I am developing an iOS app which function is to make calls using PJSUA. Concurrently my colleague is making same app for Android using PJSUA 2. When we are making call from Android to iOS or from Android to Android everything works fine. But when we are trying to make a call from iOS to Android or from iOS to iOS we get an error:
18:51:46.959 pjsua_media.c .....Call 0: updating media..
18:51:46.959 pjsua_media.c .......Media stream call00:0 is destroyed
18:51:46.959 pjsua_media.c ......pjmedia_transport_media_start() failed for call_id 0 media 0: SRTP crypto-suite name not match the offerer tag (PJMEDIA_SRTP_ECRYPTONOTMATCH)
18:51:46.959 pjsua_media.c .......Media stream call00:0 is destroyed
18:51:46.960 pjsua_media.c ......Error updating media call00:0: SRTP crypto-suite name not match the offerer tag (PJMEDIA_SRTP_ECRYPTONOTMATCH)
18:51:46.960 pjsua_core.c .....TX 362 bytes Request msg ACK/cseq=19029 (tdta0x1200c1600) to TLS 95.213.169.231:1605:
ACK sip:s2@95.213.169.231:1605;transport=tls SIP/2.0
And here is my class, which is working with PJSUA:
#import <Foundation/Foundation.h>
#import "SIPConnection.h"
#import <pjlib.h>
#import <pjsua.h>
#import <pj/log.h>
#import <pjmedia-codec.h>
#import <pjmedia.h>
@interface SIPConnection ()
@end
@implementation SIPConnection
pjsua_acc_id acc_id = -1;
pjsua_call_id current_call_id = -1;
bool registration_status_is_ok = false;
bool is_making_call = false;
-(void) registerAccount {
registerAccount();
}
-(void) makeCall {
is_making_call = true;
registerAccount();
}
void registerAccount() {
pjsua_create();
pjsua_config ua_cfg;
pjsua_logging_config log_cfg;
pjsua_media_config media_cfg;
char* user_agent_string = "XChat";
pj_str_t user_agent = pj_str(user_agent_string);
ua_cfg.user_agent = user_agent;
pjsua_config_default(&ua_cfg);
pjsua_logging_config_default(&log_cfg);
pjsua_media_config_default(&media_cfg);
ua_cfg.cb.on_reg_state = &sip_reg_state_changed;
ua_cfg.cb.on_call_state = &sip_call_state_changed;
ua_cfg.cb.on_call_media_state = &sip_call_media_state_changed;
ua_cfg.cb.on_incoming_call = &sip_incoming_call;
media_cfg.ec_options = 3;
media_cfg.ec_tail_len = 1000;
pjsua_init(&ua_cfg, &log_cfg, &media_cfg);
pjsua_transport_config transport_cfg;
pjsua_transport_config_default(&transport_cfg);
transport_cfg.port = 1605;
pjsua_transport_id trasport_id;
pjsua_transport_create(PJSIP_TRANSPORT_TLS, &transport_cfg, &trasport_id);
pjsua_acc_config acc_cfg;
pjsua_acc_config_default(&acc_cfg);
acc_cfg.register_on_acc_add = YES;
acc_cfg.priority = 100;
acc_cfg.reg_retry_interval = 3;
acc_cfg.reg_retry_random_interval = 3;
acc_cfg.publish_enabled = NO;
acc_cfg.unreg_timeout = 1500;
acc_cfg.reg_first_retry_interval = 5;
acc_cfg.reg_delay_before_refresh = 0;
char* reg_uri_string = "sip:95.213.169.231:1605;transport=TLS";
char* acc_uri_string = "sip:s2@95.213.169.231:1605";
pj_str_t reg_uri = pj_str(reg_uri_string);
pj_str_t acc_uri = pj_str(acc_uri_string);
acc_cfg.reg_uri = reg_uri;
acc_cfg.cred_info[0].scheme=pj_str("digest");
acc_cfg.cred_info[0].data_type=0;
acc_cfg.cred_info[0].username=pj_str("s2");
