PJSIP no audio on iPad - ios

Question

I have been working on a SIP client for iOS using PJSIP. Everything works great on iPhone all the models. But on iPad and iPod Touch the audio is not heard from the speakers of the iPad or the iPod Touch. The code works great for iPhone speakers. Since iPad and iPod do not have earpiece no audio is coming out from the SIP call from the speakers. Anyone who can let me know what could be wrong with the iPad implementation ? Here is the code that might help:
// Start and register sip account
int startPjsip(char sipUser, char sipDomain, char* scheme, char* realm, char* username, int dataType, char* passwd, char* proxy, char* protocol)
{
pj_status_t status;
pj_log_set_level(5);
// 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_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_reg_state = &on_reg_state;
// Init the logging config structure
pjsua_logging_config log_cfg;
pjsua_logging_config_default(&log_cfg);
log_cfg.console_level = 4;
// Init the pjsua
status = pjsua_init(&cfg, &log_cfg, NULL);
if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
// If outbound proxy is required.
if (proxy) {
cfg.outbound_proxy_cnt = 1;
cfg.outbound_proxy[0] = pj_str(proxy);
}
}
if (strcasecmp(protocol, "UDP") == 0)
{
/* iOS version above iOS 4 do not support UDP in bakground. */
// Add UDP transport.
{
// Init transport config structure
pjsua_transport_config cfg;
pjsua_transport_config_default(&cfg);
cfg.port = 5060;
// Add UDP transport.
status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &cfg, NULL);
if (status != PJ_SUCCESS) error_exit("Error creating transport", status);
}
}
if (strcasecmp(protocol, "TCP") == 0)
{
// 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
{
pjsua_acc_config cfg;
pjsua_acc_config_default(&cfg);
char sipId[MAX_SIP_ID_LENGTH];
sprintf(sipId, "sip:%s#%s", sipUser, sipDomain);
cfg.id = pj_str(sipId);
char regUri[MAX_SIP_REG_URI_LENGTH];
sprintf(regUri, "sip:%s", sipDomain);
cfg.reg_uri = pj_str(regUri);
// Add sip account credentials.
cfg.cred_count = 1;
cfg.cred_info[0].scheme = pj_str(scheme);
cfg.cred_info[0].realm = pj_str(realm);
cfg.cred_info[0].username = pj_str(username);
cfg.cred_info[0].data_type = dataType;
cfg.cred_info[0].data = pj_str(passwd);
status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
if (status != PJ_SUCCESS) error_exit("Error adding account", status);
}
return 0;
}
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);
pjsua_conf_adjust_rx_level(0, 1);
pjsua_conf_adjust_tx_level(0, 1);
}
PJ_LOG(3,(THIS_FILE, "Call %d state=%.*s", call_id,
(int)ci.state_text.slen,
ci.state_text.ptr));
}