We are having a problem with the max_calls settings on PJSIP and Asterisk. We are stress-testing our Asterisk server, but found that we had a max 32 active call limitation on our PJSIP module. We are using PJSIP to test our Asterisk server After a quick google we found the following settings would fix the problem. |Following steps can be taken to increase number of calls supported on PJSIP: Example: If you have to increase simultaneous calls to 1000 change the following: 1. Change PJSUA_MAX_CALLS to 1000 and PJSUA_MAX_ACC to 1000 2. Change PJ_IOQUEUE_MAX_HANDLES to 2000 (double of desired number of calls). 3. Change __FD_SETSIZE to double to 2000 (double of desired number of calls). 4. Change PJSUA_MAX_PLAYERS to 1000. 5. Recompile pjsip using following steps: a. ./configure --disable-ssl --disable-sound; b. make dep c. make d. make install 6. Recompile your application with new libs.| Somehow This is not working for us; What are we doing wrong here? Anybody suggestions. Help would be higly appreciated. Our code in the config_site.php file |/* * This file contains several sample settings especially for Windows * Mobile and Symbian targets. You can include this file in your * <pj/config_site.h>file. * * The Windows Mobile and Symbian settings will be activated * automatically if you include this file. * * In addition, you may specify one of these macros (before including * this file) to activate additional settings: * * #define PJ_CONFIG_NOKIA_APS_DIRECT * Use this macro to activate the APS-Direct feature. Please see * http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct for more * info. * * #define PJ_CONFIG_WIN32_WMME_DIRECT * Configuration to activate "APS-Direct" media mode on Windows or * Windows Mobile, useful for testing purposes only. */ /* * Typical configuration for WinCE target. */ #if defined(PJ_WIN32_WINCE) && PJ_WIN32_WINCE!=0 /* * PJLIB settings. */ /* Disable floating point support */ #define PJ_HAS_FLOATING_POINT 0 /* * PJMEDIA settings */ /* Select codecs to disable */ #define PJMEDIA_HAS_L16_CODEC 0 #define PJMEDIA_HAS_ILBC_CODEC 0 /* We probably need more buffers on WM, so increase the limit */ #define PJMEDIA_SOUND_BUFFER_COUNT 32 /* Fine tune Speex's default settings for best performance/quality */ #define PJMEDIA_CODEC_SPEEX_DEFAULT_QUALITY 5 /* For CPU reason, disable speex AEC and use the echo suppressor. */ #define PJMEDIA_HAS_SPEEX_AEC 0 /* Previously, resampling is disabled due to performance reason and * this condition prevented some 'light' wideband codecs (e.g: G722.1) * to work along with narrowband codecs. Lately, some tests showed * that 16kHz <-> 8kHz resampling using libresample small filter was * affordable on ARM9 260 MHz, so here we decided to enable resampling. * Note that it is important to make sure that libresample is created * using small filter. For example PJSUA_DEFAULT_CODEC_QUALITY must * be set to 3 or 4 so pjsua-lib will apply small filter resampling. */ //#define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_NONE #define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_LIBRESAMPLE /* Use the lighter WSOLA implementation */ #define PJMEDIA_WSOLA_IMP PJMEDIA_WSOLA_IMP_WSOLA_LITE /* * PJSIP settings. */ /* Set maximum number of dialog/transaction/calls to minimum to reduce * memory usage */ #define PJSIP_MAX_TSX_COUNT 31 #define PJSIP_MAX_DIALOG_COUNT 31 #define PJSUA_MAX_CALLS 64 /* * PJSUA settings */ /* Default codec quality, previously was set to 5, however it is now * set to 4 to make sure pjsua instantiates resampler with small filter. */ #define PJSUA_DEFAULT_CODEC_QUALITY 4 /* Set maximum number of objects to minimum to reduce memory usage */ #define PJSUA_MAX_ACC 64 #define PJSUA_MAX_PLAYERS 64 #define PJSUA_MAX_RECORDERS 4 #define PJSUA_MAX_CONF_PORTS (PJSUA_MAX_CALLS+2*PJSUA_MAX_PLAYERS) #define PJSUA_MAX_BUDDIES 32 #endif /* PJ_WIN32_WINCE */ /* * Typical configuration for Symbian OS target */ #if defined(PJ_SYMBIAN) && PJ_SYMBIAN!