Pjsip笔记之pjsip使用示例

学习一个程序，最希望的就是有个demo，通过demo的API调用逻辑，跟踪程序的执行过程，了解里面的设计。

Pjsip一个最简单的示例就是simple_pjsua.c，位于：pjsip_apps/src/samples目录下。不到200行的代码。却演示了pjsip初始化到拨打电话和挂点电话的API调用逻辑。

主要的逻辑在main函数中：

pjsua接口使用时，需要创建、初始化、开始和销毁的操作：pjsua_create、pjsua_init、pjsua_start、pjsua_destroy

pjsua_transport_create创建sip信令发送和接收需要的相关socket等资源

pjsua_acc_add添加拨打电话账号，账号类似于我们的手机号码，可以起到定位的功能。

拨打电话的挂断电话：pjsua_call_make_call，pjsua_call_hangup_all

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main函数代码如下：

/* * main() * * argv[1] may contain URL to call. */ int main(int argc, char *argv[]) { pjsua_acc_id acc_id; pj_status_t status; /* Create pjsua first! */ status = pjsua_create(); if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status); /* If argument is specified, it's got to be a valid SIP URL */ if (argc > 1) { status = pjsua_verify_url(argv[1]); if (status != PJ_SUCCESS) error_exit("Invalid URL in argv", status); } /* Init pjsua */ { pjsua_config cfg; pjsua_logging_config log_cfg; pjsua_config_default(&cfg); cfg.cb.on_incoming_call = &on_incoming_call; cfg.cb.on_call_media_state = &on_call_media_state; cfg.cb.on_call_state = &on_call_state; pjsua_logging_config_default(&log_cfg); log_cfg.console_level = 4; status = pjsua_init(&cfg, &log_cfg, NULL); if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status); } /* Add UDP transport. */ { pjsua_transport_config cfg; pjsua_transport_config_default(&cfg); cfg.port = 5060; status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL); if (status != PJ_SUCCESS) error_exit("Error creating transport", status); } /* Initialization is done, now start pjsua */ status = pjsua_start(); if (status != PJ_SUCCESS) error_exit("Error starting pjsua", status); /* Register to SIP server by creating SIP account. */ { pjsua_acc_config cfg; pjsua_acc_config_default(&cfg); cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN); cfg.reg_uri = pj_str("sip:" SIP_DOMAIN); cfg.cred_count = 1; cfg.cred_info[0].realm = pj_str(SIP_DOMAIN); cfg.cred_info[0].scheme = pj_str("digest"); cfg.cred_info[0].username = pj_str(SIP_USER); cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD; cfg.cred_info[0].data = pj_str(SIP_PASSWD); status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id); if (status != PJ_SUCCESS) error_exit("Error adding account", status); } /* If URL is specified, make call to the URL. */ if (argc > 1) { pj_str_t uri = pj_str(argv[1]); status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL); if (status != PJ_SUCCESS) error_exit("Error making call", status); } /* Wait until user press "q" to quit. */ for (;;) { char option[10]; puts("Press 'h' to hangup all calls, 'q' to quit"); if (fgets(option, sizeof(option), stdin) == NULL) { puts("EOF while reading stdin, will quit now.."); break; } if (option[0] == 'q') break; if (option[0] == 'h') pjsua_call_hangup_all(); } /* Destroy pjsua */ pjsua_destroy(); return 0; } 

有来电时的通知回调函数：

/* Callback called by the library upon receiving incoming call */ static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id, pjsip_rx_data *rdata) { pjsua_call_info ci; PJ_UNUSED_ARG(acc_id); PJ_UNUSED_ARG(rdata); pjsua_call_get_info(call_id, &ci); PJ_LOG(3,(THIS_FILE, "Incoming call from %.*s!!", (int)ci.remote_info.slen, ci.remote_info.ptr)); /* Automatically answer incoming calls with 200/OK */ pjsua_call_answer(call_id, 200, NULL, NULL); }

电话状态通知回调函数：

/* Callback called by the library when call's state has changed */ static void on_call_state(pjsua_call_id call_id, pjsip_event *e) { pjsua_call_info ci; PJ_UNUSED_ARG(e); pjsua_call_get_info(call_id, &ci); PJ_LOG(3,(THIS_FILE, "Call %d state=%.*s", call_id, (int)ci.state_text.slen, ci.state_text.ptr)); }

通话建立连接时，语音、视频等的相关状态通知回调函数：

/* Callback called by the library when call's media state has changed */ static void on_call_media_state(pjsua_call_id call_id) { pjsua_call_info ci; pjsua_call_get_info(call_id, &ci); if (ci.media_status == PJSUA_CALL_MEDIA_ACTIVE) { // When media is active, connect call to sound device. pjsua_conf_connect(ci.conf_slot, 0); pjsua_conf_connect(0, ci.conf_slot); } }

示例的代码很短，但是却完全的展示了pjsip的功能。

在学习Pjsip时，始终记住，Pjsip只是完成两个功能。

1、使用sip信令协商双方使用音频、视频通话使用的rtp rtcp的socket端口，视频编码器和音频编码器的类型和相关的编码参数，使用的网络类型。

2、完成音频，视频通话的socket通道，传输音频和视频数据。

整个工程就是为了上面的两个功能服务。为了保证不同网络之间的数据传输，pjsip还增加了网络穿越的的ICE，stun等协议。