@voipsdk/react-native-voipcloud 1.0.0 → 1.0.2

package/ios/VoipCloud.xcframework/ios-arm64_x86_64-simulator/Headers/pjsip/sip_config.h

@@ -0,0 +1,1545 @@
+/* 
+ * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
+ * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA 
+ */
+#ifndef __PJSIP_SIP_CONFIG_H__
+#define __PJSIP_SIP_CONFIG_H__
+
+/**
+ * @file sip_config.h
+ * @brief Compile time configuration.
+ */
+#include <pj/types.h>
+
+/**
+ * @defgroup PJSIP_CORE Core SIP Library
+ * @brief The core framework from which all other SIP components depends on.
+ * 
+ * The PJSIP Core library only provides transport framework, event
+ * dispatching/module framework, and SIP message representation and
+ * parsing. It doesn't do anything usefull in itself!
+ *
+ * If application wants the stack to do anything usefull at all,
+ * it must registers @ref PJSIP_MOD to the core library. Examples
+ * of modules are @ref PJSIP_TRANSACT and @ref PJSUA_UA.
+ */
+
+/**
+ * @defgroup PJSIP_BASE Base Types
+ * @ingroup PJSIP_CORE
+ * @brief Basic PJSIP types and configurations.
+ */
+
+/**
+ * @defgroup PJSIP_CONFIG PJSIP Configurations/Settings
+ * @ingroup PJSIP_BASE
+ * @brief PJSIP compile time configurations.
+ * @{
+ */
+
+/*
+ * Include sip_autoconf.h if autoconf is used (PJ_AUTOCONF is set)
+ */
+#if defined(PJ_AUTOCONF)
+#   include <pjsip/sip_autoconf.h>
+#endif
+
+PJ_BEGIN_DECL
+
+/**
+ * This structure describes PJSIP run-time configurations/settings.
+ * Application may use #pjsip_cfg() function to modify the settings
+ * before creating the stack.
+ */
+typedef struct pjsip_cfg_t
+{
+    /** Global settings. */
+    struct {
+        /**
+         * Specify port number should be allowed to appear in To and From
+         * header. Note that RFC 3261 disallow this, see Table 1 in section
+         * 19.1.1 of the RFC.
+         *
+         * Default is PJSIP_ALLOW_PORT_IN_FROMTO_HDR.
+         */
+        pj_bool_t allow_port_in_fromto_hdr;
+
+        /**
+         * Accept call replace in early state when invite is not initiated
+         * by the user agent. RFC 3891 Section 3 disallows this, however,
+         * for better interoperability reason, this might be ignored.
+         *
+         * Default is PJSIP_ACCEPT_REPLACE_IN_EARLY_STATE.
+         */
+        pj_bool_t accept_replace_in_early_state;
+
+        /**
+         * Allow hash character ('#') to appear in outgoing URIs. See
+         * https://github.com/pjsip/pjproject/issues/1569.
+         *
+         * Default is PJ_FALSE.
+         */
+        pj_bool_t allow_tx_hash_in_uri;
+
+        /**
+         * Disable rport in request.
+         *
+         * Default is PJ_FALSE.
+         */
+        pj_bool_t disable_rport;
+
+        /**
+         * Disable automatic switching from UDP to TCP if outgoing request
+         * is greater than 1300 bytes.
+         *
+         * Default is PJSIP_DONT_SWITCH_TO_TCP.
+         */
+        pj_bool_t disable_tcp_switch;
+
+        /**
+         * Disable automatic switching to secure transport (such as TLS)
+         * if target-URI does not use "sips" scheme nor secure transport,
+         * even when request-URI uses "sips" scheme.
+         *
+         * Default is PJSIP_DONT_SWITCH_TO_TLS.
+         */
+        pj_bool_t disable_tls_switch;
+
+        /**
+         * Enable call media session to always be updated to the latest
+         * received early media SDP when receiving forked early media
+         * (multiple 183 responses with different To tag).
+         *
+         * Default is PJSIP_FOLLOW_EARLY_MEDIA_FORK.
+         */
+        pj_bool_t follow_early_media_fork;
+
+        /**
+         * Specify whether "alias" param should be added to the Via header
+         * in any outgoing request with connection oriented transport.
+         *
+         * Default is PJSIP_REQ_HAS_VIA_ALIAS.
+         */
+        pj_bool_t req_has_via_alias;
+
+        /**
+         * Resolve hostname when trying to get the network interface to be put 
+         * in Via or Contact header.
+         *
+         * Default is PJSIP_RESOLVE_HOSTNAME_TO_GET_INTERFACE.
+         */
+        pj_bool_t resolve_hostname_to_get_interface;
+
+        /**
+         * Disable security check on incoming messages in a secure dialog.
+         * A secure dialog is created when the request that creates the dialog
+         * uses "sips" scheme in its request URI. Contact URI should use "sips"
+         * scheme and the top-most Record-Route URI, if any, should use either
+         * "sips" scheme or "transport=tls" param. See also
+         * https://github.com/pjsip/pjproject/issues/1735.
+         *
+         * Default is PJ_FALSE.
+         */
+        pj_bool_t disable_secure_dlg_check;
+
+        /**
+         * Encode SIP headers in their short forms to reduce size. By default,
+         * SIP headers in outgoing messages will be encoded in their full names.
+         * If this option is enabled, then SIP headers for outgoing messages
+         * will be encoded in their short forms, to reduce message size. 
+         * Note that this does not affect the ability of PJSIP to parse incoming
+         * SIP messages, as the parser always supports parsing both the long
+         * and short version of the headers.
+         *
+         * Default is PJSIP_ENCODE_SHORT_HNAME
+         */
+        pj_bool_t use_compact_form;
+
+        /**
+         * Accept multiple SDP answers on non-reliable 18X responses and the 2XX
+         * response when they are all received from the same source (same To tag).
+         *
+         * See also:
+         * https://tools.ietf.org/html/rfc6337#section-3.1.1
+         *
+         * Default is PJSIP_ACCEPT_MULTIPLE_SDP_ANSWERS.
+         */
+        pj_bool_t accept_multiple_sdp_answers;
+
+        /**
+         * Don't disconnect the INVITE session after an outgoing request
+         * gets timed out or responded with 408 (request timeout).
+         *
+         * Default is PJ_FALSE.
+         */
+        pj_bool_t keep_inv_after_tsx_timeout;
+
+    } endpt;
+
+    /** Transaction layer settings. */
+    struct {
+
+        /** Maximum number of transactions. The value is initialized with
+         *  PJSIP_MAX_TSX_COUNT
+         */
+        unsigned max_count;
+
+        /* Timeout values: */
+
+        /** Transaction T1 timeout, in msec. Default value is PJSIP_T1_TIMEOUT
+         */
+        unsigned t1;
+
+        /** Transaction T2 timeout, in msec. Default value is PJSIP_T2_TIMEOUT
+         */
+        unsigned t2;
+
+        /** Transaction completed timer for non-INVITE, in msec. Default value
+         *  is PJSIP_T4_TIMEOUT
+         */
+        unsigned t4;
+
+        /** Transaction completed timer for INVITE, in msec. Default value is
+         *  PJSIP_TD_TIMEOUT.