=0 /* * PJLIB settings. */ /* Disable floating point support */ #define PJ_HAS_FLOATING_POINT 0 /* Misc PJLIB setting */ #define PJ_MAXPATH 80 /* This is important for Symbian. Symbian lacks vsnprintf(), so * if the log buffer is not long enough it's possible that * large incoming packet will corrupt memory when the log tries * to log the packet. */ #define PJ_LOG_MAX_SIZE (PJSIP_MAX_PKT_LEN+500) /* Since we don't have threads, log buffer can use static buffer * rather than stack */ #define PJ_LOG_USE_STACK_BUFFER 0 /* Disable check stack since it increases footprint */ #define PJ_OS_HAS_CHECK_STACK 0 /* * PJMEDIA settings */ /* Disable non-Symbian audio devices */ #define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0 #define PJMEDIA_AUDIO_DEV_HAS_WMME 0 /* Select codecs to disable */ #define PJMEDIA_HAS_L16_CODEC 0 #define PJMEDIA_HAS_ILBC_CODEC 0 #define PJMEDIA_HAS_G722_CODEC 0 /* Fine tune Speex's default settings for best performance/quality */ #define PJMEDIA_CODEC_SPEEX_DEFAULT_QUALITY 5 /* For CPU reason, disable speex AEC and use the echo suppressor. */ #define PJMEDIA_HAS_SPEEX_AEC 0 /* Previously, resampling is disabled due to performance reason and * this condition prevented some 'light' wideband codecs (e.g: G722.1) * to work along with narrowband codecs. Lately, some tests showed * that 16kHz <-> 8kHz resampling using libresample small filter was * affordable on ARM9 222 MHz, so here we decided to enable resampling. * Note that it is important to make sure that libresample is created * using small filter. For example PJSUA_DEFAULT_CODEC_QUALITY must * be set to 3 or 4 so pjsua-lib will apply small filter resampling. */ //#define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_NONE #define PJMEDIA_RESAMPLE_IMP PJMEDIA_RESAMPLE_LIBRESAMPLE /* Use the lighter WSOLA implementation */ #define PJMEDIA_WSOLA_IMP PJMEDIA_WSOLA_IMP_WSOLA_LITE /* We probably need more buffers especially if MDA audio backend * is used, so increase the limit */ #define PJMEDIA_SOUND_BUFFER_COUNT 32 /* * PJSIP settings. */ /* Disable safe module access, since we don't use multithreading */ #define PJSIP_SAFE_MODULE 0 /* Use large enough packet size */ #define PJSIP_MAX_PKT_LEN 2000 /* Symbian has problem with too many large blocks */ #define PJSIP_POOL_LEN_ENDPT 1000 #define PJSIP_POOL_INC_ENDPT 1000 #define PJSIP_POOL_RDATA_LEN 2000 #define PJSIP_POOL_RDATA_INC 2000 #define PJSIP_POOL_LEN_TDATA 2000 #define PJSIP_POOL_INC_TDATA 512 #define PJSIP_POOL_LEN_UA 2000 #define PJSIP_POOL_INC_UA 1000 #define PJSIP_POOL_TSX_LAYER_LEN 256 #define PJSIP_POOL_TSX_LAYER_INC 256 #define PJSIP_POOL_TSX_LEN 512 #define PJSIP_POOL_TSX_INC 128 /* * PJSUA settings. */ /* Default codec quality, previously was set to 5, however it is now * set to 4 to make sure pjsua instantiates resampler with small filter. */ #define PJSUA_DEFAULT_CODEC_QUALITY 4 /* Set maximum number of dialog/transaction/calls to minimum */ #define PJSIP_MAX_TSX_COUNT 31 #define PJSIP_MAX_DIALOG_COUNT 31 #define PJSUA_MAX_CALLS 64 /* Other pjsua settings */ #define PJSUA_MAX_ACC 64 #define PJSUA_MAX_PLAYERS 64 #define PJSUA_MAX_RECORDERS 