@voipsdk/react-native-voipcloud 1.0.1 → 1.0.2

package/ios/VoipCloud.xcframework/ios-arm64_x86_64-simulator/Headers/pjmedia/config.h

@@ -0,0 +1,1713 @@
+/* 
+ * 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 __PJMEDIA_CONFIG_H__
+#define __PJMEDIA_CONFIG_H__
+
+/**
+ * @file pjmedia/config.h Compile time config
+ * @brief Contains some compile time constants.
+ */
+#include <pj/config.h>
+
+/**
+ * @defgroup PJMEDIA_BASE Base Types and Configurations
+ */
+
+/**
+ * @defgroup PJMEDIA_CONFIG Compile time configuration
+ * @ingroup PJMEDIA_BASE
+ * @brief Some compile time configuration settings.
+ * @{
+ */
+
+/*
+ * Include config_auto.h if autoconf is used (PJ_AUTOCONF is set)
+ */
+#if defined(PJ_AUTOCONF)
+#   include <pjmedia/config_auto.h>
+#endif
+
+/**
+ * Initial memory block for media endpoint.
+ */
+#ifndef PJMEDIA_POOL_LEN_ENDPT
+#   define PJMEDIA_POOL_LEN_ENDPT               8000
+#endif
+
+/**
+ * Memory increment for media endpoint.
+ */
+#ifndef PJMEDIA_POOL_INC_ENDPT
+#   define PJMEDIA_POOL_INC_ENDPT               4000
+#endif
+
+/**
+ * Initial memory block for event manager.
+ */
+#ifndef PJMEDIA_POOL_LEN_EVTMGR
+#   define PJMEDIA_POOL_LEN_EVTMGR              500
+#endif
+
+/**
+ * Memory increment for evnt manager.
+ */
+#ifndef PJMEDIA_POOL_INC_EVTMGR
+#   define PJMEDIA_POOL_INC_EVTMGR              500
+#endif
+
+/**
+ * Specify whether we prefer to use audio switch board rather than 
+ * conference bridge.
+ *
+ * Audio switch board is a kind of simplified version of conference 
+ * bridge, but not really the subset of conference bridge. It has 
+ * stricter rules on audio routing among the pjmedia ports and has
+ * no audio mixing capability. The power of it is it could work with
+ * encoded audio frames where conference brigde couldn't.
+ *
+ * Default: 0
+ */
+#ifndef PJMEDIA_CONF_USE_SWITCH_BOARD
+#   define PJMEDIA_CONF_USE_SWITCH_BOARD    0
+#endif
+
+/**
+ * Specify buffer size for audio switch board, in bytes. This buffer will
+ * be used for transmitting/receiving audio frame data (and some overheads,
+ * i.e: pjmedia_frame structure) among conference ports in the audio
+ * switch board. For example, if a port uses PCM format @44100Hz mono
+ * and frame time 20ms, the PCM audio data will require 1764 bytes,
+ * so with overhead, a safe buffer size will be ~1900 bytes.
+ *
+ * Default: PJMEDIA_MAX_MTU
+ */
+#ifndef PJMEDIA_CONF_SWITCH_BOARD_BUF_SIZE
+#   define PJMEDIA_CONF_SWITCH_BOARD_BUF_SIZE    PJMEDIA_MAX_MTU
+#endif
+
+/**
+ * Specify whether the conference bridge uses AGC, an automatic adjustment to
+ * avoid dramatic change in the signal level which can cause noise.
+ *
+ * Default: 1 (enabled)
+ */
+#ifndef PJMEDIA_CONF_USE_AGC
+#   define PJMEDIA_CONF_USE_AGC             1
+#endif
+
+
+/*
+ * Types of sound stream backends.
+ */
+
+/**
+ * This macro has been deprecated in releasee 1.1. Please see
+ * http://trac.pjsip.org/repos/wiki/Audio_Dev_API for more information.
+ */
+#if defined(PJMEDIA_SOUND_IMPLEMENTATION)
+#   error PJMEDIA_SOUND_IMPLEMENTATION has been deprecated
+#endif
+
+/**
+ * This macro has been deprecated in releasee 1.1. Please see
+ * http://trac.pjsip.org/repos/wiki/Audio_Dev_API for more information.
+ */
+#if defined(PJMEDIA_PREFER_DIRECT_SOUND)
+#   error PJMEDIA_PREFER_DIRECT_SOUND has been deprecated
+#endif
+
+/**
+ * This macro controls whether the legacy sound device API is to be
+ * implemented, for applications that still use the old sound device
+ * API (sound.h). If this macro is set to non-zero, the sound_legacy.c
+ * will be included in the compilation. The sound_legacy.c is an
+ * implementation of old sound device (sound.h) using the new Audio
+ * Device API.
+ *
+ * Please see http://trac.pjsip.org/repos/wiki/Audio_Dev_API for more
+ * info.
+ */
+#ifndef PJMEDIA_HAS_LEGACY_SOUND_API
+#   define PJMEDIA_HAS_LEGACY_SOUND_API     1
+#endif
+
+/**
+ * Specify default sound device latency, in milisecond.
+ */
+#ifndef PJMEDIA_SND_DEFAULT_REC_LATENCY
+#   define PJMEDIA_SND_DEFAULT_REC_LATENCY  100
+#endif
+
+/**
+ * Specify default sound device latency, in milisecond. 
+ *
+ * Default is 160ms for Windows Mobile and 140ms for other platforms.
+ */
+#ifndef PJMEDIA_SND_DEFAULT_PLAY_LATENCY
+#   if defined(PJ_WIN32_WINCE) && PJ_WIN32_WINCE!=0
+#       define PJMEDIA_SND_DEFAULT_PLAY_LATENCY     160
+#   else
+#       define PJMEDIA_SND_DEFAULT_PLAY_LATENCY     140
+#   endif
+#endif
+
+
+/*
+ * Types of WSOLA backend algorithm.
+ */
+
+/**
+ * This denotes implementation of WSOLA using null algorithm. Expansion
+ * will generate zero frames, and compression will just discard some
+ * samples from the input.
+ *
+ * This type of implementation may be used as it requires the least
+ * processing power.
+ */
+#define PJMEDIA_WSOLA_IMP_NULL              0
+
+/**
+ * This denotes implementation of WSOLA using fixed or floating point WSOLA
+ * algorithm. This implementation provides the best quality of the result,
+ * at the expense of one frame delay and intensive processing power 
+ * requirement.
+ */
+#define PJMEDIA_WSOLA_IMP_WSOLA             1
+
+/**
+ * This denotes implementation of WSOLA algorithm with faster waveform 
+ * similarity calculation. This implementation provides fair quality of 
+ * the result with the main advantage of low processing power requirement.
+ */
+#define PJMEDIA_WSOLA_IMP_WSOLA_LITE        2
+
+/**
+ * Specify type of Waveform based Similarity Overlap and Add (WSOLA) backend
+ * implementation to be used. WSOLA is an algorithm to expand and/or compress 
+ * audio frames without changing the pitch, and used by the delaybuf and as PLC
+ * backend algorithm.
+ *
+ * Default is PJMEDIA_WSOLA_IMP_WSOLA
+ */
+#ifndef PJMEDIA_WSOLA_IMP
+#   define PJMEDIA_WSOLA_IMP                PJMEDIA_WSOLA_IMP_WSOLA
+#endif
+
+
+/**
+ * Specify the default maximum duration of synthetic audio that is generated
+ * by WSOLA. This value should be long enough to cover burst of packet losses. 
+ * but not too long, because as the duration increases the quality would 
+ * degrade considerably.
+ *
+ * Note that this limit is only applied when fading is enabled in the WSOLA
+ * session.
+ *
+ * Default: 80
+ */
+#ifndef PJMEDIA_WSOLA_MAX_EXPAND_MSEC
+#   define PJMEDIA_WSOLA_MAX_EXPAND_MSEC    80
+#endif
+
+
+/**
+ * Specify WSOLA template length, in milliseconds. The longer the template,
+ * the smoother signal to be generated at the expense of more computation
+ * needed, since the algorithm will have to compare more samples to find
+ * the most similar pitch.
