Split reporting capability to separate file in siprtp sample

git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@437 74dad513-b988-da41-8d7b-12977e46ad98

1 files changed, 230 insertions, 0 deletions

diff --git a/pjsip-apps/src/samples/siprtp_report.c b/pjsip-apps/src/samples/siprtp_report.c
new file mode 100644
index 00000000..3b8b1ebe
--- /dev/null
+++ b/pjsip-apps/src/samples/siprtp_report.c
@@ -0,0 +1,230 @@
+/* $Id$ */
+/* 
+ * 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 
+ */
+
+
+/*
+ * DO NOT COMPILE THIS FILE ON ITS OWN!
+ *
+ * This file is included by siprtp.c to implement the reporting capability
+ * to a separate file, so that user can implement different reporting
+ * functionality (such as writing to XML file).
+ */
+
+static const char *good_number(char *buf, pj_int32_t val)
+{
+    if (val < 1000) {
+	pj_ansi_sprintf(buf, "%d", val);
+    } else if (val < 1000000) {
+	pj_ansi_sprintf(buf, "%d.%dK", 
+			val / 1000,
+			(val % 1000) / 100);
+    } else {
+	pj_ansi_sprintf(buf, "%d.%02dM", 
+			val / 1000000,
+			(val % 1000000) / 10000);
+    }
+
+    return buf;
+}
+
+
+static void print_call(int call_index)
+{
+    struct call *call = &app.call[call_index];
+    int len;
+    pjsip_inv_session *inv = call->inv;
+    pjsip_dialog *dlg = inv->dlg;
+    struct media_stream *audio = &call->media[0];
+    char userinfo[128];
+    char duration[80], last_update[80];
+    char bps[16], ipbps[16], packets[16], bytes[16], ipbytes[16];
+    pj_time_val now;
+
+    pj_gettimeofday(&now);
+
+    /* Print duration */
+    if (inv->state >= PJSIP_INV_STATE_CONFIRMED) {
+
+	PJ_TIME_VAL_SUB(now, call->connect_time);
+
+	sprintf(duration, " [duration: %02ld:%02ld:%02ld.%03ld]",
+		now.sec / 3600,
+		(now.sec % 3600) / 60,
+		(now.sec % 60),
+		now.msec);
+
+    } else {
+	duration[0] = '\0';
+    }
+
+
+
+    /* Call number and state */
+    printf("Call #%d: %s%s\n", call_index, pjsip_inv_state_name(inv->state), 
+			       duration);
+
+
+
+    /* Call identification */
+    len = pjsip_hdr_print_on(dlg->remote.info, userinfo, sizeof(userinfo));
+    if (len < 1)
+	pj_ansi_strcpy(userinfo, "<--uri too long-->");
+    else
+	userinfo[len] = '\0';
+
+    printf("   %s\n", userinfo);
+
+
+    if (call->inv == NULL || call->inv->state < PJSIP_INV_STATE_CONFIRMED ||
+	call->connect_time.sec == 0) 
+    {
+	return;
+    }
+
+
+    /* Signaling quality */
+    {
+	char pdd[64], connectdelay[64];
+	pj_time_val t;
+
+	if (call->response_time.sec) {
+	    t = call->response_time;
+	    PJ_TIME_VAL_SUB(t, call->start_time);
+	    sprintf(pdd, "got 1st response in %ld ms", PJ_TIME_VAL_MSEC(t));
+	} else {
+	    pdd[0] = '\0';
+	}
+
+	if (call->connect_time.sec) {
+	    t = call->connect_time;
+	    PJ_TIME_VAL_SUB(t, call->start_time);
+	    sprintf(connectdelay, ", connected after: %ld ms", 
+		    PJ_TIME_VAL_MSEC(t));
+	} else {
+	    connectdelay[0] = '\0';
+	}
+
+	printf("   Signaling quality: %s%s\n", pdd, connectdelay);
+    }
+
+
+    printf("   Stream #0: audio %.*s@%dHz, %dms/frame, %sB/s (%sB/s +IP hdr)\n",
+   	(int)audio->si.fmt.encoding_name.slen,
+	audio->si.fmt.encoding_name.ptr,
+	audio->clock_rate,
+	audio->samples_per_frame * 1000 / audio->clock_rate,
+	good_number(bps, audio->bytes_per_frame * audio->clock_rate / audio->samples_per_frame),
+	good_number(ipbps, (audio->bytes_per_frame+32) * audio->clock_rate / audio->samples_per_frame));
+
+    if (audio->rtcp.stat.rx.update_cnt == 0)
+	strcpy(last_update, "never");