+         *
+         *  This setting is also used for transaction timeout timer for both
+         *  INVITE and non-INVITE.
+         */
+        unsigned td;
+
+    } tsx;
+
+    /* Dialog layer settings .. TODO */
+
+    /** Client registration settings. */
+    struct {
+        /**
+         * Specify whether client registration should check for its 
+         * registered contact in Contact header of successful REGISTER 
+         * response to determine whether registration has been successful. 
+         * This setting may be disabled if non-compliant registrar is unable
+         * to return correct Contact header.
+         *
+         * Default is PJSIP_REGISTER_CLIENT_CHECK_CONTACT
+         */
+        pj_bool_t   check_contact;
+
+        /**
+         * Specify whether client registration should add "x-uid" extension
+         * parameter in all Contact URIs that it registers to assist the
+         * matching of Contact URIs in the 200/OK REGISTER response, in 
+         * case the registrar is unable to return exact Contact URI in the
+         * 200/OK response.
+         *
+         * Default is PJSIP_REGISTER_CLIENT_ADD_XUID_PARAM.
+         */
+        pj_bool_t   add_xuid_param;
+
+    } regc;
+
+    /** TCP transport settings */
+    struct {
+        /**
+         * Set the interval to send keep-alive packet for TCP transports.
+         * If the value is zero, keep-alive will be disabled for TCP.
+         *
+         * Default is PJSIP_TCP_KEEP_ALIVE_INTERVAL.
+         */
+        long keep_alive_interval;
+
+    } tcp;
+
+    /** TLS transport settings */
+    struct {
+        /**
+         * Set the interval to send keep-alive packet for TLS transports.
+         * If the value is zero, keep-alive will be disabled for TLS.
+         *
+         * Default is PJSIP_TLS_KEEP_ALIVE_INTERVAL.
+         */
+        long keep_alive_interval;
+
+    } tls;
+
+} pjsip_cfg_t;
+
+
+#ifdef PJ_DLL
+/**
+ * Get pjsip configuration instance. Application may modify the
+ * settings before creating the SIP endpoint and modules.
+ *
+ * @return  Configuration instance.
+ */
+PJ_DECL(pjsip_cfg_t*) pjsip_cfg(void);
+
+#else   /* PJ_DLL */
+
+extern pjsip_cfg_t pjsip_sip_cfg_var;
+
+/**
+ * Get pjsip configuration instance. Application may modify the
+ * settings before creating the SIP endpoint and modules.
+ *
+ * @return  Configuration instance.
+ */
+PJ_INLINE(pjsip_cfg_t*) pjsip_cfg(void)
+{
+    return &pjsip_sip_cfg_var;
+}
+
+#endif  /* PJ_DLL */
+
+
+/**
+ * Specify maximum transaction count in transaction hash table.
+ * For efficiency, the value should be 2^n-1 since it will be
+ * rounded up to 2^n.
+ *
+ * Default value is 1023
+ */
+#ifndef PJSIP_MAX_TSX_COUNT
+#   define PJSIP_MAX_TSX_COUNT          (1024-1)
+#endif
+
+/**
+ * Specify maximum number of dialogs in the dialog hash table.
+ * For efficiency, the value should be 2^n-1 since it will be
+ * rounded up to 2^n.
+ *
+ * Default value is 511.
+ */
+#ifndef PJSIP_MAX_DIALOG_COUNT
+#   define PJSIP_MAX_DIALOG_COUNT       (512-1)
+#endif
+
+
+/**
+ * Specify maximum number of transports.
+ * Default value is equal to maximum number of handles in ioqueue.
+ * See also PJSIP_TPMGR_HTABLE_SIZE.
+ */
+#ifndef PJSIP_MAX_TRANSPORTS
+#   define PJSIP_MAX_TRANSPORTS         (PJ_IOQUEUE_MAX_HANDLES)
+#endif
+
+
+/**
+ * Transport manager hash table size (must be 2^n-1). 
+ * See also PJSIP_MAX_TRANSPORTS
+ */
+#ifndef PJSIP_TPMGR_HTABLE_SIZE
+#   define PJSIP_TPMGR_HTABLE_SIZE      31
+#endif
+
+
+/**
+ * Specify maximum URL size.
+ */
+#ifndef PJSIP_MAX_URL_SIZE
+#   define PJSIP_MAX_URL_SIZE           256
+#endif
+
+
+/**
+ * Specify maximum number of modules.
+ * This mainly affects the size of mod_data array in various components.
+ */
+#ifndef PJSIP_MAX_MODULE
+#   define PJSIP_MAX_MODULE             32
+#endif
+
+
+/**
+ * Maximum packet length. We set it more than MTU since a SIP PDU
+ * containing presence information can be quite large (>1500).
+ */
+#ifndef PJSIP_MAX_PKT_LEN
+#   define PJSIP_MAX_PKT_LEN            4000
+#endif
+
+
+/**
+ * RFC 3261 section 18.1.1:
+ * If a request is within 200 bytes of the path MTU, or if it is larger
+ * than 1300 bytes and the path MTU is unknown, the request MUST be sent
+ * using an RFC 2914 [43] congestion controlled transport protocol, such
+ * as TCP.
+ *
+ * Disable the behavior of automatic switching to TCP whenever UDP packet
+ * size exceeds the threshold defined in PJSIP_UDP_SIZE_THRESHOLD.
+ *
+ * This option can also be controlled at run-time by the \a disable_tcp_switch
+ * setting in pjsip_cfg_t.
+ *
+ * Default is 0 (no).
+ */
+#ifndef PJSIP_DONT_SWITCH_TO_TCP
+#   define PJSIP_DONT_SWITCH_TO_TCP     0
+#endif
+
+
+/**
+ * As specified RFC 3261 section 8.1.2, when request-URI uses "sips" scheme,
+ * TLS must always be used regardless of the target-URI scheme or transport
+ * type.
+ * Update: Newer RFCs, such as RFC 5630 and 7118, expands this by allowing
+ * the use of other transports as long as the SIP resource designated by
+ * the target SIPS URI is contacted securely.
+ *
+ * This option will specify whether the behavior of automatic switching to secure
+ * transport (such as TLS) should be disabled, i.e: regard the target-URI scheme
+ * or transport type.
+ *
+ * This option can also be controlled at run-time by the \a disable_tls_switch
+ * setting in pjsip_cfg_t.
+ *
+ * Default is 0 (no).
+ */
+#ifndef PJSIP_DONT_SWITCH_TO_TLS
+#   define PJSIP_DONT_SWITCH_TO_TLS     0
+#endif
+
+/**
+ * Specify if #pjsip_endpt_handle_events() should sleep if ioqueue poll
+ * returns error.
+ *
+ * Default is 1 (yes).