4 #define PJSUA_MAX_CONF_PORTS (PJSUA_MAX_CALLS+2*PJSUA_MAX_PLAYERS) #define PJSUA_MAX_BUDDIES 32 #endif /* * Additional configuration to activate APS-Direct feature for * Nokia S60 target * * Please see http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct */ #ifdef PJ_CONFIG_NOKIA_APS_DIRECT /* MUST use switchboard rather than the conference bridge */ #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 /* Enable APS sound device backend and disable MDA & VAS */ #define PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA 0 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_APS 1 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_VAS 0 /* Enable passthrough codec framework */ #define PJMEDIA_HAS_PASSTHROUGH_CODECS 1 /* And selectively enable which codecs are supported by the handset */ #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 1 #endif /* * Additional configuration to activate VAS-Direct feature for * Nokia S60 target * * Please see http://trac.pjsip.org/repos/wiki/Nokia_APS_VAS_Direct */ #ifdef PJ_CONFIG_NOKIA_VAS_DIRECT /* MUST use switchboard rather than the conference bridge */ #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 /* Enable VAS sound device backend and disable MDA & APS */ #define PJMEDIA_AUDIO_DEV_HAS_SYMB_MDA 0 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_APS 0 #define PJMEDIA_AUDIO_DEV_HAS_SYMB_VAS 1 /* Enable passthrough codec framework */ #define PJMEDIA_HAS_PASSTHROUGH_CODECS 1 /* And selectively enable which codecs are supported by the handset */ #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 1 #endif /* * Configuration to activate "APS-Direct" media mode on Windows, * useful for testing purposes only. */ #ifdef PJ_CONFIG_WIN32_WMME_DIRECT /* MUST use switchboard rather than the conference bridge */ #define PJMEDIA_CONF_USE_SWITCH_BOARD 1 /* Only WMME supports the "direct" feature */ #define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0 #define PJMEDIA_AUDIO_DEV_HAS_WMME 1 /* Enable passthrough codec framework */ #define PJMEDIA_HAS_PASSTHROUGH_CODECS 1 /* Only PCMA and PCMU are supported by WMME-direct */ #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMU 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_PCMA 1 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_AMR 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_G729 0 #define PJMEDIA_HAS_PASSTHROUGH_CODEC_ILBC 0 #endif /* * iPhone sample settings. */ #if PJ_CONFIG_IPHONE /* * PJLIB settings. */ /* Both armv6 and armv7 has FP hardware support. * See https://trac.pjsip.org/repos/ticket/1589 for more info */ #define PJ_HAS_FLOATING_POINT 1 /* * PJMEDIA settings */ /* We have our own native CoreAudio backend */ #define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0 #define PJMEDIA_AUDIO_DEV_HAS_WMME 0 #define PJMEDIA_AUDIO_DEV_HAS_COREAUDIO 1 /* The CoreAudio backend has built-in echo canceller! */ #define PJMEDIA_HAS_SPEEX_AEC 0 /* Disable some codecs */ #define PJMEDIA_HAS_L16_CODEC 0 #define PJMEDIA_HAS_G722_CODEC 0 /* Use the built-in CoreAudio's iLBC codec (yay!) */ #define PJMEDIA_HAS_ILBC_CODEC 1 #define PJMEDIA_ILBC_CODEC_USE_COREAUDIO 1 /* Fine tune Speex's default settings for best performance/quality */ #define PJMEDIA_CODEC_SPEEX_DEFAULT_QUALITY 5 /* * PJSIP settings. */ /* Increase allowable packet size, just in case */ //#define PJSIP_MAX_PKT_LEN 2000 /* * PJSUA settings. */ /* Default codec quality, previously was set to 5, however it is now * set to 4 to make sure pjsua instantiates resampler with small filter. */ #define