+ *
+ * Default: 5
+ */
+#ifndef PJMEDIA_WSOLA_TEMPLATE_LENGTH_MSEC
+#   define PJMEDIA_WSOLA_TEMPLATE_LENGTH_MSEC   5
+#endif
+
+
+/**
+ * Specify WSOLA algorithm delay, in milliseconds. The algorithm delay is
+ * used to merge synthetic samples with real samples in the transition
+ * between real to synthetic and vice versa. The longer the delay, the 
+ * smoother signal to be generated, at the expense of longer latency and
+ * a slighty more computation.
+ *
+ * Default: 5
+ */
+#ifndef PJMEDIA_WSOLA_DELAY_MSEC
+#   define PJMEDIA_WSOLA_DELAY_MSEC         5
+#endif
+
+
+/**
+ * Set this to non-zero to disable fade-out/in effect in the PLC when it
+ * instructs WSOLA to generate synthetic frames. The use of fading may
+ * or may not improve the quality of audio, depending on the nature of
+ * packet loss and the type of audio input (e.g. speech vs music).
+ * Disabling fading also implicitly remove the maximum limit of synthetic
+ * audio samples generated by WSOLA (see PJMEDIA_WSOLA_MAX_EXPAND_MSEC).
+ *
+ * Default: 0
+ */
+#ifndef PJMEDIA_WSOLA_PLC_NO_FADING
+#   define PJMEDIA_WSOLA_PLC_NO_FADING      0
+#endif
+
+
+/**
+ * Limit the number of calls by stream to the PLC to generate synthetic
+ * frames to this duration. If packets are still lost after this maximum
+ * duration, silence will be generated by the stream instead. Since the
+ * PLC normally should have its own limit on the maximum duration of
+ * synthetic frames to be generated (for PJMEDIA's PLC, the limit is
+ * PJMEDIA_WSOLA_MAX_EXPAND_MSEC), we can set this value to a large number
+ * to give additional flexibility should the PLC wants to do something
+ * clever with the lost frames.
+ *
+ * Default: 240 ms
+ */
+#ifndef PJMEDIA_MAX_PLC_DURATION_MSEC
+#   define PJMEDIA_MAX_PLC_DURATION_MSEC    240
+#endif
+
+
+/**
+ * Specify number of sound buffers. Larger number is better for sound
+ * stability and to accommodate sound devices that are unable to send frames
+ * in timely manner, however it would probably cause more audio delay (and 
+ * definitely will take more memory). One individual buffer is normally 10ms
+ * or 20 ms long, depending on ptime settings (samples_per_frame value).
+ *
+ * The setting here currently is used by the conference bridge, the splitter
+ * combiner port, and dsound.c.
+ *
+ * Default: (PJMEDIA_SND_DEFAULT_PLAY_LATENCY+20)/20
+ */
+#ifndef PJMEDIA_SOUND_BUFFER_COUNT
+#   define PJMEDIA_SOUND_BUFFER_COUNT       ((PJMEDIA_SND_DEFAULT_PLAY_LATENCY+20)/20)
+#endif
+
+
+/**
+ * Specify which A-law/U-law conversion algorithm to use.
+ * By default the conversion algorithm uses A-law/U-law table which gives
+ * the best performance, at the expense of 33 KBytes of static data.
+ * If this option is disabled, a smaller but slower algorithm will be used.
+ */
+#ifndef PJMEDIA_HAS_ALAW_ULAW_TABLE
+#   define PJMEDIA_HAS_ALAW_ULAW_TABLE      1
+#endif
+
+
+/**
+ * Unless specified otherwise, G711 codec is included by default.
+ */
+#ifndef PJMEDIA_HAS_G711_CODEC
+#   define PJMEDIA_HAS_G711_CODEC           1
+#endif
+
+
+/*
+ * Warn about obsolete macros.
+ *
+ * PJMEDIA_HAS_SMALL_FILTER has been deprecated in 0.7.
+ */
+#if defined(PJMEDIA_HAS_SMALL_FILTER)
+#   ifdef _MSC_VER
+#       pragma message("Warning: PJMEDIA_HAS_SMALL_FILTER macro is deprecated"\
+                       " and has no effect")
+#   else
+#       warning "PJMEDIA_HAS_SMALL_FILTER macro is deprecated and has no effect"
+#   endif
+#endif
+
+
+/*
+ * Warn about obsolete macros.
+ *
+ * PJMEDIA_HAS_LARGE_FILTER has been deprecated in 0.7.
+ */
+#if defined(PJMEDIA_HAS_LARGE_FILTER)
+#   ifdef _MSC_VER
+#       pragma message("Warning: PJMEDIA_HAS_LARGE_FILTER macro is deprecated"\
+                       " and has no effect")
+#   else
+#       warning "PJMEDIA_HAS_LARGE_FILTER macro is deprecated"
+#   endif
+#endif
+
+
+/*
+ * These macros are obsolete in 0.7.1 so it will trigger compilation error.
+ * Please use PJMEDIA_RESAMPLE_IMP to select the resample implementation
+ * to use.
+ */
+#ifdef PJMEDIA_HAS_LIBRESAMPLE
+#   error "PJMEDIA_HAS_LIBRESAMPLE macro is deprecated. Use '#define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_LIBRESAMPLE'"
+#endif
+
+#ifdef PJMEDIA_HAS_SPEEX_RESAMPLE
+#   error "PJMEDIA_HAS_SPEEX_RESAMPLE macro is deprecated. Use '#define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_SPEEX'"
+#endif
+
+
+/*
+ * Sample rate conversion backends.
+ * Select one of these backends in PJMEDIA_RESAMPLE_IMP.
+ */
+/** No resampling */
+#define PJMEDIA_RESAMPLE_NONE               1
+
+/** Sample rate conversion using libresample */
+#define PJMEDIA_RESAMPLE_LIBRESAMPLE        2
+
+/** Sample rate conversion using Speex */
+#define PJMEDIA_RESAMPLE_SPEEX              3
+
+/** Sample rate conversion using libsamplerate (a.k.a Secret Rabbit Code) */
+#define PJMEDIA_RESAMPLE_LIBSAMPLERATE      4
+
+/**
+ * Select which resample implementation to use. Currently pjmedia supports:
+ *  - #PJMEDIA_RESAMPLE_LIBRESAMPLE, to use libresample-1.7, this is the default
+ *    implementation to be used.
+ *  - #PJMEDIA_RESAMPLE_LIBSAMPLERATE, to use libsamplerate implementation
+ *    (a.k.a. Secret Rabbit Code).
+ *  - #PJMEDIA_RESAMPLE_SPEEX, to use sample rate conversion in Speex library.
+ *  - #PJMEDIA_RESAMPLE_NONE, to disable sample rate conversion. Any calls to
+ *    resample function will return error.
+ *
+ * Default is PJMEDIA_RESAMPLE_LIBRESAMPLE
+ */
+#ifndef PJMEDIA_RESAMPLE_IMP
+#   define PJMEDIA_RESAMPLE_IMP             PJMEDIA_RESAMPLE_LIBRESAMPLE
+#endif
+
+
+/**
+ * Specify whether libsamplerate, when used, should be linked statically
+ * into the application. This option is only useful for Visual Studio
+ * projects, and when this static linking is enabled
+ */
+
+
+/**
+ * Default file player/writer buffer size.
+ */
+#ifndef PJMEDIA_FILE_PORT_BUFSIZE
+#   define PJMEDIA_FILE_PORT_BUFSIZE            4000
+#endif
+
+
+/**
+ * Maximum frame duration (in msec) to be supported.
+ * This (among other thing) will affect the size of buffers to be allocated
+ * for outgoing packets.