+ */
+
+#ifndef PJSIP_HANDLE_EVENTS_HAS_SLEEP_ON_ERR
+#   define PJSIP_HANDLE_EVENTS_HAS_SLEEP_ON_ERR     1
+#endif
+
+/**
+ * Specify whether the call media session should be updated to the latest
+ * received early media SDP when receiving forked early media (multiple 183
+ * responses with different To tag).
+ *
+ * This option can also be controlled at run-time by the
+ * \a follow_early_media_fork setting in pjsip_cfg_t.
+ *
+ * Default is PJ_TRUE.
+ */
+#ifndef PJSIP_FOLLOW_EARLY_MEDIA_FORK
+#   define PJSIP_FOLLOW_EARLY_MEDIA_FORK            PJ_TRUE
+#endif
+
+
+/**
+ * Accept multiple SDP answers on non-reliable 18X responses and the 2XX
+ * response when they are all received from the same source (same To tag).
+ *
+ * This option can also be controlled at run-time by the
+ * \a accept_multiple_sdp_answers setting in pjsip_cfg_t.
+ *
+ * Default is PJ_TRUE.
+ */
+#ifndef PJSIP_ACCEPT_MULTIPLE_SDP_ANSWERS
+#   define PJSIP_ACCEPT_MULTIPLE_SDP_ANSWERS        PJ_TRUE
+#endif
+
+
+/**
+ * Specify whether "alias" param should be added to the Via header
+ * in any outgoing request with connection oriented transport.
+ *
+ * This option can also be controlled at run-time by the
+ * \a req_has_via_alias setting in pjsip_cfg_t.
+ *
+ * Default is PJ_TRUE.
+ */
+#ifndef PJSIP_REQ_HAS_VIA_ALIAS
+#   define PJSIP_REQ_HAS_VIA_ALIAS                  PJ_TRUE
+#endif
+
+/**
+ * Resolve hostname when trying to get the network interface to be put in Via 
+ * or Contact header.
+ *
+ * This option can also be controlled at run-time by the
+ * \a resolve_hostname_to_get_interface setting in pjsip_cfg_t.
+ *
+ * Default is PJ_FALSE.
+ */
+#ifndef PJSIP_RESOLVE_HOSTNAME_TO_GET_INTERFACE
+#   define PJSIP_RESOLVE_HOSTNAME_TO_GET_INTERFACE  PJ_FALSE
+#endif
+
+/**
+ * Accept call replace in early state when invite is not initiated
+ * by the user agent. RFC 3891 Section 3 disallows this, however,
+ * for better interoperability reason, this might be ignored.
+ *
+ * This option can also be controlled at run-time by the
+ * \a accept_replace_in_early_state setting in pjsip_cfg_t.
+ *
+ * Default is 0 (no).
+ */
+#ifndef PJSIP_ACCEPT_REPLACE_IN_EARLY_STATE
+#   define PJSIP_ACCEPT_REPLACE_IN_EARLY_STATE      0
+#endif
+
+
+/**
+ * This setting controls the threshold of the UDP packet, which if it's
+ * larger than this value the request will be sent with TCP. This setting
+ * is useful only when PJSIP_DONT_SWITCH_TO_TCP is set to 0.
+ *
+ * Default is 1300 bytes.
+ */
+#ifndef PJSIP_UDP_SIZE_THRESHOLD
+#   define PJSIP_UDP_SIZE_THRESHOLD     1300
+#endif
+
+
+/**
+ * Encode SIP headers in their short forms to reduce size. By default,
+ * SIP headers in outgoing messages will be encoded in their full names. 
+ * If this option is enabled, then SIP headers for outgoing messages
+ * will be encoded in their short forms, to reduce message size. 
+ * Note that this does not affect the ability of PJSIP to parse incoming
+ * SIP messages, as the parser always supports parsing both the long
+ * and short version of the headers.
+ *
+ * This option can also be controlled at run-time by the
+ * \a use_compact_form setting in pjsip_cfg_t.
+ *
+ * Default is 0 (no)
+ */
+#ifndef PJSIP_ENCODE_SHORT_HNAME
+#   define PJSIP_ENCODE_SHORT_HNAME     0
+#endif
+
+
+/**
+ * Send Allow header in dialog establishing requests?
+ * RFC 3261 Allow header SHOULD be included in dialog establishing
+ * requests to inform remote agent about which SIP requests are
+ * allowed within dialog.
+ *
+ * Note that there is also an undocumented variable defined in sip_dialog.c
+ * to control whether Allow header should be included. The default value 
+ * of this variable is PJSIP_INCLUDE_ALLOW_HDR_IN_DLG.
+ * To change PJSIP behavior during run-time, application can use the 
+ * following construct:
+ *
+ \verbatim
+   extern pj_bool_t pjsip_include_allow_hdr_in_dlg;
+ 
+   // do not transmit Allow header
+   pjsip_include_allow_hdr_in_dlg = PJ_FALSE;
+ \endverbatim
+ *
+ * Default is 1 (Yes)
+ */
+#ifndef PJSIP_INCLUDE_ALLOW_HDR_IN_DLG
+#   define PJSIP_INCLUDE_ALLOW_HDR_IN_DLG       1
+#endif
+
+
+/**
+ * Allow SIP modules removal or insertions during operation?
+ * If yes, then locking will be employed when endpoint need to
+ * access module.
+ */
+#ifndef PJSIP_SAFE_MODULE
+#   define PJSIP_SAFE_MODULE            1
+#endif
+
+
+/**
+ * Perform Via sent-by checking as specified in RFC 3261 Section 18.1.2,
+ * which says that UAC MUST silently discard responses with Via sent-by
+ * containing values that the UAC doesn't recognize as its transport
+ * address.
+ *
+ * In PJSIP, this will cause response to be discarded and a message is
+ * written to the log, saying something like:
+ *  "Dropping response Response msg 200/INVITE/cseq=608594373 (rdata00A99EF4)
+ *   from 1.2.3.4:5060 because sent-by is mismatch"
+ *
+ * The default behavior is yes, but when the UA supports IP address change
+ * for the SIP transport, it will need to turn this checking off since
+ * when the transport address is changed between request is sent and 
+ * response is received, the response will be discarded since its Via
+ * sent-by now contains address that is different than the transport
+ * address.
+ *
+ * Update:
+ * As of version 2.1, the default value is 0. This change was part of
+ * https://github.com/pjsip/pjproject/issues/1412
+ */
+#ifndef PJSIP_CHECK_VIA_SENT_BY
+#   define PJSIP_CHECK_VIA_SENT_BY      0
+#endif
+
+
+/**
+ * If non-zero, SIP parser will unescape the escape characters ('%')
+ * in the original message, which means that it will modify the
+ * original message. Otherwise the parser will create a copy of
+ * the string and store the unescaped string to the new location.
+ *
+ * Unescaping in-place is faster, but less elegant (and it may
+ * break certain applications). So normally it's disabled, unless
+ * when benchmarking (to show off big performance).
