// helper for setting up sip library pjsua
static void setup_sip(void)
{
pj_status_t status;
log_message("Setting up pjsua ... ");
// create pjsua
status = pjsua_create();
if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);
// configure pjsua
pjsua_config cfg;
pjsua_config_default(&cfg);
// enable just 1 simultaneous call
cfg.max_calls = 1;
// callback configuration
cfg.cb.on_call_media_state = &on_call_media_state;
cfg.cb.on_call_state = &on_call_state;
// logging configuration
pjsua_logging_config log_cfg;
pjsua_logging_config_default(&log_cfg);
log_cfg.console_level = PJSUA_LOG_LEVEL;
// initialize pjsua
status = pjsua_init(&cfg, &log_cfg, NULL);
if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
// add udp transport
pjsua_transport_config udpcfg;
pjsua_transport_config_default(&udpcfg);
udpcfg.port = 5060;
status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &udpcfg, NULL);
if (status != PJ_SUCCESS) error_exit("Error creating transport", status);
// initialization is done, start pjsua
status = pjsua_start();
if (status != PJ_SUCCESS) error_exit("Error starting pjsua", status);
// disable sound - use null sound device
status = pjsua_set_null_snd_dev();
if (status != PJ_SUCCESS) error_exit("Error disabling audio", status);
log_message("Done.\n");
}
bool SipPhone::init_pjsua() {
Logger::debug("SipPhone::init_pjsua...");
// create pjsua
pj_status_t status = pjsua_create();
if (status != PJ_SUCCESS) {
Logger::error("pjsua_create() failed (%s)", Helper::getPjStatusAsString(status).c_str());
return false;
}
// configure pjsua
pjsua_config ua_cfg;
pjsua_config_default(&ua_cfg);
// enable just 1 simultaneous call
ua_cfg.max_calls = 1; // TODO
// callback configuration
ua_cfg.cb.on_call_state = &SipAccount::onCallStateCB;
ua_cfg.cb.on_incoming_call = &SipAccount::onIncomingCallCB;
//ua_cfg.cb.on_call_media_state = &SipAccount::onCallMediaStateCB;
// logging configuration
pjsua_logging_config log_cfg;
pjsua_logging_config_default(&log_cfg);
log_cfg.level = pj_log_get_level();
// media configuration
pjsua_media_config media_cfg;
pjsua_media_config_default(&media_cfg);
media_cfg.clock_rate = 8000; // TODO: default of 16000 seems not to work :-(
// initialize pjsua
status = pjsua_init(&ua_cfg, &log_cfg, &media_cfg);
if (status != PJ_SUCCESS) {
Logger::error("pjsua_init() failed (%s)", Helper::getPjStatusAsString(status).c_str());
return false;
}
// add udp transport
pjsua_transport_config udpcfg;
pjsua_transport_config_default(&udpcfg);
status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &udpcfg, NULL);
if (status != PJ_SUCCESS) {
Logger::error("pjsua_transport_create() failed (%s)", Helper::getPjStatusAsString(status).c_str());
return false;
}
// disable sound device - use null sound device
status = pjsua_set_null_snd_dev();
if (status != PJ_SUCCESS) {
Logger::error("pjsua_set_null_snd_dev() failed (%s)", Helper::getPjStatusAsString(status).c_str());
return false;
}
// initialization is done, start pjsua
status = pjsua_start();
if (status != PJ_SUCCESS) {
Logger::error("pjsua_start() failed (%s)", Helper::getPjStatusAsString(status).c_str());
return false;
}
#if 0
m_Pool = pjsua_pool_create("SipPhone.cpp", 128, 128);
if (m_Pool == NULL) {
Logger::error("pjsua_pool_create() failed");
return false;
}
#endif
return true;
}
/*
* main()
*
* argv[] contains the registration information.
*/
int main(int argc, char *argv[]) {
pj_status_t status;
int loglevel = 0;
if (argc < 4) {
usage(argv[0]);
exit(1);
}
if (argc > 5){
sscanf(argv[5], "%d", &loglevel);
}
pjsua_acc_id acc_id;
/* Create pjsua first! */
status = pjsua_create();
if (status != PJ_SUCCESS) error_exit("Error in pjsua_create()", status);
/* Init pjsua */
{
pjsua_config cfg;
pjsua_logging_config log_cfg;
pjsua_media_config media_cfg;
// This will prevent the SIP stack from trying to switch to TCP.
