Added simple_pjsua example
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@876 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/pjsip-apps/src/samples/simple_pjsua.c b/pjsip-apps/src/samples/simple_pjsua.c
new file mode 100644
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+++ b/pjsip-apps/src/samples/simple_pjsua.c
@@ -0,0 +1,197 @@
+/* $Id$ */
+/*
+ * Copyright (C) 2003-2006 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
+ */
+
+/**
+ * simple_pjsua.c
+ *
+ * This is a very simple but fully featured SIP user agent, with the
+ * following capabilities:
+ * - SIP registration
+ * - Making and receiving call
+ * - Audio/media to sound device.
+ *
+ * Usage:
+ * - To make outgoing call, start simple_pjsua with the URL of remote
+ * destination to contact.
+ * E.g.:
+ * simpleua sip:user@remote
+ *
+ * - Incoming calls will automatically be answered with 200.
+ *
+ * This program will quit once it has completed a single call.
+ */
+
+#include <pjsua-lib/pjsua.h>
+
+#define THIS_FILE "APP"
+
+#define SIP_DOMAIN "example.com"
+#define SIP_USER "alice"
+#define SIP_PASSWD "secret"
+
+
+/* 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(3,(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(3,(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);
+
+ /* If argument is specified, it's got to be a valid SIP URL */
+ if (argc > 1) {
+ status = pjsua_verify_sip_url(argv[1]);
+ if (status != PJ_SUCCESS) error_exit("Invalid URL in argv", status);
+ }
+
+ /* Init pjsua */
+ {
+ pjsua_config cfg;
+ pjsua_logging_config log_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;
+
+ pjsua_logging_config_default(&log_cfg);
+ log_cfg.console_level = 4;
+
+ status = pjsua_init(&cfg, &log_cfg, NULL);
+ if (status != PJ_SUCCESS) error_exit("Error in pjsua_init()", status);
+ }
+
+ /* 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_DOMAIN);
+ cfg.cred_info[0].scheme = pj_str("digest");
+ 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 URL is specified, make call to the URL. */
+ 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);
+ }
+
+ /* Wait until user press "q" to quit. */
+ for (;;) {
+ char option[10];
+
+ puts("Press 'h' to hangup all calls, 'q' to quit");
+ fgets(option, sizeof(option), stdin);
+
+ if (option[0] == 'q')
+ break;
+
+ if (option[0] == 'h')
+ pjsua_call_hangup_all();
+ }
+
+ /* Destroy pjsua */
+ pjsua_destroy();
+
+ return 0;
+}