+ *
+ * Default: 0
+ */
+#ifndef PJSIP_UNESCAPE_IN_PLACE
+#   define PJSIP_UNESCAPE_IN_PLACE      0
+#endif
+
+
+/**
+ * Specify port number should be allowed to appear in To and From
+ * header. Note that RFC 3261 disallow this, see Table 1 in section
+ * 19.1.1 of the RFC. This setting can also be altered at run-time
+ * via pjsip_cfg setting, see pjsip_cfg_t.allow_port_in_fromto_hdr
+ * field.
+ *
+ * Default: 0
+ */
+#ifndef PJSIP_ALLOW_PORT_IN_FROMTO_HDR
+#   define PJSIP_ALLOW_PORT_IN_FROMTO_HDR       0
+#endif
+
+/**
+ * This macro controls maximum numbers of ioqueue events to be processed
+ * in a single pjsip_endpt_handle_events() poll. When PJSIP detects that
+ * there are probably more events available from the network and total
+ * events so far is less than this value, PJSIP will call pj_ioqueue_poll()
+ * again to get more events.
+ *
+ * Value 1 works best for ioqueue with select() back-end, while for IOCP it is
+ * probably best to set this value equal to PJSIP_MAX_TIMED_OUT_ENTRIES
+ * since IOCP only processes one event at a time.
+ *
+ * Default: 1
+ */
+#ifndef PJSIP_MAX_NET_EVENTS
+#   define PJSIP_MAX_NET_EVENTS         1
+#endif
+
+
+/**
+ * Max entries to process in timer heap per poll. 
+ * 
+ * Default: 10
+ */
+#ifndef PJSIP_MAX_TIMED_OUT_ENTRIES
+#   define PJSIP_MAX_TIMED_OUT_ENTRIES  10
+#endif
+
+
+/**
+ * Idle timeout interval to be applied to outgoing transports (i.e. client
+ * side) with no usage before the transport is destroyed. Value is in
+ * seconds.
+ *
+ * Note that if the value is put lower than 33 seconds, it may cause some
+ * pjsip test units to fail. See the comment on the following link:
+ * https://github.com/pjsip/pjproject/issues/1465#comment:4
+ *
+ * Default: 33
+ */
+#ifndef PJSIP_TRANSPORT_IDLE_TIME
+#   define PJSIP_TRANSPORT_IDLE_TIME    33
+#endif
+
+
+/**
+ * Idle timeout interval to be applied to incoming transports (i.e. server
+ * side) with no usage before the transport is destroyed. Server typically
+ * should let client close the connection, hence set this interval to a large
+ * value. Value is in seconds.
+ *
+ * Default: 600
+ */
+#ifndef PJSIP_TRANSPORT_SERVER_IDLE_TIME
+#   define PJSIP_TRANSPORT_SERVER_IDLE_TIME     600
+#endif
+
+
+/**
+ * The initial timeout interval for incoming TCP/TLS transports
+ * (i.e. server side) in the event that no valid SIP message is received
+ * following a successful connection. The value is in seconds.
+ * Disable the timeout by setting it to 0.
+ *
+ * Note that even if this is disabled, the connection might still get closed
+ * when it is idle or not referred anymore. Have a look at \a
+ * PJSIP_TRANSPORT_SERVER_IDLE_TIME.
+ * 
+ * Notes:
+ * - keep-alive packet is not considered as a valid message.
+ *
+ * Default: 0
+*/
+#ifndef PJSIP_TRANSPORT_SERVER_IDLE_TIME_FIRST
+#   define PJSIP_TRANSPORT_SERVER_IDLE_TIME_FIRST     0
+#endif
+
+/**
+ * Maximum number of usages for a transport before a new transport is
+ * created. This only applies for ephemeral transports such as TCP.
+ *
+ * Currently this is not used.
+ * 
+ * Default: -1
+ */
+#ifndef PJSIP_MAX_TRANSPORT_USAGE
+#   define PJSIP_MAX_TRANSPORT_USAGE    ((unsigned)-1)
+#endif
+
+
+/**
+ * The TCP incoming connection backlog number to be set in accept().
+ *
+ * Default: 5
+ *
+ * @see PJSIP_TLS_TRANSPORT_BACKLOG
+ */
+#ifndef PJSIP_TCP_TRANSPORT_BACKLOG
+#   define PJSIP_TCP_TRANSPORT_BACKLOG  5
+#endif
+
+
+/**
+ * Specify whether TCP listener should use SO_REUSEADDR option. This constant
+ * will be used as the default value for the "reuse_addr" field in the
+ * pjsip_tcp_transport_cfg structure.
+ *
+ * Default is 0 on Windows and 1 on non-Windows.
+ *
+ * @see PJSIP_TLS_TRANSPORT_REUSEADDR
+ */
+#ifndef PJSIP_TCP_TRANSPORT_REUSEADDR
+# if (defined(PJ_WIN32) && PJ_WIN32) || (defined(PJ_WIN64) && PJ_WIN64)
+#   define PJSIP_TCP_TRANSPORT_REUSEADDR        0
+# else
+#   define PJSIP_TCP_TRANSPORT_REUSEADDR        1
+# endif
+#endif
+
+
+/**
+ * Specify whether TCP transport should skip creating the listener.
+ * Not having a listener means that application will not be able to
+ * function in server mode and accept incoming connections.
+ *
+ * When enabling this setting, if you use PJSUA, it is recommended to set 
+ * pjsua_acc_config.contact_use_src_port to PJ_TRUE.
+ * Warning: If contact_use_src_port is disabled or failed (because it's
+ * unsupported in some platforms or automatically turned off due to
+ * DNS server resolution), Contact header will be generated from
+ * pj_getipinterface()/pj_gethostip(), but the address will not be
+ * able to accept connections. 
+ *
+ * Default is 0 (listener will be created).
+ */
+#ifndef PJSIP_TCP_TRANSPORT_DONT_CREATE_LISTENER
+#   define PJSIP_TCP_TRANSPORT_DONT_CREATE_LISTENER 0
+#endif
+
+
+/**
+ * Specify whether TLS transport should skip creating the listener.
+ * Not having a listener means that application will not be able to
+ * function in server mode and accept incoming connections.
+ *
+ * When enabling this setting, if you use PJSUA, it is recommended to set 
+ * pjsua_acc_config.contact_use_src_port to PJ_TRUE.
+ * Warning: If contact_use_src_port is disabled or failed (because it's
+ * unsupported in some platforms or automatically turned off due to
+ * DNS server resolution), Contact header will be generated from
+ * pj_getipinterface()/pj_gethostip(), but the address will not be
+ * able to accept connections.
+ *
+ * Default is 0 (listener will be created).
+ */
+#ifndef PJSIP_TLS_TRANSPORT_DONT_CREATE_LISTENER
+#   define PJSIP_TLS_TRANSPORT_DONT_CREATE_LISTENER 0
+#endif
+
+
+/**
+ * Set the interval to send keep-alive packet for TCP transports.
