PJSIP在linux上回应取消

时间:2016-04-08 16:22:10

标签: linux voip pjsip

我正在开发一个小型VOIP客户端应用程序,在C中运行在Linux中。我面临的问题是我需要声学回声消除,而我无法使其正常工作。

基于pjsip示例之一的源代码是

/* $Id: simple_pjsua.c 3553 2011-05-05 06:14:19Z nanang $ */
/*
 * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com)
 * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org>
 *
 * This program is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * This program is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with this program; if not, write to the Free Software
 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
 */

#include <pjsua-lib/pjsua.h>
#define THIS_FILE "APP"

#define SIP_DOMAIN "mydomain"
#define SIP_USER "myuser"
#define SIP_PASSWD "mypass"
#define SIP_REALM "myrealm"
#define SIP_SCHEME "myscheme"


static void call_on_dtmf_callback(pjsua_call_id call_id, int dtmf);

struct pjsua_player_eof_data {
    pj_pool_t *pool;
    pjsua_player_id player_id;
};
struct pjsua_player_eof_data *eof_data = NULL;
struct timeval t_rly;
pjmedia_echo_state *ec;
pjmedia_frame play_frame, rec_frame;

/* Callback called by the library upon receiving incoming call */
static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id,
        pjsip_rx_data *rdata) {
    pjsua_call_info ci;

    PJ_UNUSED_ARG(acc_id);
    PJ_UNUSED_ARG(rdata);

    pjsua_call_get_info(call_id, &ci);

    PJ_LOG(2, (THIS_FILE, "Incoming call from %.*s!!",
            (int) ci.remote_info.slen,
            ci.remote_info.ptr));

    /* Automatically answer incoming calls with 200/OK */
    pjsua_call_answer(call_id, 200, NULL, NULL);
}

/* Callback called by the library when call's state has changed */
static void on_call_state(pjsua_call_id call_id, pjsip_event *e) {
    pjsua_call_info ci;

    PJ_UNUSED_ARG(e);

    pjsua_call_get_info(call_id, &ci);
    PJ_LOG(2, (THIS_FILE, "Call %d state=%.*s", call_id,
            (int) ci.state_text.slen,
            ci.state_text.ptr));

}

/* Callback called by the library when call's media state has changed */
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);
    }
}

/* Display error and exit application */
static void error_exit(const char *title, pj_status_t status) {
    pjsua_perror(THIS_FILE, title, status);
    pjsua_destroy();
    exit(1);
}

/*
 * main()
 *
 * argv[1] may contain URL to call.
 */
int main(int argc, char *argv[]) {
    pjsua_acc_id acc_id;
    pj_status_t status;



    /* 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;

        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.max_calls = 1;

        pjsua_logging_config_default(&log_cfg);
        log_cfg.console_level = 6;

        pjsua_media_config_default(&media_cfg);
        media_cfg.ec_options = PJMEDIA_ECHO_DEFAULT;
        media_cfg.ec_tail_len = 250;


        status = pjsua_init(&cfg, &log_cfg, &media_cfg);
        // status = pjsua_init(&cfg, &log_cfg, NULL);

        if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);

        pj_pool_t *pool = pjsua_pool_create("my_echo", 1000, 1000);
        status = pjmedia_echo_create(pool, 16000, 320, 500, 500, PJMEDIA_ECHO_DEFAULT, &ec);
        play_frame.buf = pj_pool_alloc(pool, 320);
        rec_frame.buf = pj_pool_alloc(pool, 320);

    }

    /* Add UDP transport. */
    {
        pjsua_transport_config cfg;

        pjsua_transport_config_default(&cfg);
        cfg.port = 5060;
        status = pjsua_transport_create(PJSIP_TRANSPORT_UDP, &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 to SIP server by creating SIP account. */
    {
        pjsua_acc_config cfg;

        pjsua_acc_config_default(&cfg);
        cfg.id = pj_str("sip:" SIP_USER "@" SIP_DOMAIN);
        cfg.reg_uri = pj_str("sip:" SIP_DOMAIN);
        cfg.cred_count = 1;
        cfg.cred_info[0].realm = pj_str(SIP_REALM);
        cfg.cred_info[0].scheme = pj_str(SIP_SCHEME);
        cfg.cred_info[0].username = pj_str(SIP_USER);
        cfg.cred_info[0].data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
        cfg.cred_info[0].data = pj_str(SIP_PASSWD);

        status = pjsua_acc_add(&cfg, PJ_TRUE, &acc_id);
        if (status != PJ_SUCCESS) error_exit("Error adding account", status);
    }

    if (argc > 1) {
        pj_str_t uri = pj_str(argv[1]);
        status = pjsua_call_make_call(acc_id, &uri, 0, NULL, NULL, NULL);
        if (status != PJ_SUCCESS) error_exit("Error making call", status);
    }


    char option[10];
    while (1) {

        if (fgets(option, sizeof (option), stdin) == NULL) {
            puts("EOF while reading stdin, will quit now..");
            break;
        }

        if (option[0] == 'q')
            break;

    }

    /* Destroy pjsua */
    pjsua_destroy();

    return 0;
}

回显取消器已创建,但未运行,如日志所示:

16:10:45.203          speex !warning: No playback frame available (your application is buggy and/or got xruns)
16:10:45.207          speex !warning: Auto-filling the buffer (your application is buggy and/or got xruns)
16:10:45.216          speex !warning: internal playback buffer corruption?
16:10:45.221          speex !warning: Auto-filling the buffer (your application is buggy and/or got xruns)
16:10:45.279          speex  warning: Had to discard a playback frame (your application is buggy and/or got xruns)
16:10:45.393          speex  warning: Auto-filling the buffer (your application is buggy and/or got xruns)
16:10:45.462          speex  warning: Had to discard a playback frame (your application is buggy and/or got xruns)
16:10:45.522          speex  warning: Auto-filling the buffer (your application is buggy and/or got xruns)
16:10:45.562          speex  warning: Had to discard a playback frame (your application is buggy and/or got xruns)

Pjsip版本:2.4

任何人都能说出遗失的内容吗?

感谢。

0 个答案:

没有答案