// It will prevent the stack from giving "Transport unavailable" errors.
// http://trac.pjsip.org/repos/wiki/Using_SIP_TCP
pjsip_cfg()->endpt.disable_tcp_switch = PJ_TRUE;
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;
cfg.cb.on_call_tsx_state = &on_call_tsx_state;
pjsua_logging_config_default(&log_cfg);
log_cfg.console_level = loglevel; // 0 = Mute console, 3 = somewhat useful, 4 = overly verbose.
pjsua_media_config_default(&media_cfg);
status = pjsua_init(&cfg, &log_cfg, &media_cfg);
if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
}
/* Add UDP transport. */
{
pjsua_transport_config cfg;
pjsua_transport_config_default(&cfg);
sscanf(argv[4], "%d", &cfg.port);
status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
if (status != PJ_SUCCESS) error_exit("Error creating transport", status);
printf("Created UDP transport on port %d.\n", cfg.port);
}
/* 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. */
{
char reg_uri_buf[80] = "sip:";
char uri_buf[80] = "sip:";
pjsua_acc_config cfg;
char* username = argv[1];
char* password = argv[2];
char* domain = argv[3];
strcat (uri_buf, username);
strcat (reg_uri_buf, domain);
strcat (uri_buf, "@");
strcat (uri_buf, domain);
pjsua_acc_config_default(&cfg);
cfg.id = pj_str(uri_buf);
printf("Registering: %.*s.\n", (int)cfg.id.slen, cfg.id.ptr);
cfg.reg_uri = pj_str(reg_uri_buf);
cfg.cred_count = 1;
cfg.cred_info[0].realm = pj_str(domain);
cfg.cred_info[0].scheme = pj_str("Digest");
cfg.cred_info[0].username = pj_str(username);
cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
cfg.cred_info[0].data = pj_str(password);
cfg.register_on_acc_add = PJ_FALSE;
status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
if (status != PJ_SUCCESS) error_exit("Error adding account", status);
or_event(OR_READY, "Agent initialized..");
}
/* Start by muting the audio device. This a passive agent, and sound
is just wasted CPU time and uwanted side effects. */
pjsua_set_null_snd_dev();
fflush(stdout);
/* Wait until user press "q" to quit. */
for (;;) {
char option[256];
if (fgets(option, sizeof (option), stdin) == NULL) {
or_status (OR_ERROR, "EOF while reading stdin, will quit now..");
break;
}
/* Dial command. */
if (option[0] == 'd') {
pj_str_t uri = pj_str(&option[1]);
status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);
if (status != PJ_SUCCESS) {
or_status (OR_ERROR, "Could not make call");
}
else {
or_status (OR_OK, "Dialling...");
}
}
/* Register */
else if (option[0] == 'r') {
register_account(acc_id);
}
/* Unregister */
else if (option[0] == 'u') {
unregister_account(acc_id);
}
/* Enable autoanswer */
else if (option[0] == 'a') {
if (option[1] == '1') {
autoanswer[0] = true;
or_status(OR_OK, "Autoanswer enabled.");
} else {
or_status(OR_ERROR, "Invalid account.");
}
}
/* Disable autoanswer (manual answer) */
else if (option[0] == 'm') {
autoanswer[0] = false;
or_status(OR_OK, "Autoanswer disabled.");
}
/* Pickup incoming call, unsupported for now. */
else if (option[0] == 'p') {
if (current_call != PJSUA_INVALID_ID) {
pjsua_call_answer(current_call, 200, NULL, NULL);
or_status(OR_OK, "Call picked up.");
} else {
or_status(OR_ERROR, "No call to pick up.");
}
}
/* Hang up current call */
else if (option[0] == 'H') {
if (current_call != PJSUA_INVALID_ID) {
pjsua_call_hangup (current_call, 0,NULL, NULL);
or_status (OR_OK, "Hanging up current call...");
} else {
or_status(OR_ERROR, "No call to hang up.");
}
}
/* Full hangup.. */
else if (option[0] == 'h') {
pjsua_call_hangup_all();
or_status (OR_OK, "Hanging up all calls...");
}
/* Status */
else if (option[0] == 's') {
or_dump_status();
}
/* Exit application. */
else if (option[0] == 'q') {
break;
}
else {
or_status (OR_ERROR, "Unknown command:");
}
}
pjsua_destroy();
or_status (OR_OK, "Exiting...");
return 0;
}