+ * If the value is zero, keep-alive will be disabled for TCP.
+ *
+ * This option can be changed in run-time by settting
+ * \a tcp.keep_alive_interval field of pjsip_cfg().
+ *
+ * Default: 90 (seconds)
+ *
+ * @see PJSIP_TCP_KEEP_ALIVE_DATA
+ */
+#ifndef PJSIP_TCP_KEEP_ALIVE_INTERVAL
+#   define PJSIP_TCP_KEEP_ALIVE_INTERVAL    90
+#endif
+
+
+/**
+ * Set the payload of the TCP keep-alive packet.
+ *
+ * Default: CRLF
+ */
+#ifndef PJSIP_TCP_KEEP_ALIVE_DATA
+#   define PJSIP_TCP_KEEP_ALIVE_DATA        { "\r\n\r\n", 4 }
+#endif
+
+
+/**
+ * The initial timeout interval for incoming TCP transports
+ * (i.e. server side) in the event that no valid SIP message is received
+ * following a successful connection. The value is in seconds.
+ * Disable the timeout by setting it to 0.
+ *
+ * Note that even if this is disabled, the connection might still get closed
+ * when it is idle or not referred anymore. Have a look at \a
+ * PJSIP_TRANSPORT_SERVER_IDLE_TIME.
+ *
+ * Notes:
+ * - keep-alive packet is not considered as a valid message.
+ * - This macro is specific to TCP usage and takes precedence over
+ *   a\ PJSIP_TRANSPORT_SERVER_IDLE_TIME_FIRST when both are set.
+ * 
+ * Default: 0 (disabled)
+ */
+#ifndef PJSIP_TCP_INITIAL_TIMEOUT
+#   define PJSIP_TCP_INITIAL_TIMEOUT        0
+#endif
+
+/**
+ * Set the interval to send keep-alive packet for TLS transports.
+ * If the value is zero, keep-alive will be disabled for TLS.
+ *
+ * This option can be changed in run-time by settting
+ * \a tls.keep_alive_interval field of pjsip_cfg().
+ *
+ * Default: 90 (seconds)
+ *
+ * @see PJSIP_TLS_KEEP_ALIVE_DATA
+ */
+#ifndef PJSIP_TLS_KEEP_ALIVE_INTERVAL
+#   define PJSIP_TLS_KEEP_ALIVE_INTERVAL    90
+#endif
+
+
+/**
+ * Set the payload of the TLS keep-alive packet.
+ *
+ * Default: CRLF
+ */
+#ifndef PJSIP_TLS_KEEP_ALIVE_DATA
+#   define PJSIP_TLS_KEEP_ALIVE_DATA        { "\r\n\r\n", 4 }
+#endif
+
+
+/**
+ * This macro specifies whether full DNS resolution should be used.
+ * When enabled, #pjsip_resolve() will perform asynchronous DNS SRV and
+ * A (or AAAA, when IPv6 is supported) resolution to resolve the SIP
+ * domain.
+ *
+ * Note that even when this setting is enabled, asynchronous DNS resolution
+ * will only be done when application calls #pjsip_endpt_create_resolver(),
+ * configure the nameservers with pj_dns_resolver_set_ns(), and configure
+ * the SIP endpoint's DNS resolver with #pjsip_endpt_set_resolver(). If
+ * these steps are not followed, the domain will be resolved with normal
+ * pj_gethostbyname() function.
+ *
+ * Turning off this setting will save the footprint by about 16KB, since
+ * it should also exclude dns.o and resolve.o from PJLIB-UTIL.
+ *
+ * Default: 1 (enabled)
+ *
+ * @see PJSIP_MAX_RESOLVED_ADDRESSES
+ */
+#ifndef PJSIP_HAS_RESOLVER
+#   define PJSIP_HAS_RESOLVER           1
+#endif
+
+
+/** 
+ * Maximum number of addresses returned by the resolver. The number here 
+ * will slightly affect stack usage, since each entry will occupy about
+ * 32 bytes of stack memory.
+ *
+ * Default: 16 (or 32 if IPv6 support is enabled)
+ *
+ * @see PJSIP_HAS_RESOLVER
+ */
+#ifndef PJSIP_MAX_RESOLVED_ADDRESSES
+#   if defined(PJ_HAS_IPV6) && PJ_HAS_IPV6
+#       define PJSIP_MAX_RESOLVED_ADDRESSES         32
+#   else
+#       define PJSIP_MAX_RESOLVED_ADDRESSES         16
+#   endif
+#endif
+
+
+/**
+ * Enable TLS SIP transport support. For most systems this means that
+ * OpenSSL must be installed.
+ *
+ * Default: follow PJ_HAS_SSL_SOCK setting, which is 0 (disabled) by default.
+ */
+#ifndef PJSIP_HAS_TLS_TRANSPORT
+#   define PJSIP_HAS_TLS_TRANSPORT          PJ_HAS_SSL_SOCK
+#endif
+
+
+/**
+ * The TLS pending incoming connection backlog number to be set in accept().
+ *
+ * Default: 5
+ *
+ * @see PJSIP_TCP_TRANSPORT_BACKLOG
+ */
+#ifndef PJSIP_TLS_TRANSPORT_BACKLOG
+#   define PJSIP_TLS_TRANSPORT_BACKLOG      5
+#endif
+
+
+/**
+ * Specify whether TLS listener should use SO_REUSEADDR option.
+ *
+ * Default is 0 on Windows and 1 on non-Windows.
+ *
+ * @see PJSIP_TCP_TRANSPORT_REUSEADDR
+ */
+#ifndef PJSIP_TLS_TRANSPORT_REUSEADDR
+# if (defined(PJ_WIN32) && PJ_WIN32) || (defined(PJ_WIN64) && PJ_WIN64)
+#   define PJSIP_TLS_TRANSPORT_REUSEADDR        0
+# else
+#   define PJSIP_TLS_TRANSPORT_REUSEADDR        1
+# endif
+#endif
+
+
+/**
+ * Specify the maximum number of timer entries initially allocated by
+ * endpoint. If the application registers more entries during runtime,
+ * then the timer will automatically resize.
+ *
+ * Default: (2*pjsip_cfg()->tsx.max_count) + (2*PJSIP_MAX_DIALOG_COUNT)
+ */
+#ifndef PJSIP_MAX_TIMER_COUNT
+#   define PJSIP_MAX_TIMER_COUNT        (2*pjsip_cfg()->tsx.max_count + \
+                                         2*PJSIP_MAX_DIALOG_COUNT)
+#endif
+
+/**
+ * Initial memory block for the endpoint.
+ */
+#ifndef PJSIP_POOL_LEN_ENDPT
+#   define PJSIP_POOL_LEN_ENDPT         (16000)
+#endif
+
+/**
+ * Memory increment for endpoint.
+ */
+#ifndef PJSIP_POOL_INC_ENDPT
+#   define PJSIP_POOL_INC_ENDPT         (4000)
+#endif
+
+
+/* Transport related constants. */
+
+/**
+ * Initial memory block for rdata.
