在集成PJSIP后,要从我的应用程序拨打电话,我正在使用此代码:
sip_dial(voipManager._sip_acc_id, [dialedUser UTF8String], &id_call);
但它在跟踪时返回420006状态代码,显示:
无法找到默认音频设备(PJMEDIA_EAUD_NODEFDEV)[status = 420006]
我已经从经理文件中启用了编解码器,并且在编译时也显示“已启用”。我错过或误导的地方在哪里?
答案 0 :(得分:4)
- (int)startPjsipAndRegisterOnServer:(char *)sipDomain withUserName:(char *)sipUser andPassword:(char *)password callback:(RegisterCallBack)callback
{
//Disconnect connection(Registration).
// pjsua_destroy();
pj_status_t status;
// Create pjsua first
status = pjsua_create();
if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);
// Init pjsua
{
// Init the config structure
pjsua_config cfg;
pjsua_logging_config log_cfg;
pjsua_config_default (&cfg);
//Media
pjsua_media_config media_cfg;//Extra
pjsua_media_config_default(&media_cfg);//Extra
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;
cfg.cb.on_reg_state2 = &on_reg_state2;//Extra
// Init the logging config structure
pjsua_logging_config_default(&log_cfg);
log_cfg.console_level = 4;
// Init the pjsua
status = pjsua_init(&cfg, &log_cfg, &media_cfg);
if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
}
// Add UDP transport.
{
// Init transport config structure
pjsua_transport_config cfg;
pjsua_transport_config_default(&cfg);
cfg.port = 5060;
// Add TCP transport.
status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
if (status != PJ_SUCCESS) error_exit("Error creating transport", status);
}
/*
// Add TCP transport.
{
// Init transport config structure
pjsua_transport_config cfg;
pjsua_transport_config_default(&cfg);
cfg.port = 5060;
// Add TCP transport.
status = pjsua_transport_create(PJSIP_TRANSPORT_TCP, &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 the account on local sip server/* Register to SIP server by creating SIP account. */
{
pjsua_acc_config cfg;
pjsua_acc_config_default(&cfg);
// Account ID
char sipId[MAX_SIP_ID_LENGTH];
sprintf(sipId, "sip:%s@%s", sipUser, sipDomain);
cfg.id = pj_str(sipId);
// Reg URI
char regUri[MAX_SIP_REG_URI_LENGTH];
sprintf(regUri, "sip:%s", sipDomain);
cfg.reg_uri = pj_str(regUri);
// Account cred info
cfg.cred_count = 1;
cfg.cred_info[0].scheme = pj_str("digest");
cfg.cred_info[0].realm = pj_str(sipDomain);
cfg.cred_info[0].username = pj_str(sipUser);
cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
cfg.cred_info[0].data = pj_str(password);
status = pjsua_acc_add(&cfg, PJ_TRUE, &_acc_id);
if (status != PJ_SUCCESS) error_exit("Error adding account", status);
}
_registerCallBack = callback;
return 0;
}
- (void)makeCallTo:(char*)destUri
{
pj_status_t status;
pj_str_t uri = pj_str(destUri);
//current register id _acc_id
status = pjsua_call_make_call(_acc_id, &uri, 0, NULL, NULL, NULL);
if (status != PJ_SUCCESS) error_exit("Error making call", status);
}