c - PJSip 拨出电话不起作用

Question

我正在开发一个iOS应用程序，其功能是使用PJSUA调用电话。同时，我的同事正在使用 PJSUA 2 为 Android 制作相同的应用程序。当我们从 Android 到 iOS 或从 Android 到 Android 进行调用时，一切正常。但是当我们尝试从 iOS 调用 Android 或从 iOS 调用 iOS 时，我们会收到错误:

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

这是我的类(class)，它正在使用 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

有人可以帮忙吗？我实在是没主意了。

Answer 1

似乎您在调用电话时还需要发送 callID，

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, &call_id);