Added more stats and options in siprtp samples
git-svn-id: https://svn.pjsip.org/repos/pjproject/trunk@379 74dad513-b988-da41-8d7b-12977e46ad98
diff --git a/pjsip-apps/build/Samples-vc.mak b/pjsip-apps/build/Samples-vc.mak
index 59f1fde..43ed926 100644
--- a/pjsip-apps/build/Samples-vc.mak
+++ b/pjsip-apps/build/Samples-vc.mak
@@ -40,7 +40,7 @@
SAMPLES = $(BINDIR)\simpleua.exe $(BINDIR)\playfile.exe $(BINDIR)\playsine.exe \
$(BINDIR)\confsample.exe $(BINDIR)\sndinfo.exe \
$(BINDIR)\level.exe $(BINDIR)\recfile.exe \
- $(BINDIR)\resampleplay.exe
+ $(BINDIR)\resampleplay.exe $(BINDIR)\siprtp.exe
all: $(OBJDIR) $(SAMPLES)
diff --git a/pjsip-apps/build/Samples.mak b/pjsip-apps/build/Samples.mak
index 66479ff..749c939 100644
--- a/pjsip-apps/build/Samples.mak
+++ b/pjsip-apps/build/Samples.mak
@@ -38,7 +38,8 @@
OBJDIR := ./output/samples-$(MACHINE_NAME)-$(OS_NAME)-$(CC_NAME)
BINDIR := ../bin/samples
-SAMPLES := simpleua playfile playsine confsample sndinfo level recfile resampleplay
+SAMPLES := simpleua playfile playsine confsample sndinfo level recfile resampleplay \
+ siprtp
EXES := $(foreach file, $(SAMPLES), $(BINDIR)/$(file)-$(MACHINE_NAME)-$(OS_NAME)-$(CC_NAME)$(HOST_EXE))
diff --git a/pjsip-apps/src/samples/siprtp.c b/pjsip-apps/src/samples/siprtp.c
index 72ab5c7..1c16b9c 100644
--- a/pjsip-apps/src/samples/siprtp.c
+++ b/pjsip-apps/src/samples/siprtp.c
@@ -34,6 +34,30 @@
#define RTP_START_PORT 44100
+/* Codec descriptor: */
+struct codec
+{
+ unsigned pt;
+ char* name;
+ unsigned clock_rate;
+ unsigned bit_rate;
+ unsigned ptime;
+ char* description;
+};
+
+
+/* Unidirectional media stat: */
+struct stream_stat
+{
+ pj_uint32_t pkt, payload;
+ pj_uint32_t discard, reorder;
+ unsigned loss_min, loss_avg, loss_max;
+ char *loss_type;
+ unsigned jitter_min, jitter_avg, jitter_max;
+ unsigned rtcp_cnt;
+};
+
+
/* A bidirectional media stream */
struct media_stream
{
@@ -60,6 +84,10 @@
pjmedia_rtcp_session rtcp; /* incoming RTCP session. */
pjmedia_rtcp_pkt rem_rtcp; /* received RTCP stat. */
+ /* More stats: */
+ struct stream_stat rx_stat; /* incoming stream stat */
+ struct stream_stat tx_stat; /* outgoing stream stat. */
+
/* Thread: */
pj_bool_t thread_quit_flag; /* worker thread quit flag */
pj_thread_t *thread; /* RTP/RTCP worker thread */
@@ -72,6 +100,9 @@
pjsip_inv_session *inv;
unsigned media_count;
struct media_stream media[2];
+ pj_time_val start_time;
+ pj_time_val response_time;
+ pj_time_val connect_time;
};
@@ -84,6 +115,12 @@
char *local_addr;
pj_str_t local_uri;
pj_str_t local_contact;
+
+ int app_log_level;
+ int log_level;
+ char *log_filename;
+
+ struct codec audio_codec;
pj_str_t uri_to_call;
@@ -134,6 +171,8 @@
pj_status_t status);
+
+
/* This is a PJSIP module to be registered by application to handle
* incoming requests outside any dialogs/transactions. The main purpose
* here is to handle incoming INVITE request message, where we will
@@ -157,12 +196,22 @@