+ */
+#ifndef PJSIP_POOL_RDATA_LEN
+#   define PJSIP_POOL_RDATA_LEN         4000
+#endif
+
+/**
+ * Memory increment for rdata.
+ */
+#ifndef PJSIP_POOL_RDATA_INC
+#   define PJSIP_POOL_RDATA_INC         4000
+#endif
+
+/**
+ * Initial memory block for SIP transport.
+ */
+#ifndef PJSIP_POOL_LEN_TRANSPORT
+#   define PJSIP_POOL_LEN_TRANSPORT     512
+#endif
+
+/**
+ * Memory increment for SIP transport.
+ */
+#ifndef PJSIP_POOL_INC_TRANSPORT
+#   define PJSIP_POOL_INC_TRANSPORT     512
+#endif
+
+/**
+ * Initial memory block size for tdata.
+ */
+#ifndef PJSIP_POOL_LEN_TDATA
+#   define PJSIP_POOL_LEN_TDATA         4000
+#endif
+
+/**
+ * Memory increment for tdata.
+ */
+#ifndef PJSIP_POOL_INC_TDATA
+#   define PJSIP_POOL_INC_TDATA         4000
+#endif
+
+/**
+ * Initial memory size for UA layer
+ */
+#ifndef PJSIP_POOL_LEN_UA
+#   define PJSIP_POOL_LEN_UA            512
+#endif
+
+/**
+ * Memory increment for UA layer.
+ */
+#ifndef PJSIP_POOL_INC_UA
+#   define PJSIP_POOL_INC_UA            512
+#endif
+
+/**
+ * Initial memory block for event subscription module.
+ */
+#ifndef PJSIP_POOL_EVSUB_LEN
+#   define PJSIP_POOL_EVSUB_LEN         512
+#endif
+
+/**
+ * Memory increment for event subscription module.
+ */
+#ifndef PJSIP_POOL_EVSUB_INC
+#   define PJSIP_POOL_EVSUB_INC         512
+#endif
+
+/**
+ * Default value for Max-Forwards header
+ */
+#define PJSIP_MAX_FORWARDS_VALUE        70
+
+/**
+ * Via branch parameter prefix
+ */
+#define PJSIP_RFC3261_BRANCH_ID         "z9hG4bK"
+
+/**
+ * Length of PJSIP_RFC3261_BRANCH_ID
+ */
+#define PJSIP_RFC3261_BRANCH_LEN        7
+
+/* Transaction related constants. */
+
+/**
+ * Initial memory size for transaction layer. The bulk of pool usage
+ * for transaction layer will be used to create the hash table, so 
+ * setting this value too high will not help too much with reducing
+ * fragmentation and the memory will most likely be wasted.
+ */
+#ifndef PJSIP_POOL_TSX_LAYER_LEN
+#   define PJSIP_POOL_TSX_LAYER_LEN     512
+#endif
+
+/**
+ * Memory increment for transaction layer. The bulk of pool usage
+ * for transaction layer will be used to create the hash table, so 
+ * setting this value too high will not help too much with reducing
+ * fragmentation and the memory will most likely be wasted.
+ */
+#ifndef PJSIP_POOL_TSX_LAYER_INC
+#   define PJSIP_POOL_TSX_LAYER_INC     512
+#endif
+
+/**
+ * Initial memory size for a SIP transaction object.
+ */
+#ifndef PJSIP_POOL_TSX_LEN
+#   define PJSIP_POOL_TSX_LEN           1536 /* 768 */
+#endif
+
+/**
+ * Memory increment for transaction object.
+ */
+#ifndef PJSIP_POOL_TSX_INC
+#   define PJSIP_POOL_TSX_INC           256
+#endif
+
+/**
+ * Delay for non-100 1xx retransmission, in seconds.
+ * Set to 0 to disable this feature.
+ *
+ * Default: 60 seconds
+ */
+#ifndef PJSIP_TSX_1XX_RETRANS_DELAY
+#   define PJSIP_TSX_1XX_RETRANS_DELAY  60
+#endif
+
+/**
+ * Setting to determine if certain SIP UAS transaction, such as
+ * INVITE UAS tsx that hasn't been confirmed, is allowed to continue
+ * upon transport error. If disabled, the transaction will always be
+ * terminated, which is the default behavior prior to the introduction
+ * of this setting.
+ *
+ * Default: 1 (transaction will continue)
+ */
+#ifndef PJSIP_TSX_UAS_CONTINUE_ON_TP_ERROR
+#   define PJSIP_TSX_UAS_CONTINUE_ON_TP_ERROR 1
+#endif
+
+#define PJSIP_MAX_TSX_KEY_LEN           (PJSIP_MAX_URL_SIZE*2)
+
+/* User agent's pool setting. */
+#define PJSIP_POOL_LEN_USER_AGENT       1024
+/* User agent's pool setting. */
+#define PJSIP_POOL_INC_USER_AGENT       1024
+
+/** Message/URL related constants. */
+#define PJSIP_MAX_CALL_ID_LEN           pj_GUID_STRING_LENGTH()
+/** Message/URL related constants. */
+#define PJSIP_MAX_TAG_LEN               pj_GUID_STRING_LENGTH()
+/** Message/URL related constants. */
+#define PJSIP_MAX_BRANCH_LEN            (PJSIP_RFC3261_BRANCH_LEN + pj_GUID_STRING_LENGTH() + 2)
+/** Message/URL related constants. */
+#define PJSIP_MAX_HNAME_LEN             64
+
+/** Dialog's pool setting. */
+#define PJSIP_POOL_LEN_DIALOG           4000
+/** Dialog's pool setting. */
+#define PJSIP_POOL_INC_DIALOG           4000
+
+/** Maximum header types. */
+#define PJSIP_MAX_HEADER_TYPES          72
+
+/** Maximum URI types. */
+#define PJSIP_MAX_URI_TYPES             4
+
+/*****************************************************************************
+ *  Default timeout settings, in miliseconds. 
+ */
+
+/** Transaction T1 timeout value. */
+#if !defined(PJSIP_T1_TIMEOUT)
+#  define PJSIP_T1_TIMEOUT      500
+#endif
+
+/** Transaction T2 timeout value. */
+#if !defined(PJSIP_T2_TIMEOUT)
+#  define PJSIP_T2_TIMEOUT      4000
+#endif
+
+/** Transaction completed timer for non-INVITE */
+#if !defined(PJSIP_T4_TIMEOUT)
+#  define PJSIP_T4_TIMEOUT      5000
+#endif
+
+/**
+ * Transaction completed timer for INVITE.
+ *
+ * This setting is also used for transaction timeout timer for both
+ * INVITE and non-INVITE.