+    else {
+	pj_gettimeofday(&now);
+	PJ_TIME_VAL_SUB(now, audio->rtcp.stat.rx.update);
+	sprintf(last_update, "%02ldh:%02ldm:%02ld.%03lds ago",
+		now.sec / 3600,
+		(now.sec % 3600) / 60,
+		now.sec % 60,
+		now.msec);
+    }
+
+    printf("              RX stat last update: %s\n"
+	   "                 total %s packets %sB received (%sB +IP hdr)%s\n"
+	   "                 pkt loss=%d (%3.1f%%), dup=%d (%3.1f%%), reorder=%d (%3.1f%%)%s\n"
+	   "                       (msec)    min     avg     max     last\n"
+	   "                 loss period: %7.3f %7.3f %7.3f %7.3f%s\n"
+	   "                 jitter     : %7.3f %7.3f %7.3f %7.3f%s\n",
+	   last_update,
+	   good_number(packets, audio->rtcp.stat.rx.pkt),
+	   good_number(bytes, audio->rtcp.stat.rx.bytes),
+	   good_number(ipbytes, audio->rtcp.stat.rx.bytes + audio->rtcp.stat.rx.pkt * 32),
+	   "",
+	   audio->rtcp.stat.rx.loss,
+	   audio->rtcp.stat.rx.loss * 100.0 / audio->rtcp.stat.rx.pkt,
+	   audio->rtcp.stat.rx.dup, 
+	   audio->rtcp.stat.rx.dup * 100.0 / audio->rtcp.stat.rx.pkt,
+	   audio->rtcp.stat.rx.reorder, 
+	   audio->rtcp.stat.rx.reorder * 100.0 / audio->rtcp.stat.rx.pkt,
+	   "",
+	   audio->rtcp.stat.rx.loss_period.min / 1000.0, 
+	   audio->rtcp.stat.rx.loss_period.avg / 1000.0, 
+	   audio->rtcp.stat.rx.loss_period.max / 1000.0,
+	   audio->rtcp.stat.rx.loss_period.last / 1000.0,
+	   "",
+	   audio->rtcp.stat.rx.jitter.min / 1000.0,
+	   audio->rtcp.stat.rx.jitter.avg / 1000.0,
+	   audio->rtcp.stat.rx.jitter.max / 1000.0,
+	   audio->rtcp.stat.rx.jitter.last / 1000.0,
+	   ""
+	   );
+
+
+    if (audio->rtcp.stat.tx.update_cnt == 0)
+	strcpy(last_update, "never");
+    else {
+	pj_gettimeofday(&now);
+	PJ_TIME_VAL_SUB(now, audio->rtcp.stat.tx.update);
+	sprintf(last_update, "%02ldh:%02ldm:%02ld.%03lds ago",
+		now.sec / 3600,
+		(now.sec % 3600) / 60,
+		now.sec % 60,
+		now.msec);
+    }
+
+    printf("              TX stat last update: %s\n"
+	   "                 total %s packets %sB sent (%sB +IP hdr)%s\n"
+	   "                 pkt loss=%d (%3.1f%%), dup=%d (%3.1f%%), reorder=%d (%3.1f%%)%s\n"
+	   "                       (msec)    min     avg     max     last\n"
+	   "                 loss period: %7.3f %7.3f %7.3f %7.3f%s\n"
+	   "                 jitter     : %7.3f %7.3f %7.3f %7.3f%s\n",
+	   last_update,
+	   good_number(packets, audio->rtcp.stat.tx.pkt),
+	   good_number(bytes, audio->rtcp.stat.tx.bytes),
+	   good_number(ipbytes, audio->rtcp.stat.tx.bytes + audio->rtcp.stat.tx.pkt * 32),
+	   "",
+	   audio->rtcp.stat.tx.loss,
+	   audio->rtcp.stat.tx.loss * 100.0 / audio->rtcp.stat.tx.pkt,
+	   audio->rtcp.stat.tx.dup, 
+	   audio->rtcp.stat.tx.dup * 100.0 / audio->rtcp.stat.tx.pkt,
+	   audio->rtcp.stat.tx.reorder, 
+	   audio->rtcp.stat.tx.reorder * 100.0 / audio->rtcp.stat.tx.pkt,
+	   "",
+	   audio->rtcp.stat.tx.loss_period.min / 1000.0, 
+	   audio->rtcp.stat.tx.loss_period.avg / 1000.0, 
+	   audio->rtcp.stat.tx.loss_period.max / 1000.0,
+	   audio->rtcp.stat.tx.loss_period.last / 1000.0,
+	   "",
+	   audio->rtcp.stat.tx.jitter.min / 1000.0,
+	   audio->rtcp.stat.tx.jitter.avg / 1000.0,
+	   audio->rtcp.stat.tx.jitter.max / 1000.0,
+	   audio->rtcp.stat.tx.jitter.last / 1000.0,
+	   ""
+	   );
+
+
+    printf("             RTT delay      : %7.3f %7.3f %7.3f %7.3f%s\n", 
+	   audio->rtcp.stat.rtt.min / 1000.0,
+	   audio->rtcp.stat.rtt.avg / 1000.0,
+	   audio->rtcp.stat.rtt.max / 1000.0,
+	   audio->rtcp.stat.rtt.last / 1000.0,
+	   ""
+	   );
+
+}
+