};
+/* Codec constants */
+struct codec audio_codecs[] =
+{
+ { 0, "pcmu", 8000, 64000, 20, "G.711 ULaw" },
+ { 3, "gsm", 8000, 13200, 20, "GSM" },
+ { 4, "g723", 8000, 6400, 30, "G.723.1" },
+ { 8, "pcma", 8000, 64000, 20, "G.711 ALaw" },
+ { 18, "g729", 8000, 8000, 20, "G.729" },
+};
+
+
/*
* Init SIP stack
*/
static pj_status_t init_sip()
{
- unsigned i;
pj_status_t status;
/* init PJLIB-UTIL: */
@@ -241,12 +290,6 @@
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
- /* Start worker threads */
- for (i=0; i<app.thread_count; ++i) {
- pj_thread_create( app.pool, "app", &worker_thread, NULL,
- 0, 0, &app.thread[i]);
- }
-
/* Done */
return PJ_SUCCESS;
}
@@ -462,6 +505,9 @@
/* Attach call data to invite session */
call->inv->mod_data[mod_siprtp.id] = call;
+ /* Mark start of call */
+ pj_gettimeofday(&call->start_time);
+
/* Create initial INVITE request.
* This INVITE request will contain a perfectly good request and
@@ -528,17 +574,34 @@
/* Create UAS invite session */
status = pjsip_inv_create_uas( dlg, rdata, sdp, 0, &call->inv);
if (status != PJ_SUCCESS) {
- pjsip_dlg_terminate(dlg);
+ pjsip_dlg_create_response(dlg, rdata, 500, NULL, &tdata);
+ pjsip_dlg_send_response(dlg, pjsip_rdata_get_tsx(rdata), tdata);
return;
}
+
/* Attach call data to invite session */
call->inv->mod_data[mod_siprtp.id] = call;
+ /* Mark start of call */
+ pj_gettimeofday(&call->start_time);
+
+
+
/* Create 200 response .*/
status = pjsip_inv_initial_answer(call->inv, rdata, 200,
NULL, NULL, &tdata);
- PJ_ASSERT_ON_FAIL(status == PJ_SUCCESS, return);
+ if (status != PJ_SUCCESS) {
+ status = pjsip_inv_initial_answer(call->inv, rdata,
+ PJSIP_SC_NOT_ACCEPTABLE,
+ NULL, NULL, &tdata);
+ if (status == PJ_SUCCESS)
+ pjsip_inv_send_msg(call->inv, tdata);
+ else
+ pjsip_inv_terminate(call->inv, 500, PJ_FALSE);
+ return;
+ }
+
/* Send the 200 response. */
status = pjsip_inv_send_msg(call->inv, tdata);
@@ -562,6 +625,10 @@
/* Callback to be called to handle incoming requests outside dialogs: */
static pj_bool_t on_rx_request( pjsip_rx_data *rdata )
{
+ /* Ignore strandled ACKs (must not send respone */
+ if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD)
+ return PJ_FALSE;
+
/* Respond (statelessly) any non-INVITE requests with 500 */
if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) {
pj_str_t reason = pj_str("Unsupported Operation");
@@ -583,18 +650,50 @@
static void call_on_state_changed( pjsip_inv_session *inv,
pjsip_event *e)
{
+ struct call *call = inv->mod_data[mod_siprtp.id];
+
PJ_UNUSED_ARG(e);
- if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
- struct call *call = inv->mod_data[mod_siprtp.id];
+ if (!call)
+ return;
- if (!call)
- return;
+ if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
+
+ pj_time_val null_time = {0, 0};
call->inv = NULL;
inv->mod_data[mod_siprtp.id] = NULL;
destroy_call_media(call->index);
+
+ call->start_time = null_time;
