))) {
                        append_history(p, "CancelDestroy", "");
                        p->autokillid = -1;
                        dialog_unref(p, "dialog unrefd because autokillid is de-sched'd");
                }
        }
        return res;
}

/*! \brief Acknowledges receipt of a packet and stops retransmission
 * called with p locked*/
int __sip_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
{
        struct sip_pkt *cur, *prev = NULL;
        const char *msg = "Not Found";  /* used only for debugging */
        int res = FALSE;

        /* If we have an outbound proxy for this dialog, then delete it now since
          the rest of the requests in this dialog needs to follow the routing.
          If obforcing is set, we will keep the outbound proxy during the whole
          dialog, regardless of what the SIP rfc says
        */
        if (p->outboundproxy && !p->outboundproxy->force){
                ref_proxy(p, NULL);
        }

        for (cur = p->packets; cur; prev = cur, cur = cur->next) {
                if (cur->seqno != seqno || cur->is_resp != resp)
                        continue;
                if (cur->is_resp || cur->method == sipmethod) {
                        res = TRUE;
                        msg = "Found";
                        if (!resp && (seqno == p->pendinginvite)) {
                                ast_debug(1, "Acked pending invite %d\n", p->pendinginvite);
                                p->pendinginvite = 0;
                        }
                        if (cur->retransid > -1) {
                                if (sipdebug)
                                        ast_debug(4, "** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #%d\n", cur->retransid);
                        }
                        /* This odd section is designed to thwart a
                         * race condition in the packet scheduler. There are
                         * two conditions under which deleting the packet from the
                         * scheduler can fail.
                         *
                         * 1. The packet has been removed from the scheduler because retransmission
                         * is being attempted. The problem is that if the packet is currently attempting
                         * retransmission and we are at this point in the code, then that MUST mean
                         * that retrans_pkt is waiting on p's lock. Therefore we will relinquish the
                         * lock temporarily to allow retransmission.
                         *
                         * 2. The packet has reached its maximum number of retransmissions and has
                         * been permanently removed from the packet scheduler. If this is the case, then
                         * the packet's retransid will be set to -1. The atomicity of the setting and checking
                         * of the retransid to -1 is ensured since in both cases p's lock is held.
                         */
                        while (cur->retransid > -1 && ast_sched_del(sched, cur->retransid)) {
                                sip_pvt_unlock(p);
                                usleep(1);
                                sip_pvt_lock(p);
                        }
                        UNLINK(cur, p->packets, prev);
                        dialog_unref(cur->owner, "unref pkt cur->owner dialog from sip ack before freeing pkt");
                        if (cur->data)
                                ast_free(cur->data);
                        ast_free(cur);
                        break;
                }
        }
        ast_debug(1, "Stopping retransmission on '%s' of %s %d: Match %s\n",
                p->callid, resp ? "Response" : "Request", seqno, msg);
        return res;
}

/*! \brief Pretend to ack all packets
 * called with p locked */
void __sip_pretend_ack(struct sip_pvt *p)
{
        struct sip_pkt *cur = NULL;

        while (p->packets) {
                int method;
                if (cur == p->packets) {
                        ast_log(LOG_WARNING, "Have a packet that doesn't want to give up! %s\n", sip_methods[cur->method].text);
                        return;
                }
                cur = p->packets;
                method = (cur->method) ? cur->method : find_sip_method(cur->data->str);
                __sip_ack(p, cur->seqno, cur->is_resp, method);
        }
}

/*! \brief Acks receipt of packet, keep it around (used for provisional responses) */
int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, int sipmethod)
{
        struct sip_pkt *cur;
        int res = FALSE;

        for (cur = p->packets; cur; cur = cur->next) {
                if (cur->seqno == seqno && cur->is_resp == resp &&
                        (cur->is_resp || method_match(sipmethod, cur->data->str))) {
                        /* this is our baby */
                        if (cur->retransid > -1) {
                                if (sipdebug)
                                        ast_debug(4, "*** SIP TIMER: Cancelling retransmission #%d - %s (got response)\n", cur->retransid, sip_methods[sipmethod].text);
                        }
                        AST_SCHED_DEL(sched, cur->retransid);
                        res = TRUE;
                        break;
                }
        }
        ast_debug(1, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res == -1 ? "Not Found" : "Found");
        return res;
}