+ */
+#ifndef PJMEDIA_MAX_FRAME_DURATION_MS   
+#   define PJMEDIA_MAX_FRAME_DURATION_MS        200
+#endif
+
+
+/**
+ * Max packet size for transmitting direction.
+ */
+#ifndef PJMEDIA_MAX_MTU                 
+#  define PJMEDIA_MAX_MTU                       1500
+#endif
+
+
+/**
+ * Max packet size for receiving direction.
+ */
+#ifndef PJMEDIA_MAX_MRU                 
+#  define PJMEDIA_MAX_MRU                       2000
+#endif
+
+
+/**
+ * DTMF/telephone-event duration, in timestamp. To specify the duration in
+ * milliseconds, use the setting PJMEDIA_DTMF_DURATION_MSEC instead.
+ *
+ * Note that for a clockrate of 8 KHz, a dtmf duration of 1600 timestamp
+ * units (the default value of PJMEDIA_DTMF_DURATION) is equivalent to 200 ms. 
+ */
+#ifndef PJMEDIA_DTMF_DURATION           
+#  define PJMEDIA_DTMF_DURATION                 1600    /* in timestamp */
+#endif
+
+
+/**
+ * DTMF/telephone-event duration, in milliseconds. If the value is greater
+ * than zero, than this setting will be used instead of PJMEDIA_DTMF_DURATION.
+ */
+#ifndef PJMEDIA_DTMF_DURATION_MSEC              
+#  define PJMEDIA_DTMF_DURATION_MSEC            200
+#endif
+
+
+/**
+ * Number of RTP packets received from different source IP address from the
+ * remote address required to make the stream switch transmission
+ * to the source address.
+ */
+#ifndef PJMEDIA_RTP_NAT_PROBATION_CNT   
+#  define PJMEDIA_RTP_NAT_PROBATION_CNT         10
+#endif
+
+
+/**
+ * Number of RTCP packets received from different source IP address from the
+ * remote address required to make the stream switch RTCP transmission
+ * to the source address.
+ */
+#ifndef PJMEDIA_RTCP_NAT_PROBATION_CNT
+#  define PJMEDIA_RTCP_NAT_PROBATION_CNT        3
+#endif
+
+
+/**
+ * Specify whether RTCP should be advertised in SDP. This setting would
+ * affect whether RTCP candidate will be added in SDP when ICE is used.
+ * Application might want to disable RTCP advertisement in SDP to
+ * reduce the message size.
+ *
+ * Default: 1 (yes)
+ */
+#ifndef PJMEDIA_ADVERTISE_RTCP
+#   define PJMEDIA_ADVERTISE_RTCP               1
+#endif
+
+
+/**
+ * Interval to send regular RTCP packets, in msec.
+ */
+#ifndef PJMEDIA_RTCP_INTERVAL
+#   define PJMEDIA_RTCP_INTERVAL                5000    /* msec*/
+#endif
+
+
+/**
+ * Minimum interval between two consecutive outgoing RTCP-FB packets,
+ * such as Picture Loss Indication, in msec.
+ */
+#ifndef PJMEDIA_RTCP_FB_INTERVAL
+#   define PJMEDIA_RTCP_FB_INTERVAL             50      /* msec*/
+#endif
+
+
+/**
+ * Tell RTCP to ignore the first N packets when calculating the
+ * jitter statistics. From experimentation, the first few packets
+ * (25 or so) have relatively big jitter, possibly because during
+ * this time, the program is also busy setting up the signaling,
+ * so they make the average jitter big.
+ *
+ * Default: 25.
+ */
+#ifndef PJMEDIA_RTCP_IGNORE_FIRST_PACKETS
+#   define  PJMEDIA_RTCP_IGNORE_FIRST_PACKETS   25
+#endif
+
+
+/**
+ * Specify whether RTCP statistics includes raw jitter statistics.
+ * Raw jitter is defined as absolute value of network transit time
+ * difference of two consecutive packets; refering to "difference D"
+ * term in interarrival jitter calculation in RFC 3550 section 6.4.1.
+ *
+ * Default: 0 (no).
+ */
+#ifndef PJMEDIA_RTCP_STAT_HAS_RAW_JITTER
+#   define PJMEDIA_RTCP_STAT_HAS_RAW_JITTER     0
+#endif
+
+/**
+ * Specify the factor with wich RTCP RTT statistics should be normalized 
+ * if exceptionally high. For e.g. mobile networks with potentially large
+ * fluctuations, this might be unwanted.
+ *
+ * Use (0) to disable this feature.
+ *
+ * Default: 3.
+ */
+#ifndef PJMEDIA_RTCP_NORMALIZE_FACTOR
+#   define PJMEDIA_RTCP_NORMALIZE_FACTOR        3
+#endif
+
+
+/**
+ * Specify whether RTCP statistics includes IP Delay Variation statistics.
+ * IPDV is defined as network transit time difference of two consecutive
+ * packets. The IPDV statistic can be useful to inspect clock skew existance
+ * and level, e.g: when the IPDV mean values were stable in positive numbers,
+ * then the remote clock (used in sending RTP packets) is faster than local
+ * system clock. Ideally, the IPDV mean values are always equal to 0.
+ *
+ * Default: 0 (no).
+ */
+#ifndef PJMEDIA_RTCP_STAT_HAS_IPDV
+#   define PJMEDIA_RTCP_STAT_HAS_IPDV           0
+#endif
+
+
+/**
+ * Specify whether RTCP XR support should be built into PJMEDIA. Disabling
+ * this feature will reduce footprint slightly. Note that even when this 
+ * setting is enabled, RTCP XR processing will only be performed in stream 
+ * if it is enabled on run-time on per stream basis. See  
+ * PJMEDIA_STREAM_ENABLE_XR setting for more info.
+ *
+ * Default: 0 (no).
+ */
+#ifndef PJMEDIA_HAS_RTCP_XR
+#   define PJMEDIA_HAS_RTCP_XR                  0
+#endif
+
+
+/**
+ * The RTCP XR feature is activated and used by stream if \a enable_rtcp_xr
+ * field of \a pjmedia_stream_info structure is non-zero. This setting 
+ * controls the default value of this field.
+ *
+ * Default: 0 (disabled)
+ */
+#ifndef PJMEDIA_STREAM_ENABLE_XR
+#   define PJMEDIA_STREAM_ENABLE_XR             0
+#endif
+
+
+/**
+ * Specify the buffer length for storing any received RTCP SDES text
+ * in a stream session. Usually RTCP contains only the mandatory SDES
+ * field, i.e: CNAME.
+ * 
+ * Default: 64 bytes.
+ */
+#ifndef PJMEDIA_RTCP_RX_SDES_BUF_LEN
+#   define PJMEDIA_RTCP_RX_SDES_BUF_LEN         64
+#endif
+
+
+/**
+ * Specify the maximum number of RTCP Feedback capability definition.
+ * 
+ * Default: 16
+ */
+#ifndef PJMEDIA_RTCP_FB_MAX_CAP
+#   define PJMEDIA_RTCP_FB_MAX_CAP              16
+#endif
+
+
+/**
+ * Specify how long (in miliseconds) the stream should suspend the
+ * silence detector/voice activity detector (VAD) during the initial
+ * period of the session. This feature is useful to open bindings in
+ * all NAT routers between local and remote endpoint since most NATs
+ * do not allow incoming packet to get in before local endpoint sends
+ * outgoing packets.
+ *
+ * Specify zero to disable this feature.
+ *
+ * Default: 600 msec (which gives good probability that some RTP 
+ *                    packets will reach the destination, but without
+ *                    filling up the jitter buffer on the remote end).
+ */
+#ifndef PJMEDIA_STREAM_VAD_SUSPEND_MSEC
+#   define PJMEDIA_STREAM_VAD_SUSPEND_MSEC      600
+#endif
+
+/**
+ * Perform RTP payload type checking in the audio stream. Normally the peer
+ * MUST send RTP with payload type as we specified in our SDP. Certain
+ * agents may not be able to follow this hence the only way to have
+ * communication is to disable this check.