+ call->response_time = null_time;
+ call->connect_time = null_time;
+
+ PJ_LOG(3,(THIS_FILE, "Call #%d disconnected. Reason=%s",
+ call->index,
+ pjsip_get_status_text(inv->cause)->ptr));
+
+ } else if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
+
+ pj_time_val t;
+
+ pj_gettimeofday(&call->connect_time);
+ if (call->response_time.sec == 0)
+ call->response_time = call->connect_time;
+
+ t = call->connect_time;
+ PJ_TIME_VAL_SUB(t, call->start_time);
+
+ PJ_LOG(3,(THIS_FILE, "Call #%d connected in %d ms", call->index,
+ PJ_TIME_VAL_MSEC(t)));
+
+ } else if ( inv->state == PJSIP_INV_STATE_EARLY ||
+ inv->state == PJSIP_INV_STATE_CONNECTING) {
+
+ if (call->response_time.sec == 0)
+ pj_gettimeofday(&call->response_time);
+
}
}
@@ -626,16 +725,30 @@
/* Usage */
static const char *USAGE =
-"Usage: \n"
-" siprtp [options] => to start in server mode \n"
-" siprtp [options] URL => to start in client mode \n"
+"Usage:\n"
+" siprtp [options] => to start in server mode\n"
+" siprtp [options] URL => to start in client mode\n"
"\n"
-"where options are: \n"
-" --count=N, -c Set number of calls to create (default:1) \n"
-" --port=PORT -p Set local SIP port (default: 5060) \n"
-" --rtp-port=PORT -r Set start of RTP port (default: 4000) \n"
-" --ip-addr=IP -i Set local IP address to use (otherwise it will\n"
+"Program options:\n"
+" --count=N, -c Set number of calls to create (default:1) \n"
+"\n"
+"Address and ports options:\n"
+" --local-port=PORT,-p Set local SIP port (default: 5060)\n"
+" --rtp-port=PORT, -r Set start of RTP port (default: 4000)\n"
+" --ip-addr=IP, -i Set local IP address to use (otherwise it will\n"
" try to determine local IP address from hostname)\n"
+"\n"
+"Logging Options:\n"
+" --log-level=N, -l Set log verbosity level (default=5)\n"
+" --app-log-level=N Set app screen log verbosity (default=3)\n"
+" --log-file=FILE Write log to file FILE\n"
+"\n"
+"Codec Options:\n"
+" --a-pt=PT Set audio payload type to PT (default=0)\n"
+" --a-name=NAME Set audio codec name to NAME (default=pcmu)\n"
+" --a-clock=RATE Set audio codec rate to RATE Hz (default=8000 Hz)\n"
+" --a-bitrate=BPS Set audio codec bitrate to BPS (default=64000 bps)\n"
+" --a-ptime=MS Set audio frame time to MS msec (default=20 msec)\n"
;
@@ -645,11 +758,25 @@
static char ip_addr[32];
static char local_uri[64];
+ enum { OPT_START,
+ OPT_APP_LOG_LEVEL, OPT_LOG_FILE,
+ OPT_A_PT, OPT_A_NAME, OPT_A_CLOCK, OPT_A_BITRATE, OPT_A_PTIME };
+
struct pj_getopt_option long_options[] = {
- { "count", 1, 0, 'c' },
- { "port", 1, 0, 'p' },
- { "rtp-port", 1, 0, 'r' },
- { "ip-addr", 1, 0, 'i' },
+ { "count", 1, 0, 'c' },
+ { "local-port", 1, 0, 'p' },
+ { "rtp-port", 1, 0, 'r' },
+ { "ip-addr", 1, 0, 'i' },
+
+ { "log-level", 1, 0, 'l' },
+ { "app-log-level", 1, 0, OPT_APP_LOG_LEVEL },
+ { "log-file", 1, 0, OPT_LOG_FILE },
+ { "a-pt", 1, 0, OPT_A_PT },
+ { "a-name", 1, 0, OPT_A_NAME },
+ { "a-clock", 1, 0, OPT_A_CLOCK },