/*! \brief Copy SIP request, parse it */
static void parse_copy(struct sip_request *dst, const struct sip_request *src)
{
        copy_request(dst, src);
        parse_request(dst);
}

/*! \brief add a blank line if no body */
static void add_blank(struct sip_request *req)
{
        if (!req->lines) {
                /* Add extra empty return. add_header() reserves 4 bytes so cannot be truncated */
                ast_str_append(&req->data, 0, "\r\n");
                req->len = ast_str_strlen(req->data);
        }
}

static int send_provisional_keepalive_full(struct sip_pvt *pvt, int with_sdp)
{
        const char *msg = NULL;

        if (!pvt->last_provisional || !strncasecmp(pvt->last_provisional, "100", 3)) {
                msg = "183 Session Progress";
        }

        if (pvt->invitestate < INV_COMPLETED) {
                if (with_sdp) {
                        transmit_response_with_sdp(pvt, S_OR(msg, pvt->last_provisional), &pvt->initreq, XMIT_UNRELIABLE, FALSE, FALSE);
                } else {
                        transmit_response(pvt, S_OR(msg, pvt->last_provisional), &pvt->initreq);
                }
                return PROVIS_KEEPALIVE_TIMEOUT;
        }

        return 0;
}

static int send_provisional_keepalive(const void *data) {
        struct sip_pvt *pvt = (struct sip_pvt *) data;

        return send_provisional_keepalive_full(pvt, 0);
}

static int send_provisional_keepalive_with_sdp(const void *data) {
        struct sip_pvt *pvt = (void *)data;

        return send_provisional_keepalive_full(pvt, 1);
}

static void update_provisional_keepalive(struct sip_pvt *pvt, int with_sdp)
{
        AST_SCHED_DEL_UNREF(sched, pvt->provisional_keepalive_sched_id, dialog_unref(pvt, "when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr"));

        pvt->provisional_keepalive_sched_id = ast_sched_add(sched, PROVIS_KEEPALIVE_TIMEOUT,
                with_sdp ? send_provisional_keepalive_with_sdp : send_provisional_keepalive, dialog_ref(pvt, "Increment refcount to pass dialog pointer to sched callback"));
}

/*! \brief Transmit response on SIP request*/
static int send_response(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
{
        int res;

        add_blank(req);
        if (sip_debug_test_pvt(p)) {
                const struct sockaddr_in *dst = sip_real_dst(p);

                ast_verbose("\n<--- %sTransmitting (%s) to %s:%d --->\n%s\n<------------>\n",
                        reliable ? "Reliably " : "", sip_nat_mode(p),
                        ast_inet_ntoa(dst->sin_addr),
                        ntohs(dst->sin_port), req->data->str);
        }
        if (p->do_history) {
                struct sip_request tmp = { .rlPart1 = 0, };
                parse_copy(&tmp, req);
                append_history(p, reliable ? "TxRespRel" : "TxResp", "%s / %s - %s", tmp.data->str, get_header(&tmp, "CSeq"),
                        (tmp.method == SIP_RESPONSE || tmp.method == SIP_UNKNOWN) ? REQ_OFFSET_TO_STR(&tmp, rlPart2) : sip_methods[tmp.method].text);
                ast_free(tmp.data);
        }

        /* If we are sending a final response to an INVITE, stop retransmitting provisional responses */
        if (p->initreq.method == SIP_INVITE && reliable == XMIT_CRITICAL) {
                AST_SCHED_DEL_UNREF(sched, p->provisional_keepalive_sched_id, dialog_unref(p, "when you delete the provisional_keepalive_sched_id, you should dec the refcount for the stored dialog ptr"));
        }

        res = (reliable) ?
                 __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable == XMIT_CRITICAL), req->method) :
                __sip_xmit(p, req->data, req->len);
        ast_free(req->data);
        req->data = NULL;
        if (res > 0)
                return 0;
        return res;
}

/*! \brief Send SIP Request to the other part of the dialogue 
        \return see \ref __sip_xmit
*/
static int send_request(struct sip_pvt *p, struct sip_request *req, enum xmittype reliable, int seqno)
{
        int res;

        /* If we have an outbound proxy, reset peer address
                Only do this once.
        */
        if (p->outboundproxy) {
                p->sa = p->outboundproxy->ip;
        }

        add_blank(req);
        if (sip_debug_test_pvt(p)) {
                if (ast_test_flag(&p->flags[0], SIP_NAT_FORCE_RPORT))