acc_cfg.cred_info[0].realm=pj_str("*");
acc_cfg.cred_info[0].data=pj_str("LHillKkF4Aflc3rR");
acc_cfg.cred_count = 1;
acc_cfg.use_srtp = PJMEDIA_SRTP_MANDATORY;
acc_cfg.srtp_secure_signaling = 1;
acc_cfg.allow_sdp_nat_rewrite = 1;
acc_cfg.allow_contact_rewrite = 1;
acc_cfg.id = acc_uri;
pjsua_acc_add(&acc_cfg, YES, &acc_id);
pjsua_start();
}
static void sip_reg_state_changed(pjsua_acc_id acc_id) {
pjsua_acc_info acc_info;
pj_status_t status = pjsua_acc_get_info(acc_id, &acc_info);
if (status != PJ_SUCCESS) {
NSLog(@"Couldn't get account status");
registration_status_is_ok = false;
return;
}
if (acc_info.status == 200) {
NSLog(@"Account is registered!");
registration_status_is_ok = true;
make_call();
}
}
static void sip_call_state_changed(pjsua_call_id call_id, pjsip_event* event) {
}
static void sip_call_media_state_changed(pjsua_call_id call_id) {
pjsua_call_info call_info;
unsigned mi;
pj_bool_t has_error = PJ_FALSE;
pjsua_call_get_info(call_id, &call_info);
current_call_id = call_id;
mi=0;
//for (mi=0; mi<call_info.media_cnt; ++mi) {
on_call_generic_media_state(&call_info, mi, &has_error);
switch (call_info.media[mi].type) {
case PJMEDIA_TYPE_AUDIO:
on_call_audio_state(&call_info, mi, &has_error);
break;
}
//}
}
static void on_call_audio_state(pjsua_call_info *ci, unsigned mi, pj_bool_t *has_error)
{
PJ_UNUSED_ARG(has_error);
/* Stop ringback */
//ring_stop(ci->id);
/* Connect ports appropriately when media status is ACTIVE or REMOTE HOLD,
457 * otherwise we should NOT connect the ports.
458 */
if (ci->media[mi].status == PJSUA_CALL_MEDIA_ACTIVE || ci->media[mi].status == PJSUA_CALL_MEDIA_REMOTE_HOLD)
{
pj_bool_t connect_sound = PJ_TRUE;
pj_bool_t disconnect_mic = PJ_FALSE;
pjsua_conf_port_id call_conf_slot;
call_conf_slot = ci->media[mi].stream.aud.conf_slot;
if (connect_sound) {
pj_status_t res = pjsua_conf_connect(call_conf_slot, 0);
if (!disconnect_mic)
{
res = pjsua_conf_connect(0, call_conf_slot);
}
}
}
}
static void on_call_generic_media_state(pjsua_call_info *ci, unsigned mi, pj_bool_t *has_error)
{
const char *status_name[] = {
"None",
"Active",
"Local hold",
"Remote hold",
"Error"
};
PJ_UNUSED_ARG(has_error);
pj_assert(ci->media[mi].status <= PJ_ARRAY_SIZE(status_name));
pj_assert(PJSUA_CALL_MEDIA_ERROR == 4);
}
static void sip_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id, pjsip_rx_data *rdata)
{
pjsua_call_info call_info;
pjsua_call_get_info(call_id, &call_info);
pjsua_call_answer(call_id, 200, NULL, NULL);
}
static void make_call()
{
if ((registration_status_is_ok) && (is_making_call))
{
pjsua_call_setting call_stg;
pjsua_call_setting_default(&call_stg);
call_stg.aud_cnt = 1;
call_stg.vid_cnt = 0;
// pjsua_call_id call_id;
pj_str_t dst_uri = pj_str("sip:s3@95.213.169.231:1605;transport=TLS");
pjsua_call_make_call(acc_id, &dst_uri, &call_stg, NULL, NULL, NULL);
}
}
static void end_call()
{
/*if (current_call_id != -1)
{
pjsua_call_dump(current_call_id, YES, NULL, 8, NULL);
}*/
if (acc_id != -1)
{
pjsua_acc_del(acc_id);
}
pjsua_destroy();
}
- (void)endCall
{
end_call();
}
@end
Can somebody help please? I am really out of ideas.