+ *
+ * Default: 1
+ */
+#ifndef PJMEDIA_STREAM_CHECK_RTP_PT
+#   define PJMEDIA_STREAM_CHECK_RTP_PT          1
+#endif
+
+/**
+ * Reserve some space for application extra data, e.g: SRTP auth tag,
+ * in RTP payload, so the total payload length will not exceed the MTU.
+ */
+#ifndef PJMEDIA_STREAM_RESV_PAYLOAD_LEN
+#   define PJMEDIA_STREAM_RESV_PAYLOAD_LEN      20
+#endif
+
+
+/**
+ * Specify the maximum duration of silence period in the codec, in msec. 
+ * This is useful for example to keep NAT binding open in the firewall
+ * and to prevent server from disconnecting the call because no 
+ * RTP packet is received.
+ *
+ * This only applies to codecs that use PJMEDIA's VAD such as G711, GSM,
+ * iLBC, G722, G722.1, L16. Some other codecs, such as Speex, Opus, G729,
+ * have their own VAD/DTX mechanism will not be affected by this setting.
+ *
+ * Use (-1) to disable this feature.
+ *
+ * Default: 5000 ms
+ */
+#ifndef PJMEDIA_CODEC_MAX_SILENCE_PERIOD
+#   define PJMEDIA_CODEC_MAX_SILENCE_PERIOD     5000
+#endif
+
+
+/**
+ * Suggested or default threshold to be set for fixed silence detection
+ * or as starting threshold for adaptive silence detection. The threshold
+ * has the range from zero to 0xFFFF.
+ */
+#ifndef PJMEDIA_SILENCE_DET_THRESHOLD
+#   define PJMEDIA_SILENCE_DET_THRESHOLD        4
+#endif
+
+
+/**
+ * Maximum silence threshold in the silence detector. The silence detector
+ * will not cut the audio transmission if the audio level is above this
+ * level.
+ *
+ * Use 0x10000 (or greater) to disable this feature.
+ *
+ * Default: 0x10000 (disabled)
+ */
+#ifndef PJMEDIA_SILENCE_DET_MAX_THRESHOLD
+#   define PJMEDIA_SILENCE_DET_MAX_THRESHOLD    0x10000
+#endif
+
+
+/**
+ * Speex Accoustic Echo Cancellation (AEC).
+ * By default is enabled.
+ */
+#ifndef PJMEDIA_HAS_SPEEX_AEC
+#   define PJMEDIA_HAS_SPEEX_AEC                1
+#endif
+
+
+/**
+ * Specify whether Automatic Gain Control (AGC) should also be enabled in
+ * Speex AEC.
+ *
+ * Default: 1 (yes)
+ */
+#ifndef PJMEDIA_SPEEX_AEC_USE_AGC
+#   define PJMEDIA_SPEEX_AEC_USE_AGC            1
+#endif
+
+
+/**
+ * Specify whether denoise should also be enabled in Speex AEC.
+ *
+ * Default: 1 (yes)
+ */
+#ifndef PJMEDIA_SPEEX_AEC_USE_DENOISE
+#   define PJMEDIA_SPEEX_AEC_USE_DENOISE        1
+#endif
+
+
+/**
+ * WebRtc Accoustic Echo Cancellation (AEC).
+ * By default is disabled.
+ */
+#ifndef PJMEDIA_HAS_WEBRTC_AEC
+#   define PJMEDIA_HAS_WEBRTC_AEC               0
+#endif
+
+/**
+ * Specify whether WebRtc EC should use its mobile version AEC.
+ *
+ * Default: 0 (no)
+ */
+#ifndef PJMEDIA_WEBRTC_AEC_USE_MOBILE
+#   define PJMEDIA_WEBRTC_AEC_USE_MOBILE        0
+#endif
+
+
+/**
+ * Maximum number of parameters in SDP fmtp attribute.
+ *
+ * Default: 16
+ */
+#ifndef PJMEDIA_CODEC_MAX_FMTP_CNT
+#   define PJMEDIA_CODEC_MAX_FMTP_CNT           16
+#endif
+
+
+/**
+ * This specifies the behavior of the SDP negotiator when responding to an
+ * offer, whether it should rather use the codec preference as set by
+ * remote, or should it rather use the codec preference as specified by
+ * local endpoint.
+ *
+ * For example, suppose incoming call has codec order "8 0 3", while 
+ * local codec order is "3 0 8". If remote codec order is preferable,
+ * the selected codec will be 8, while if local codec order is preferable,
+ * the selected codec will be 3.
+ *
+ * If set to non-zero, the negotiator will use the codec order as specified
+ * by remote in the offer.
+ *
+ * Note that this behavior can be changed during run-time by calling
+ * pjmedia_sdp_neg_set_prefer_remote_codec_order().
+ *
+ * Default is 1 (to maintain backward compatibility)
+ */
+#ifndef PJMEDIA_SDP_NEG_PREFER_REMOTE_CODEC_ORDER
+#   define PJMEDIA_SDP_NEG_PREFER_REMOTE_CODEC_ORDER    1
+#endif
+
+/**
+ * This specifies the behavior of the SDP negotiator when responding to an
+ * offer, whether it should answer with multiple formats or not.
+ *
+ * Note that this behavior can be changed during run-time by calling
+ * pjmedia_sdp_neg_set_allow_multiple_codecs().
+ *
+ * Default is 0 (to maintain backward compatibility)
+ */
+#ifndef PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS
+#   define PJMEDIA_SDP_NEG_ANSWER_MULTIPLE_CODECS       0
+#endif
+
+
+/**
+ * This specifies the maximum number of the customized SDP format
+ * negotiation callbacks.
+ */
+#ifndef PJMEDIA_SDP_NEG_MAX_CUSTOM_FMT_NEG_CB
+#   define PJMEDIA_SDP_NEG_MAX_CUSTOM_FMT_NEG_CB        8
+#endif
+
+
+/**
+ * This specifies if the SDP negotiator should rewrite answer payload
+ * type numbers to use the same payload type numbers as the remote offer
+ * for all matched codecs.
+ *
+ * Default is 1 (yes)
+ */
+#ifndef PJMEDIA_SDP_NEG_ANSWER_SYMMETRIC_PT
+#   define PJMEDIA_SDP_NEG_ANSWER_SYMMETRIC_PT          1
+#endif
+
+/**
+ * The SDP negotiator will maintain that the mapping from a particular
+ * dynamic payload type number to a particular codec does not change,
+ * as mandated by RFC 3264 section 8.3.2.
+ * By default, the mapping is maintained for local endpoint only, i.e.
+ * it only takes into account local offer and local answer.
+ * Enable this if application wishes to maintain PT->codec mapping for
+ * remote endpoint as well, i.e. to update the mapping based on remote
+ * offer and answer too.
+ *
+ * Default is 0 (no)
+ */
+#ifndef PJMEDIA_SDP_NEG_MAINTAIN_REMOTE_PT_MAP
+#   define PJMEDIA_SDP_NEG_MAINTAIN_REMOTE_PT_MAP       0
+#endif
+
+
+/**
+ * This specifies if the SDP negotiator should compare its content before 
+ * incrementing the origin version on the subsequent offer/answer. 
+ * If this is set to 1, origin version will only by incremented if the 
+ * new offer/answer is different than the previous one. For backward 
+ * compatibility and performance this is set to 0.
+ *
+ * Default is 0 (No)
+ * 
+ * This macro has been deprecated in version 2.14.
+ * See https://github.com/pjsip/pjproject/pull/3322 for more info.