PJSUA_DEFAULT_CODEC_QUALITY 4 /* Set maximum number of dialog/transaction/calls to minimum */ #define PJSIP_MAX_TSX_COUNT 31 #define PJSIP_MAX_DIALOG_COUNT 31 #define PJSUA_MAX_CALLS 64 /* Other pjsua settings */ #define PJSUA_MAX_ACC 64 #define PJSUA_MAX_PLAYERS 64 #define PJSUA_MAX_RECORDERS 4 #define PJSUA_MAX_CONF_PORTS (PJSUA_MAX_CALLS+2*PJSUA_MAX_PLAYERS) #define PJSUA_MAX_BUDDIES 32 #endif /* * Android sample settings. */ #if PJ_CONFIG_ANDROID #define PJ_ANDROID 1 /* * PJLIB settings. */ /* Disable floating point support */ #undef PJ_HAS_FLOATING_POINT #define PJ_HAS_FLOATING_POINT 0 /* * PJMEDIA settings */ /* We have our own OpenSL ES backend */ #define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0 #define PJMEDIA_AUDIO_DEV_HAS_WMME 0 #define PJMEDIA_AUDIO_DEV_HAS_OPENSL 0 #define PJMEDIA_AUDIO_DEV_HAS_ANDROID_JNI 1 /* Disable some codecs */ #define PJMEDIA_HAS_L16_CODEC 0 #define PJMEDIA_HAS_G722_CODEC 0 /* Fine tune Speex's default settings for best performance/quality */ #define PJMEDIA_CODEC_SPEEX_DEFAULT_QUALITY 5 /* * PJSIP settings. */ /* Increase allowable packet size, just in case */ //#define PJSIP_MAX_PKT_LEN 2000 /* * PJSUA settings. */ /* Default codec quality, previously was set to 5, however it is now * set to 4 to make sure pjsua instantiates resampler with small filter. */ #define PJSUA_DEFAULT_CODEC_QUALITY 4 /* Set maximum number of dialog/transaction/calls to minimum */ #define PJSIP_MAX_TSX_COUNT 31 #define PJSIP_MAX_DIALOG_COUNT 31 #define PJSUA_MAX_CALLS 64 /* Other pjsua settings */ #define PJSUA_MAX_ACC 64 #define PJSUA_MAX_PLAYERS 64 #define PJSUA_MAX_RECORDERS 4 #define PJSUA_MAX_CONF_PORTS (PJSUA_MAX_CALLS+2*PJSUA_MAX_PLAYERS) #define PJSUA_MAX_BUDDIES 32 #endif /* * BB10 */ #if defined(PJ_CONFIG_BB10) && PJ_CONFIG_BB10 /* Quality 3 - 4 to use resampling small filter */ #define PJSUA_DEFAULT_CODEC_QUALITY 4 #define PJMEDIA_HAS_LEGACY_SOUND_API 0 #undef PJMEDIA_HAS_SPEEX_AEC #define PJMEDIA_HAS_SPEEX_AEC 0 #undef PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO #define PJMEDIA_AUDIO_DEV_HAS_PORTAUDIO 0 #endif /* * Minimum size */ #ifdef PJ_CONFIG_MINIMAL_SIZE # undef PJ_OS_HAS_CHECK_STACK # define PJ_OS_HAS_CHECK_STACK 0 # define PJ_LOG_MAX_LEVEL 0 # define PJ_ENABLE_EXTRA_CHECK 0 # define PJ_HAS_ERROR_STRING 0 # undef PJ_IOQUEUE_MAX_HANDLES /* Putting max handles to lower than 32 will make pj_fd_set_t size smaller * than native fdset_t and will trigger assertion on sock_select.c. */ # define PJ_IOQUEUE_MAX_HANDLES 128 # define PJ_CRC32_HAS_TABLES 0 # define PJSIP_MAX_TSX_COUNT 15 # define PJSIP_MAX_DIALOG_COUNT 15 # define PJSIP_UDP_SO_SNDBUF_SIZE 4000 # define PJSIP_UDP_SO_RCVBUF_SIZE 4000 # define PJMEDIA_HAS_ALAW_ULAW_TABLE 0 #elif defined(PJ_CONFIG_MAXIMUM_SPEED) # define PJ_SCANNER_USE_BITWISE 0 # undef PJ_OS_HAS_CHECK_STACK # define PJ_OS_HAS_CHECK_STACK 0 # define PJ_LOG_MAX_LEVEL 3 # define PJ_ENABLE_EXTRA_CHECK 0 # define PJ_IOQUEUE_MAX_HANDLES 5000 # define PJSIP_MAX_TSX_COUNT ((640*1024)-1) # define PJSIP_MAX_DIALOG_COUNT ((640*1024)-1) # define PJSIP_UDP_SO_SNDBUF_SIZE (24*1024*1024) # define PJSIP_UDP_SO_RCVBUF_SIZE (24*1024*1024) # define PJ_DEBUG 0 # define PJSIP_SAFE_MODULE 0 # define PJ_HAS_STRICMP_ALNUM 0 # define PJ_HASH_USE_OWN_TOLOWER 1 # define PJSIP_UNESCAPE_IN_PLACE 1 # if defined(PJ_WIN32) || defined(PJ_WIN64) # define PJSIP_MAX_NET_EVENTS 10 # endif # define PJSUA_MAX_CALLS 512 #endif| We have also set the |__FD_SETSIZE| We are quite lost as this moment, what are we doing wrong here. Help would be highly appreciated.