+ */
+#if !defined(PJSIP_TD_TIMEOUT)
+#  define PJSIP_TD_TIMEOUT      32000
+#endif
+
+
+/*****************************************************************************
+ *  Authorization
+ */
+
+/**
+ * If this flag is set, the stack will keep the Authorization/Proxy-Authorization
+ * headers that are sent in a cache. Future requests with the same realm and
+ * the same method will use the headers in the cache (as long as no qop is
+ * required by server).
+ *
+ * Turning on this flag will make authorization process goes faster, but
+ * will grow the memory usage undefinitely until the dialog/registration
+ * session is terminated.
+ *
+ * Default: 0
+ */
+#if !defined(PJSIP_AUTH_HEADER_CACHING)
+#   define PJSIP_AUTH_HEADER_CACHING        0
+#endif
+
+/**
+ * If this flag is set, the stack will proactively send Authorization/Proxy-
+ * Authorization header for next requests. If next request has the same method
+ * with any of previous requests, then the last header which is saved in
+ * the cache will be used (if PJSIP_AUTH_CACHING is set). Otherwise a fresh
+ * header will be recalculated. If a particular server has requested qop, then
+ * a fresh header will always be calculated.
+ *
+ * If this flag is NOT set, then the stack will only send Authorization/Proxy-
+ * Authorization headers when it receives 401/407 response from server.
+ *
+ * Turning ON this flag will grow memory usage of a dialog/registration pool
+ * indefinitely until it is terminated, because the stack needs to keep the
+ * last WWW-Authenticate/Proxy-Authenticate challenge.
+ *
+ * Default: 0
+ */
+#if !defined(PJSIP_AUTH_AUTO_SEND_NEXT)
+#   define PJSIP_AUTH_AUTO_SEND_NEXT        0
+#endif
+
+/**
+ * Support qop="auth" directive.
+ * This option also requires client to cache the last challenge offered by
+ * server.
+ *
+ * Default: 1
+ */
+#if !defined(PJSIP_AUTH_QOP_SUPPORT)
+#   define PJSIP_AUTH_QOP_SUPPORT           1
+#endif
+
+
+/**
+ * Maximum number of stale retries when server keeps rejecting our request
+ * with stale=true.
+ *
+ * Default: 3
+ */
+#ifndef PJSIP_MAX_STALE_COUNT
+#   define PJSIP_MAX_STALE_COUNT            3
+#endif
+
+
+/**
+ * Specify support for IMS/3GPP digest AKA authentication version 1 and 2
+ * (AKAv1-MD5 and AKAv2-MD5 respectively).
+ *
+ * Note that if this is enabled, application would need to link with
+ * <b>libmilenage</b> library from \a third_party directory.
+ *
+ * Default: 0 (for now)
+ */
+#ifndef PJSIP_HAS_DIGEST_AKA_AUTH
+#   define PJSIP_HAS_DIGEST_AKA_AUTH        0
+#endif
+
+
+/**
+ * Specify the number of seconds to refresh the client registration
+ * before the registration expires.
+ *
+ * Default: 5 seconds
+ */
+#ifndef PJSIP_REGISTER_CLIENT_DELAY_BEFORE_REFRESH
+#   define PJSIP_REGISTER_CLIENT_DELAY_BEFORE_REFRESH  5
+#endif
+
+
+/**
+ * Specify whether client registration should check for its registered
+ * contact in Contact header of successful REGISTE response to determine
+ * whether registration has been successful. This setting may be disabled
+ * if non-compliant registrar is unable to return correct Contact header.
+ *
+ * This setting can be changed in run-time by settting \a regc.check_contact
+ * field of pjsip_cfg().
+ *
+ * Default is 1
+ */
+#ifndef PJSIP_REGISTER_CLIENT_CHECK_CONTACT
+#   define PJSIP_REGISTER_CLIENT_CHECK_CONTACT  1
+#endif
+
+
+/**
+ * Specify whether client registration should add "x-uid" extension
+ * parameter in all Contact URIs that it registers to assist the
+ * matching of Contact URIs in the 200/OK REGISTER response, in 
+ * case the registrar is unable to return exact Contact URI in the
+ * 200/OK response.
+ *
+ * This setting can be changed in run-time by setting 
+ * \a regc.add_xuid_param field of pjsip_cfg().
+ *
+ * Default is 0.
+ */
+#ifndef PJSIP_REGISTER_CLIENT_ADD_XUID_PARAM
+#   define PJSIP_REGISTER_CLIENT_ADD_XUID_PARAM 0
+#endif
+
+
+/**
+ * Allow client to send refresh registration when the registrar sent a Contact
+ * header with expire parameter 0 in the 200/OK REGISTER response.
+ * Refer to https://github.com/pjsip/pjproject/pull/2809 for more info.
+ *
+ * Default is 1.
+ */
+#ifndef PJSIP_REGISTER_ALLOW_EXP_REFRESH
+#   define PJSIP_REGISTER_ALLOW_EXP_REFRESH     1
+#endif
+
+
+/**
+ * Maximum size of pool allowed for auth client session in pjsip_regc.
+ * After the size exceeds because of Digest authentication processing,
+ * the pool is reset.
+ *
+ * Default is 20 kB
+ */
+#ifndef PJSIP_AUTH_CACHED_POOL_MAX_SIZE
+#   define PJSIP_AUTH_CACHED_POOL_MAX_SIZE      (20 * 1024)
+#endif
+
+
+/**
+ * Specify whether the cnonce used for SIP authentication contain digits only.
+ * The "cnonce" value is setup using GUID generator, i.e:
+ * pj_create_unique_string(), and the GUID string may contain hyphen character
+ * ("-"). Some SIP servers do not like this GUID format, so this option will
+ * strip any hyphens from the GUID string.
+ *
+ * Default is 1 (cnonce will not contain any hyphen characters).
+ */
+#ifndef PJSIP_AUTH_CNONCE_USE_DIGITS_ONLY
+#   define PJSIP_AUTH_CNONCE_USE_DIGITS_ONLY    1
+#endif
+
+/**
+ * Allow client to send multiple Authorization header when receiving multiple 
+ * WWW-Authenticate header fields. If this is disabled, the stack will send
+ * Authorization header field containing credentials that match the
+ * topmost header field.
+ *
+ * Default is 0
+ */
+#ifndef PJSIP_AUTH_ALLOW_MULTIPLE_AUTH_HEADER
+#   define PJSIP_AUTH_ALLOW_MULTIPLE_AUTH_HEADER 0
+#endif
+
+/*****************************************************************************
+ *  SIP Event framework and presence settings.
+ */
+
+/**
+ * Specify the time (in seconds) to send SUBSCRIBE to refresh client 
+ * subscription before the actual interval expires.
+ *
+ * Default: 5 seconds
+ */
+#ifndef PJSIP_EVSUB_TIME_UAC_REFRESH
+#   define PJSIP_EVSUB_TIME_UAC_REFRESH         5
+#endif
+
+
+/**
+ * Specify the time (in seconds) to send PUBLISH to refresh client 
+ * publication before the actual interval expires.