+ */
+#ifdef PJMEDIA_SDP_NEG_COMPARE_BEFORE_INC_VERSION
+#   ifdef _MSC_VER
+#       pragma message("Warning: PJMEDIA_SDP_NEG_COMPARE_BEFORE_INC_VERSION macro is"\
+                       " deprecated and has no effect")
+#   else
+#       warning "PJMEDIA_SDP_NEG_COMPARE_BEFORE_INC_VERSION macro is deprecated"\
+                " and has no effect"
+#   endif
+#endif
+
+
+/**
+ * Support for sending and decoding RTCP port in SDP (RFC 3605).
+ * Default is equal to PJMEDIA_ADVERTISE_RTCP setting.
+ */
+#ifndef PJMEDIA_HAS_RTCP_IN_SDP
+#   define PJMEDIA_HAS_RTCP_IN_SDP              (PJMEDIA_ADVERTISE_RTCP)
+#endif
+
+
+/**
+ * This macro controls whether pjmedia should include SDP
+ * bandwidth modifier "TIAS" (RFC3890).
+ *
+ * Note that there is also a run-time variable to turn this setting
+ * on or off, defined in endpoint.c. To access this variable, use
+ * the following construct
+ *
+ \verbatim
+    extern pj_bool_t pjmedia_add_bandwidth_tias_in_sdp;
+
+    // Do not enable bandwidth information inclusion in sdp
+    pjmedia_add_bandwidth_tias_in_sdp = PJ_FALSE;
+ \endverbatim
+ *
+ * Default: 1 (yes)
+ */
+#ifndef PJMEDIA_ADD_BANDWIDTH_TIAS_IN_SDP
+#   define PJMEDIA_ADD_BANDWIDTH_TIAS_IN_SDP    1
+#endif
+
+
+/**
+ * This macro controls whether pjmedia should include SDP rtpmap 
+ * attribute for static payload types. SDP rtpmap for static
+ * payload types are optional, although they are normally included
+ * for interoperability reason.
+ *
+ * Note that there is also a run-time variable to turn this setting
+ * on or off, defined in endpoint.c. To access this variable, use
+ * the following construct
+ *
+ \verbatim
+    extern pj_bool_t pjmedia_add_rtpmap_for_static_pt;
+
+    // Do not include rtpmap for static payload types (<96)
+    pjmedia_add_rtpmap_for_static_pt = PJ_FALSE;
+ \endverbatim
+ *
+ * Default: 1 (yes)
+ */
+#ifndef PJMEDIA_ADD_RTPMAP_FOR_STATIC_PT
+#   define PJMEDIA_ADD_RTPMAP_FOR_STATIC_PT     1
+#endif
+
+
+/**
+ * This macro declares the start payload type for telephone-event
+ * that is advertised by PJMEDIA for outgoing SDP. If this macro
+ * is set to zero, telephone events would not be advertised nor
+ * supported.
+ */
+#ifndef PJMEDIA_RTP_PT_TELEPHONE_EVENTS
+#   define PJMEDIA_RTP_PT_TELEPHONE_EVENTS          120
+#endif
+
+
+/**
+ * This macro declares whether PJMEDIA should generate multiple
+ * telephone-event formats in SDP offer, i.e: one for each audio codec
+ * clock rate (see also ticket #2088). If this macro is set to zero, only
+ * one telephone event format will be generated and it uses clock rate 8kHz
+ * (old behavior before ticket #2088).
+ *
+ * Default: 1 (yes)
+ */
+#ifndef PJMEDIA_TELEPHONE_EVENT_ALL_CLOCKRATES
+#   define PJMEDIA_TELEPHONE_EVENT_ALL_CLOCKRATES   1
+#endif
+
+
+/**
+ * Maximum tones/digits that can be enqueued in the tone generator.
+ */
+#ifndef PJMEDIA_TONEGEN_MAX_DIGITS
+#   define PJMEDIA_TONEGEN_MAX_DIGITS               32
+#endif
+
+
+/* 
+ * Below specifies the various tone generator backend algorithm.
+ */
+
+/** 
+ * The math's sine(), floating point. This has very good precision 
+ * but it's the slowest and requires floating point support and
+ * linking with the math library.
+ */
+#define PJMEDIA_TONEGEN_SINE                        1
+
+/**
+ * Floating point approximation of sine(). This has relatively good
+ * precision and much faster than plain sine(), but it requires floating-
+ * point support and linking with the math library.
+ */
+#define PJMEDIA_TONEGEN_FLOATING_POINT              2
+
+/**
+ * Fixed point using sine signal generated by Cordic algorithm. This
+ * algorithm can be tuned to provide balance between precision and
+ * performance by tuning the PJMEDIA_TONEGEN_FIXED_POINT_CORDIC_LOOP 
+ * setting, and may be suitable for platforms that lack floating-point
+ * support.
+ */
+#define PJMEDIA_TONEGEN_FIXED_POINT_CORDIC          3
+
+/**
+ * Fast fixed point using some approximation to generate sine waves.
+ * The tone generated by this algorithm is not very precise, however
+ * the algorithm is very fast.
+ */
+#define PJMEDIA_TONEGEN_FAST_FIXED_POINT            4
+
+
+/**
+ * Specify the tone generator algorithm to be used. Please see 
+ * http://trac.pjsip.org/repos/wiki/Tone_Generator for the performance
+ * analysis results of the various tone generator algorithms.
+ *
+ * Default value:
+ *  - PJMEDIA_TONEGEN_FLOATING_POINT when PJ_HAS_FLOATING_POINT is set
+ *  - PJMEDIA_TONEGEN_FIXED_POINT_CORDIC when PJ_HAS_FLOATING_POINT is not set
+ */
+#ifndef PJMEDIA_TONEGEN_ALG
+#   if defined(PJ_HAS_FLOATING_POINT) && PJ_HAS_FLOATING_POINT
+#       define PJMEDIA_TONEGEN_ALG      PJMEDIA_TONEGEN_FLOATING_POINT
+#   else
+#       define PJMEDIA_TONEGEN_ALG      PJMEDIA_TONEGEN_FIXED_POINT_CORDIC
+#   endif
+#endif
+
+
+/**
+ * Specify the number of calculation loops to generate the tone, when
+ * PJMEDIA_TONEGEN_FIXED_POINT_CORDIC algorithm is used. With more calculation
+ * loops, the tone signal gets more precise, but this will add more 
+ * processing.
+ *
+ * Valid values are 1 to 28.
+ *
+ * Default value: 10
+ */
+#ifndef PJMEDIA_TONEGEN_FIXED_POINT_CORDIC_LOOP
+#   define PJMEDIA_TONEGEN_FIXED_POINT_CORDIC_LOOP  10
+#endif
+
+
+/**
+ * Enable high quality of tone generation, the better quality will cost
+ * more CPU load. This is only applied to floating point enabled machines.
+ *
+ * By default it is enabled when PJ_HAS_FLOATING_POINT is set.
+ *
+ * This macro has been deprecated in version 1.0-rc3.
+ */
+#ifdef PJMEDIA_USE_HIGH_QUALITY_TONEGEN
+#   error   "The PJMEDIA_USE_HIGH_QUALITY_TONEGEN macro is obsolete"
+#endif
+
+
+/**
+ * Fade-in duration for the tone, in milliseconds. Set to zero to disable
+ * this feature.
+ *
+ * Default: 1 (msec)
+ */
+#ifndef PJMEDIA_TONEGEN_FADE_IN_TIME
+#   define PJMEDIA_TONEGEN_FADE_IN_TIME             1
+#endif
+
+
+/**
+ * Fade-out duration for the tone, in milliseconds. Set to zero to disable
+ * this feature.
+ *
+ * Default: 2 (msec)
+ */
+#ifndef PJMEDIA_TONEGEN_FADE_OUT_TIME
+#   define PJMEDIA_TONEGEN_FADE_OUT_TIME            2
+#endif
+
+
+/**
+ * The default tone generator amplitude (1-32767).
+ *
+ * Default value: 12288
+ */
+#ifndef PJMEDIA_TONEGEN_VOLUME
+#   define PJMEDIA_TONEGEN_VOLUME                   12288
+#endif
+
+
+/**
+ * Enable support for SRTP media transport. This will require linking
+ * with libsrtp from the third_party directory.
