如何使用 PJSIP 同时创建多个 SIP 呼叫

时间:2021-04-10 03:19:56

标签: c sip voip pjsip pjsua2

我正在开发一个应用程序,可以同时从我的 PC 到 32 个使用 PJSIP 的 PC 进行 32 个呼叫(不保持呼叫进行新呼叫)。我的电脑有 32 个声卡设备,我将每个呼叫都匹配到一个声卡,所有呼叫都通过端口 5060 使用 SIP。我成功地进行了一次这样的呼叫:

 #include <pjsua-lib/pjsua.h>
 #define THIS_FILE "APP"
 
 #define SIP_DOMAIN "example.com"
 #define SIP_USER "alice"
 #define SIP_PASSWD "secret"
 
 
 /* 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);
 }
 
 /* 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));
 }
 
 /* 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);
  }
 }
 
 /* Display error and exit application */
 static void error_exit(const char *title, pj_status_t status)
 {
  pjsua_perror(THIS_FILE, title, status);
  pjsua_destroy();
  exit(1);
 }
 
 /*
  * 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((char )"");
  cfg.cred_info[0].scheme = pj_str((char *)"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 (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();
  }
 
  pjsua_destroy();
 
  return 0;
 }

此代码现在可以通过创建一个 PJSUA 成功呼叫 SIP 服务器,在该服务器上注册一个帐户并在一个声音设备上创建呼叫流程。为了同时进行多个调用,我创建了 32 个 pjsua_config cfg,每个 pjsua_acc_id 一个,但它无法运行,所以我认为我不能在一个进程中进行多个 pjsua。那么我应该如何创建对 SIP 服务器的多个呼叫?

0 个答案:

没有答案
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