+ *
+ * Default: 5 seconds
+ */
+#ifndef PJSIP_PUBLISHC_DELAY_BEFORE_REFRESH
+#   define PJSIP_PUBLISHC_DELAY_BEFORE_REFRESH  5
+#endif
+
+
+/**
+ * Specify the time (in seconds) to wait for the final NOTIFY from the
+ * server after client has sent un-SUBSCRIBE request.
+ *
+ * Default: 5 seconds
+ */
+#ifndef PJSIP_EVSUB_TIME_UAC_TERMINATE
+#   define PJSIP_EVSUB_TIME_UAC_TERMINATE       5
+#endif
+
+
+/**
+ * Specify the time (in seconds) for client subscription to wait for another
+ * NOTIFY from the server, if it has rejected the last NOTIFY with non-2xx
+ * final response (such as 401). If further NOTIFY is not received within
+ * this period, the client will unsubscribe.
+ *
+ * Default: 5 seconds
+ */
+#ifndef PJSIP_EVSUB_TIME_UAC_WAIT_NOTIFY
+#   define PJSIP_EVSUB_TIME_UAC_WAIT_NOTIFY     5
+#endif
+
+
+/**
+ * Specify the default expiration time for presence event subscription, for
+ * both client and server subscription. For client subscription, application
+ * can override this by specifying positive non-zero value in "expires" 
+ * parameter when calling #pjsip_pres_initiate(). For server subscription,
+ * we would take the expiration value from the Expires header sent by client
+ * in the SUBSCRIBE request if the header exists and its value is less than 
+ * this setting, otherwise this setting will be used.
+ *
+ * Default: 600 seconds (10 minutes)
+ */
+#ifndef PJSIP_PRES_DEFAULT_EXPIRES
+#   define PJSIP_PRES_DEFAULT_EXPIRES           600
+#endif
+
+
+/**
+ * Specify the status code value to respond to bad message body in NOTIFY
+ * request for presence. Scenarios that are considered bad include non-
+ * PIDF/XML and non-XPIDF/XML body, multipart message bodies without PIDF/XML
+ * nor XPIDF/XML part, and bad (parsing error) PIDF and X-PIDF bodies
+ * themselves.
+ *
+ * Default value is 488. Application may change this to 200 to ignore the
+ * unrecognised content (this is useful if the application wishes to handle
+ * the content itself). Only non-3xx final response code is allowed here.
+ *
+ * Default: 488 (Not Acceptable Here)
+ */
+#ifndef PJSIP_PRES_BAD_CONTENT_RESPONSE
+#   define PJSIP_PRES_BAD_CONTENT_RESPONSE      488
+#endif
+
+
+/**
+ * Add "timestamp" information in generated PIDF document for both server
+ * subscription and presence publication.
+ *
+ * Default: 1 (yes)
+ */
+#ifndef PJSIP_PRES_PIDF_ADD_TIMESTAMP
+#   define PJSIP_PRES_PIDF_ADD_TIMESTAMP        1
+#endif
+
+
+/**
+ * Default session interval for Session Timer (RFC 4028) extension, in
+ * seconds. As specified in RFC 4028 Section 4, this value must not be 
+ * less than the absolute minimum for the Session-Expires header field
+ * 90 seconds, and the recommended value is 1800 seconds.
+ *
+ * Default: 1800 seconds
+ */
+#ifndef PJSIP_SESS_TIMER_DEF_SE
+#   define PJSIP_SESS_TIMER_DEF_SE              1800
+#endif
+
+
+/**
+ * Default delay for retrying session refresh request upon
+ * receiving transport error (503). Set it to -1 to end the session
+ * immediately instead.
+ *
+ * Default: 10 seconds
+ */
+#ifndef PJSIP_SESS_TIMER_RETRY_DELAY
+#   define PJSIP_SESS_TIMER_RETRY_DELAY         10
+#endif
+
+
+/**
+ * Specify whether the client publication session should queue the
+ * PUBLISH request should there be another PUBLISH transaction still
+ * pending. If this is set to false, the client will return error
+ * on the PUBLISH request if there is another PUBLISH transaction still
+ * in progress.
+ *
+ * Default: 1 (yes)
+ */
+#ifndef PJSIP_PUBLISHC_QUEUE_REQUEST
+#   define PJSIP_PUBLISHC_QUEUE_REQUEST         1
+#endif
+
+
+/**
+ * Specify the default expiration time for Message Waiting Indication
+ * (RFC 3842) event subscription, for both client and server subscription.
+ * For client subscription, application can override this by specifying
+ * positive non-zero value in "expires" parameter when calling
+ * #pjsip_mwi_initiate(). For server subscription, we would take the
+ * expiration value from the Expires header sent by client in the SUBSCRIBE
+ * request if the header exists and its value is less than  this setting,
+ * otherwise this setting will be used.
+ *
+ * Default: 3600 seconds
+ */
+#ifndef PJSIP_MWI_DEFAULT_EXPIRES
+#   define PJSIP_MWI_DEFAULT_EXPIRES            3600
+#endif
+
+
+/**
+ * Specify whether transport manager should maintain a list of transmit
+ * buffer instances, so any possible dangling instance can be cleaned up
+ * when the transport manager is shutdown (see also ticket #1671).
+ * Note that this feature will have slight impact on the performance as
+ * mutex is employed in updating the list, i.e: on creation and destruction
+ * of transmit data.
+ *
+ * Default: 0 (no)
+ */
+#ifndef PJSIP_HAS_TX_DATA_LIST
+#   define PJSIP_HAS_TX_DATA_LIST               0
+#endif
+
+/** 
+ * Specify whether to accept INVITE/re-INVITE with unknown content type,
+ * by default the stack will reject this type of message as specified in 
+ * RFC3261 section 8.2.3.
+ * Application that wishes to process the body could set this to PJ_TRUE,
+ * be informed that SDP offer/answer will still be present.
+ *
+ * Default: PJ_FALSE
+ */
+#ifndef PJSIP_INV_ACCEPT_UNKNOWN_BODY
+#   define PJSIP_INV_ACCEPT_UNKNOWN_BODY    PJ_FALSE
+#endif
+
+/** 
+ * Specify whether to check if UPDATE sent in EARLY state has already
+ * completed SDP negotiation using reliable provisional responses, as
+ * specified in RFC3311 section 5.1.
+ *
+ * By default, the library will disable the check and allow the UPDATE
+ * to be sent for backward compatibility.
+ *
+ * Default: 0 (disabled)
+ */
+#ifndef PJSIP_INV_UPDATE_EARLY_CHECK_RELIABLE
+#   define PJSIP_INV_UPDATE_EARLY_CHECK_RELIABLE    0
+#endif
+
+/**
+ * Dump configuration to log with verbosity equal to info(3).
+ */
+PJ_DECL(void) pjsip_dump_config(void);
+
+PJ_END_DECL
+
+/**
+ * @}
+ */
+
+
+#include <pj/config.h>
+
+
+#endif  /* __PJSIP_SIP_CONFIG_H__ */
+