+ *
+ * By default it is enabled.
+ */
+#ifndef PJMEDIA_HAS_SRTP
+#   define PJMEDIA_HAS_SRTP                         1
+#endif
+
+
+/**
+ * Enable session description for SRTP keying.
+ *
+ * By default it is enabled.
+ */
+#ifndef PJMEDIA_SRTP_HAS_SDES
+#   define PJMEDIA_SRTP_HAS_SDES                    1
+#endif
+
+
+/**
+ * Enable DTLS for SRTP keying.
+ *
+ * Default value: 0 (disabled)
+ */
+#ifndef PJMEDIA_SRTP_HAS_DTLS
+#   define PJMEDIA_SRTP_HAS_DTLS                    0
+#endif
+
+
+/**
+ * Set OpenSSL ciphers for DTLS-SRTP.
+ *
+ * Default value: "DEFAULT"
+ */
+#ifndef PJMEDIA_SRTP_DTLS_OSSL_CIPHERS
+#   define PJMEDIA_SRTP_DTLS_OSSL_CIPHERS           "DEFAULT"
+#endif
+
+
+/**
+ * Maximum number of SRTP cryptos.
+ *
+ * Default: 16
+ */
+#ifndef PJMEDIA_SRTP_MAX_CRYPTOS
+#   define PJMEDIA_SRTP_MAX_CRYPTOS                 16
+#endif
+
+
+/**
+ * Enable AES_CM_256 cryptos in SRTP.
+ * Default: enabled.
+ */
+#ifndef PJMEDIA_SRTP_HAS_AES_CM_256
+#   define PJMEDIA_SRTP_HAS_AES_CM_256              1
+#endif
+
+
+/**
+ * Enable AES_CM_192 cryptos in SRTP.
+ * It was reported that this crypto only works among libsrtp backends,
+ * so we recommend to disable this.
+ *
+ * To enable this, you would require OpenSSL which supports it.
+ * See https://github.com/pjsip/pjproject/issues/1943 for more info.
+ *
+ * Default: disabled.
+ */
+#ifndef PJMEDIA_SRTP_HAS_AES_CM_192
+#   define PJMEDIA_SRTP_HAS_AES_CM_192              0
+#endif
+
+
+/**
+ * Enable AES_CM_128 cryptos in SRTP.
+ * Default: enabled.
+ */
+#ifndef PJMEDIA_SRTP_HAS_AES_CM_128
+#   define PJMEDIA_SRTP_HAS_AES_CM_128              1
+#endif
+
+
+/**
+ * Enable AES_GCM_256 cryptos in SRTP.
+ *
+ * To enable this, you would require OpenSSL which supports it.
+ * See https://github.com/pjsip/pjproject/issues/1943 for more info. 
+ *
+ * Default: disabled.
+ */
+#ifndef PJMEDIA_SRTP_HAS_AES_GCM_256
+#   define PJMEDIA_SRTP_HAS_AES_GCM_256             0
+#endif
+
+
+/**
+ * Enable AES_GCM_128 cryptos in SRTP.
+ *
+ * To enable this, you would require OpenSSL which supports it.
+ * See https://github.com/pjsip/pjproject/issues/1943 for more info.
+ *
+ * Default: disabled.
+ */
+#ifndef PJMEDIA_SRTP_HAS_AES_GCM_128
+#   define PJMEDIA_SRTP_HAS_AES_GCM_128             0
+#endif
+
+
+/**
+ * Specify whether SRTP needs to handle condition that old packets with
+ * incorect RTP seq are still coming when SRTP is restarted.
+ *
+ * Default: enabled.
+ */
+#ifndef PJMEDIA_SRTP_CHECK_RTP_SEQ_ON_RESTART
+#   define PJMEDIA_SRTP_CHECK_RTP_SEQ_ON_RESTART    1
+#endif
+
+
+/**
+ * Specify whether SRTP needs to handle condition that remote may reset
+ * or maintain ROC when SRTP is restarted.
+ *
+ * Default: enabled.
+ */
+#ifndef PJMEDIA_SRTP_CHECK_ROC_ON_RESTART
+#   define PJMEDIA_SRTP_CHECK_ROC_ON_RESTART        1
+#endif
+
+
+/**
+ * Let the library handle libsrtp initialization and deinitialization.
+ * Application may want to disable this and manually perform libsrtp
+ * initialization and deinitialization when it needs to use libsrtp
+ * before the library is initialized or after the library is shutdown.
+ *
+ * By default it is enabled.
+ */
+#ifndef PJMEDIA_LIBSRTP_AUTO_INIT_DEINIT
+#   define PJMEDIA_LIBSRTP_AUTO_INIT_DEINIT         1
+#endif
+
+
+/**
+ * Enable support to handle codecs with inconsistent clock rate
+ * between clock rate in SDP/RTP & the clock rate that is actually used.
+ * This happens for example with G.722 and MPEG audio codecs.
+ * See:
+ *  - G.722      : RFC 3551 4.5.2
+ *  - MPEG audio : RFC 3551 4.5.13 & RFC 3119
+ *  - OPUS       : RFC 7587
+ *
+ * Also when this feature is enabled, some handling will be performed
+ * to deal with clock rate incompatibilities of some phones.
+ *
+ * By default it is enabled.
+ */
+#ifndef PJMEDIA_HANDLE_G722_MPEG_BUG
+#   define PJMEDIA_HANDLE_G722_MPEG_BUG             1
+#endif
+
+
+/* Setting to determine if media transport should switch RTP and RTCP
+ * remote address to the source address of the packets it receives.
+ *
+ * By default it is enabled.
+ */
+#ifndef PJMEDIA_TRANSPORT_SWITCH_REMOTE_ADDR
+#   define PJMEDIA_TRANSPORT_SWITCH_REMOTE_ADDR     1
+#endif
+
+
+/**
+ * Transport info (pjmedia_transport_info) contains a socket info and list
+ * of transport specific info, since transports can be chained together 
+ * (for example, SRTP transport uses UDP transport as the underlying 
+ * transport). This constant specifies maximum number of transport specific
+ * infos that can be held in a transport info.
+ */
+#ifndef PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXCNT
+#   define PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXCNT   4
+#endif
+
+
+/**
+ * Maximum size in bytes of storage buffer of a transport specific info.
+ */
+#ifndef PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXSIZE
+#   define PJMEDIA_TRANSPORT_SPECIFIC_INFO_MAXSIZE  (50*sizeof(long))
+#endif
+
+
+/**
+ * Value to be specified in PJMEDIA_STREAM_ENABLE_KA setting.
+ * This indicates that an empty RTP packet should be used as
+ * the keep-alive packet.
+ */
+#define PJMEDIA_STREAM_KA_EMPTY_RTP                 1
+
+/**
+ * Value to be specified in PJMEDIA_STREAM_ENABLE_KA setting.
+ * This indicates that a user defined packet should be used
+ * as the keep-alive packet. The content of the user-defined
+ * packet is specified by PJMEDIA_STREAM_KA_USER_PKT. Default
+ * content is a CR-LF packet.
+ */
+#define PJMEDIA_STREAM_KA_USER                      2
+
+/**
+ * The content of the user defined keep-alive packet. The format
+ * of the packet is initializer to pj_str_t structure. Note that
+ * the content may contain NULL character.
+ */
+#ifndef PJMEDIA_STREAM_KA_USER_PKT
+#   define PJMEDIA_STREAM_KA_USER_PKT   { "\r\n", 2 }
+#endif
+
+
+/**
+ * Specify another type of keep-alive and NAT hole punching 
+ * mechanism (the other type is PJMEDIA_STREAM_VAD_SUSPEND_MSEC
+ * and PJMEDIA_CODEC_MAX_SILENCE_PERIOD) to be used by stream. 
