PJSIP介绍
PJSIP是一个用C语言编写的免费开源多媒体通信库，实现了基于标准的协议，如SIP，SDP，RTP，STUN，TURN和ICE。它将**信令协议（SIP）**与丰富的多媒体框架和NAT遍历功能结合到高级API中，该API可移植，适用于从桌面，嵌入式系统到移动手持设备的几乎任何类型的系统。PJSIP既紧凑又功能丰富。它支持音频，视频，状态和即时消息，并具有丰富的文档。PJSIP 非常便携。在移动设备上，它抽象出系统相关的功能，并且在许多情况下能够利用设备的本机多媒体功能。PJSIP由一个自2005年以来专门为该项目工作的小团队开发，有来自世界各地的数百名开发人员参与，并自2007年起在SIP互操作性事件（SIPit）中定期进行测试。
PJSIP的下载
直接去开源的官方网站下载[https://www.pjsip.org/]
VS2015下载安装编译库
解压PJSIP -> pjproject-2.8
- 解压后的文件，具体文件内容我也是初学者没有具体的分析，这边不做介绍。
- VS2015添加库，VS添加库的方法比较简单
- 以添加pjsip库为例：
VS创建新的工程，添加项目，寻找项目的路径。\libpjsip\pjlib\build
寻找VS工程添加即可。 - 编译
编译的时候回出问题，显示找不到头文件config_site.h
只要自己按需求在目录\libpjsip\pjlib\include\pj添加一个config_site.h头文件即可。 - 头文件内容参考
#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 4
/*
* 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 4
#define PJSUA_MAX_PLAYERS 4
#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 4
/* Other pjsua settings */
#define PJSUA_MAX_ACC 4
#define PJSUA_MAX_PLAYERS 4
#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 4
/* Other pjsua settings */
#define PJSUA_MAX_ACC 4
#define PJSUA_MAX_PLAYERS 4
#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
/*
* 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 4
/* Separate worker thread for timer and ioqueue */
// #define PJSUA_SEPARATE_WORKER_FOR_TIMER 1
/* Other pjsua settings */
#define PJSUA_MAX_ACC 4
#define PJSUA_MAX_PLAYERS 4
#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
#undef PJMEDIA_AUDIO_DEV_HAS_ALSA
#define PJMEDIA_AUDIO_DEV_HAS_ALSA 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 32
# 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 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
- 成功截图
- 测试程序
#include <mysip.h>
/* Init random seed */
static void init_random_seed(void)
{
pj_sockaddr addr;
const pj_str_t* hostname;
pj_uint32_t pid;
pj_time_val t;
unsigned seed = 0;
/* Add hostname */
hostname = pj_gethostname();
seed = pj_hash_calc(seed, hostname->ptr, (int)hostname->slen);
/* Add primary IP address */
if (pj_gethostip(pj_AF_INET(), &addr) == PJ_SUCCESS) {
seed = pj_hash_calc(seed, &addr.ipv4.sin_addr, 4);
}
/* Get timeofday */
pj_gettimeofday(&t);
seed = pj_hash_calc(seed, &t, sizeof(t));
/* Add PID */
pid = pj_getpid();
seed = pj_hash_calc(seed, &pid, sizeof(pid));
/* Init random seed */
pj_srand(seed);
}
int main()
{
pj_status_t status;
pj_str_t str[1] = { "TEST", 4 };
pj_str_t str1[1];
str1->ptr = NULL;
str1->slen = 0;
// Error handling omited for clarity
// Must initialize PJLIB first!
status = pj_init();
/* Init random seed */
init_random_seed();
printf("%.*s\n", str->slen, str->ptr);
getchar();
return 0;
}
- 成功的截图
-