+ * When this feature is enabled, the stream will initially 
+ * transmit one packet to punch a hole in NAT, and periodically
+ * transmit keep-alive packets.
+ *
+ * When this alternative keep-alive mechanism is used, application
+ * may disable the other keep-alive mechanisms, i.e: by setting 
+ * PJMEDIA_STREAM_VAD_SUSPEND_MSEC to zero and 
+ * PJMEDIA_CODEC_MAX_SILENCE_PERIOD to -1.
+ *
+ * The value of this macro specifies the type of packet used
+ * for the keep-alive mechanism. Valid values are
+ * PJMEDIA_STREAM_KA_EMPTY_RTP and PJMEDIA_STREAM_KA_USER.
+ * 
+ * The duration of the keep-alive interval further can be set
+ * with PJMEDIA_STREAM_KA_INTERVAL setting.
+ *
+ * Default: 0 (disabled)
+ */
+#ifndef PJMEDIA_STREAM_ENABLE_KA
+#   define PJMEDIA_STREAM_ENABLE_KA                 0
+#endif
+
+
+/**
+ * Specify the keep-alive interval of PJMEDIA_STREAM_ENABLE_KA
+ * mechanism, in seconds.
+ *
+ * Default: 5 seconds
+ */
+#ifndef PJMEDIA_STREAM_KA_INTERVAL
+#   define PJMEDIA_STREAM_KA_INTERVAL               5
+#endif
+
+
+/**
+ * Specify the number of keep-alive needed to be sent after the stream is
+ * created.
+ *
+ * Setting this to 0 will disable it.
+ *
+ * Default : 2
+ */
+#ifndef PJMEDIA_STREAM_START_KA_CNT
+#   define PJMEDIA_STREAM_START_KA_CNT  2
+#endif
+
+
+/**
+ * Specify the interval to send keep-alive after the stream is created,
+ * in msec.
+ *
+ * Default : 1000
+ */
+#ifndef PJMEDIA_STREAM_START_KA_INTERVAL_MSEC
+#   define PJMEDIA_STREAM_START_KA_INTERVAL_MSEC  1000
+#endif
+
+
+/**
+ * Specify the number of identical consecutive error that will be ignored when 
+ * receiving RTP/RTCP data before the library tries to restart the transport.
+ *
+ * When receiving RTP/RTCP data, the library will ignore error besides 
+ * PJ_EPENDING or PJ_ECANCELLED and continue the loop to receive the data. 
+ * If the OS always return error, then the loop will continue non stop.
+ * This setting will limit the number of the identical consecutive error, 
+ * before the library start to restart the transport. If error still happens
+ * after transport restart, then PJMEDIA_EVENT_MEDIA_TP_ERR event will be 
+ * publish as a notification.
+ *
+ * If PJ_ESOCKETSTOP is raised, then transport will be restarted regardless
+ * of this setting.
+ * 
+ * To always ignore the error when receving RTP/RTCP, set this to 0.
+ *
+ * Default : 20
+ */
+#ifndef PJMEDIA_IGNORE_RECV_ERR_CNT
+#   define PJMEDIA_IGNORE_RECV_ERR_CNT          20
+#endif
+
+
+/*
+ * .... new stuffs ...
+ */
+
+/*
+ * Video
+ */
+
+/**
+ * Top level option to enable/disable video features.
+ *
+ * Default: 0 (disabled)
+ */
+#ifndef PJMEDIA_HAS_VIDEO
+#   define PJMEDIA_HAS_VIDEO                            0
+#endif
+
+
+/**
+ * Specify if FFMPEG is available. The value here will be used as the default
+ * value for other FFMPEG settings below.
+ *
+ * Default: 0
+ */
+#ifndef PJMEDIA_HAS_FFMPEG
+#   define PJMEDIA_HAS_FFMPEG                           0
+#endif
+
+/**
+ * Specify if FFMPEG libavformat is available.
+ *
+ * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
+ */
+#ifndef PJMEDIA_HAS_LIBAVFORMAT
+#   define PJMEDIA_HAS_LIBAVFORMAT                      PJMEDIA_HAS_FFMPEG
+#endif
+
+/**
+ * Specify if FFMPEG libavformat is available.
+ *
+ * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
+ */
+#ifndef PJMEDIA_HAS_LIBAVCODEC
+#   define PJMEDIA_HAS_LIBAVCODEC                       PJMEDIA_HAS_FFMPEG
+#endif
+
+/**
+ * Specify if FFMPEG libavutil is available.
+ *
+ * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
+ */
+#ifndef PJMEDIA_HAS_LIBAVUTIL
+#   define PJMEDIA_HAS_LIBAVUTIL                        PJMEDIA_HAS_FFMPEG
+#endif
+
+/**
+ * Specify if FFMPEG libswscale is available.
+ *
+ * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
+ */
+#ifndef PJMEDIA_HAS_LIBSWSCALE
+#   define PJMEDIA_HAS_LIBSWSCALE                       PJMEDIA_HAS_FFMPEG
+#endif
+
+/**
+ * Specify if FFMPEG libavdevice is available.
+ *
+ * Default: PJMEDIA_HAS_FFMPEG (or detected by configure)
+ */
+#ifndef PJMEDIA_HAS_LIBAVDEVICE
+#   define PJMEDIA_HAS_LIBAVDEVICE                      PJMEDIA_HAS_FFMPEG
+#endif
+
+/**
+ * Maximum video planes.
+ *
+ * Default: 4
+ */
+#ifndef PJMEDIA_MAX_VIDEO_PLANES
+#   define PJMEDIA_MAX_VIDEO_PLANES                     4
+#endif
+
+/**
+ * Maximum number of video formats.
+ *
+ * Default: 32
+ */
+#ifndef PJMEDIA_MAX_VIDEO_FORMATS
+#   define PJMEDIA_MAX_VIDEO_FORMATS                    32
+#endif
+
+/**
+ * Specify the maximum time difference (in ms) for synchronization between
+ * two medias. If the synchronization media source is ahead of time
+ * greater than this duration, it is considered to make a very large jump
+ * and the synchronization will be reset.
+ *
+ * Default: 20000
+ */
+#ifndef PJMEDIA_CLOCK_SYNC_MAX_SYNC_MSEC
+#   define PJMEDIA_CLOCK_SYNC_MAX_SYNC_MSEC         20000
+#endif
+
+/**
+ * Maximum video frame size.
+ * Default: 128kB
+ */
+#ifndef PJMEDIA_MAX_VIDEO_ENC_FRAME_SIZE
+#  define PJMEDIA_MAX_VIDEO_ENC_FRAME_SIZE          (1<<17)
+#endif
+
+
+/**
+ * Specify the maximum duration (in ms) for resynchronization. When a media
+ * is late to another media it is supposed to be synchronized to, it is
+ * guaranteed to be synchronized again after this duration. While if the
+ * media is ahead/early by t ms, it is guaranteed to be synchronized after
+ * t + this duration. This timing only applies if there is no additional
+ * resynchronization required during the specified duration.
+ *
+ * Default: 2000
+ */
+#ifndef PJMEDIA_CLOCK_SYNC_MAX_RESYNC_DURATION
+#   define PJMEDIA_CLOCK_SYNC_MAX_RESYNC_DURATION 2000
+#endif
+
+
+/**
+ * Minimum gap between two consecutive discards in jitter buffer,
+ * in milliseconds.
+ *
+ * Default: 200 ms
+ */
+#ifndef PJMEDIA_JBUF_DISC_MIN_GAP
+#   define PJMEDIA_JBUF_DISC_MIN_GAP                200
+#endif
+
+
+/**
+ * Minimum burst level reference used for calculating discard duration
+ * in jitter buffer progressive discard algorithm, in frames.
+ * 
+ * Default: 1 frame
+ */
+#ifndef PJMEDIA_JBUF_PRO_DISC_MIN_BURST
+#   define PJMEDIA_JBUF_PRO_DISC_MIN_BURST          1
+#endif
+
+
+/**
+ * Maximum burst level reference used for calculating discard duration
+ * in jitter buffer progressive discard algorithm, in frames.
+ * 
+ * Default: 200 frames
+ */
+#ifndef PJMEDIA_JBUF_PRO_DISC_MAX_BURST
+#   define PJMEDIA_JBUF_PRO_DISC_MAX_BURST          100
+#endif
+
+
+/**
+ * Duration for progressive discard algotithm in jitter buffer to discard
+ * an excessive frame when burst is equal to or lower than
+ * PJMEDIA_JBUF_PRO_DISC_MIN_BURST, in milliseconds.
+ *
+ * Default: 2000 ms
+ */
+#ifndef PJMEDIA_JBUF_PRO_DISC_T1
+#   define PJMEDIA_JBUF_PRO_DISC_T1                 2000
+#endif
+
+
+/**
+ * Duration for progressive discard algotithm in jitter buffer to discard
+ * an excessive frame when burst is equal to or greater than
+ * PJMEDIA_JBUF_PRO_DISC_MAX_BURST, in milliseconds.
+ *
+ * Default: 10000 ms
+ */
+#ifndef PJMEDIA_JBUF_PRO_DISC_T2
+#   define PJMEDIA_JBUF_PRO_DISC_T2                 10000
+#endif
+
+
+/**
+ * Reset jitter buffer and return silent audio on stream playback start
+ * (first get_frame()). This is useful to avoid possible noise that may be
+ * introduced by discard algorithm and neutralize latency when audio device
+ * is started later than the stream.
+ *
+ * Set this to N>0 to allow N silent audio frames returned on stream playback
+ * start, this will allow about N frames to be buffered in the jitter buffer
+ * before the playback is started (prefetching effect).
+ * Set this to zero to disable this feature.
+ *
+ * Default: 1
+ */
+#ifndef PJMEDIA_STREAM_SOFT_START
+#   define PJMEDIA_STREAM_SOFT_START                1
+#endif
+
+
+/**
+ * Video stream will discard old picture from the jitter buffer as soon as
+ * new picture is received, to reduce latency.
+ *
+ * Default: 0
+ */
+#ifndef PJMEDIA_VID_STREAM_SKIP_PACKETS_TO_REDUCE_LATENCY
+#   define PJMEDIA_VID_STREAM_SKIP_PACKETS_TO_REDUCE_LATENCY    0
+#endif
+
+
+/**
+ * Maximum video payload size. Note that this must not be greater than
+ * PJMEDIA_MAX_MTU.
+ *
+ * Default: (PJMEDIA_MAX_MTU - 20 - (128+16)) if SRTP is enabled, 
+ *          otherwise (PJMEDIA_MAX_MTU - 20). 
+ *          Note that (128+16) constant value is taken from libSRTP macro 
+ *          SRTP_MAX_TRAILER_LEN.
+ */
+#ifndef PJMEDIA_MAX_VID_PAYLOAD_SIZE
+#  if PJMEDIA_HAS_SRTP
+#     define PJMEDIA_MAX_VID_PAYLOAD_SIZE     (PJMEDIA_MAX_MTU - 20 - (128+16))
+#  else
+#     define PJMEDIA_MAX_VID_PAYLOAD_SIZE     (PJMEDIA_MAX_MTU - 20)
+#  endif
+#endif
+
+
+/**
+ * Specify target value for socket receive buffer size. It will be
+ * applied to RTP socket of media transport using setsockopt(). When
+ * transport failed to set the specified size, it will try with lower
+ * value until the highest possible is successfully set.
+ *
+ * Setting this to zero will leave the socket receive buffer size to
+ * OS default (e.g: usually 8 KB on desktop platforms).
+ *
+ * Default: 64 KB when video is enabled, otherwise zero (OS default)
+ */
+#ifndef PJMEDIA_TRANSPORT_SO_RCVBUF_SIZE
+#   if PJMEDIA_HAS_VIDEO
+#       define PJMEDIA_TRANSPORT_SO_RCVBUF_SIZE (64*1024)
+#   else
+#       define PJMEDIA_TRANSPORT_SO_RCVBUF_SIZE 0
+#   endif
+#endif
+
+
+/**
+ * Specify target value for socket send buffer size. It will be
+ * applied to RTP socket of media transport using setsockopt(). When
+ * transport failed to set the specified size, it will try with lower
+ * value until the highest possible is successfully set.
+ *
+ * Setting this to zero will leave the socket send buffer size to
+ * OS default (e.g: usually 8 KB on desktop platforms).
+ *
+ * Default: 64 KB when video is enabled, otherwise zero (OS default)
+ */
+#ifndef PJMEDIA_TRANSPORT_SO_SNDBUF_SIZE
+#   if PJMEDIA_HAS_VIDEO
+#       define PJMEDIA_TRANSPORT_SO_SNDBUF_SIZE (64*1024)
+#   else
+#       define PJMEDIA_TRANSPORT_SO_SNDBUF_SIZE 0
+#   endif
+#endif
+
+
+/**
+ * Specify if libyuv is available.
+ *
+ * Default: 0 (disable)
+ */
+#ifndef PJMEDIA_HAS_LIBYUV
+#   define PJMEDIA_HAS_LIBYUV                           0
+#endif
+
+
+/**
+ * Specify if dtmf flash in RFC 2833 is available.
+ */
+#ifndef PJMEDIA_HAS_DTMF_FLASH
+#   define PJMEDIA_HAS_DTMF_FLASH                       1
+#endif
+
+
+/**
+ * Specify the number of keyframe needed to be sent after the stream is 
+ * created. Setting this to 0 will disable it.
+ *
+ * Default : 5
+ */
+#ifndef PJMEDIA_VID_STREAM_START_KEYFRAME_CNT
+#   define PJMEDIA_VID_STREAM_START_KEYFRAME_CNT        5
+#endif
+
+
+/**
+ * Specify the interval to send keyframe after the stream is created, in msec.
+ *
+ * Default : 1000
+ */
+#ifndef PJMEDIA_VID_STREAM_START_KEYFRAME_INTERVAL_MSEC
+#   define PJMEDIA_VID_STREAM_START_KEYFRAME_INTERVAL_MSEC  1000
+#endif
+
+
+/**
+ * Specify the minimum interval to send video keyframe, in msec.
+ *
+ * Default : 1000
+ */
+#ifndef PJMEDIA_VID_STREAM_MIN_KEYFRAME_INTERVAL_MSEC
+#   define PJMEDIA_VID_STREAM_MIN_KEYFRAME_INTERVAL_MSEC    1000
+#endif
+
+
+/**
+ * Specify minimum delay of video decoding, in milliseconds. Lower value may
+ * degrade video quality significantly in a bad network environment (e.g:
+ * with persistent late and out-of-order RTP packets). Note that the value
+ * must be lower than jitter buffer maximum delay (configurable via
+ * pjmedia_stream_info.jb_max or pjsua_media_config.jb_max).
+ *
+ * Default : 100
+ */
+#ifndef PJMEDIA_VID_STREAM_DECODE_MIN_DELAY_MSEC
+#   define PJMEDIA_VID_STREAM_DECODE_MIN_DELAY_MSEC         100
+#endif
+
+
+/**
+ * Perform RTP payload type checking in the video stream. Normally the peer
+ * MUST send RTP with payload type as we specified in our SDP. Certain
+ * agents may not be able to follow this hence the only way to have
+ * communication is to disable this check.
+ *
+ * Default: PJMEDIA_STREAM_CHECK_RTP_PT (follow audio stream's setting)
+ */
+#ifndef PJMEDIA_VID_STREAM_CHECK_RTP_PT
+#   define PJMEDIA_VID_STREAM_CHECK_RTP_PT      PJMEDIA_STREAM_CHECK_RTP_PT
+#endif
+
+/**
+ * @}
+ */
+
+
+#endif  /* __PJMEDIA_CONFIG_H__ */
+
+