+ { "a-bitrate", 1, 0, OPT_A_BITRATE },
+ { "a-ptime", 1, 0, OPT_A_PTIME },
+
{ NULL, 0, 0, 0 },
};
int c;
@@ -667,16 +794,22 @@
pj_ansi_strcpy(ip_addr, addr);
}
- /* Init default */
+ /* Init defaults */
app.max_calls = 1;
app.thread_count = 1;
app.sip_port = 5060;
app.rtp_start_port = 4000;
app.local_addr = ip_addr;
+ app.log_level = 5;
+ app.app_log_level = 3;
+ app.log_filename = NULL;
+
+ /* Default codecs: */
+ app.audio_codec = audio_codecs[0];
/* Parse options */
pj_optind = 0;
- while((c=pj_getopt_long(argc,argv, "c:p:r:i:",
+ while((c=pj_getopt_long(argc,argv, "c:p:r:i:l:",
long_options, &option_index))!=-1)
{
switch (c) {
@@ -696,6 +829,33 @@
case 'i':
app.local_addr = pj_optarg;
break;
+
+ case 'l':
+ app.log_level = atoi(pj_optarg);
+ break;
+ case OPT_APP_LOG_LEVEL:
+ app.app_log_level = atoi(pj_optarg);
+ break;
+ case OPT_LOG_FILE:
+ app.log_filename = pj_optarg;
+ break;
+
+ case OPT_A_PT:
+ app.audio_codec.pt = atoi(pj_optarg);
+ break;
+ case OPT_A_NAME:
+ app.audio_codec.name = pj_optarg;
+ break;
+ case OPT_A_CLOCK:
+ app.audio_codec.clock_rate = atoi(pj_optarg);
+ break;
+ case OPT_A_BITRATE:
+ app.audio_codec.bit_rate = atoi(pj_optarg);
+ break;
+ case OPT_A_PTIME:
+ app.audio_codec.ptime = atoi(pj_optarg);
+ break;
+
default:
puts(USAGE);
return 1;
@@ -716,8 +876,7 @@
}
-//////////////////////////////////////////////////////////////////////////////
-/*
+/*****************************************************************************
* MEDIA STUFFS
*/
@@ -780,13 +939,13 @@
{
pjmedia_sdp_rtpmap rtpmap;
pjmedia_sdp_attr *attr;
+ char ptstr[10];
- PJ_TODO(PARAMETERIZE_CODEC);
-
- m->desc.fmt[0] = pj_str("0");
- rtpmap.pt = pj_str("0");
- rtpmap.clock_rate = 8000;
- rtpmap.enc_name = pj_str("pcmu");
+ sprintf(ptstr, "%d", app.audio_codec.pt);
+ pj_strdup2(pool, &m->desc.fmt[0], ptstr);
+ rtpmap.pt = m->desc.fmt[0];
+ rtpmap.clock_rate = app.audio_codec.clock_rate;
+ rtpmap.enc_name = pj_str(app.audio_codec.name);
rtpmap.param.slen = 0;
pjmedia_sdp_rtpmap_to_attr(pool, &rtpmap, &attr);
@@ -802,16 +961,16 @@
/*
* Add support telephony event
*/
- m->desc.fmt[m->desc.fmt_count++] = pj_str("101");
+ m->desc.fmt[m->desc.fmt_count++] = pj_str("121");
/* Add rtpmap. */
attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
attr->name = pj_str("rtpmap");
- attr->value = pj_str(":101 telephone-event/8000");
+ attr->value = pj_str(":121 telephone-event/8000");
m->attr[m->attr_count++] = attr;
/* Add fmtp */
attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
attr->name = pj_str("fmtp");
- attr->value = pj_str(":101 0-15");
+ attr->value = pj_str(":121 0-15");
m->attr[m->attr_count++] = attr;
#endif
@@ -822,7 +981,11 @@
}
-/* Media thread */
+/*
+ * Media thread
+ *
+ * This is the thread to send and receive both RTP and RTCP packets.
+ */
static int media_thread(void *arg)
{
struct media_stream *strm = arg;
@@ -880,6 +1043,9 @@
continue;
}
+ ++strm->rx_stat.pkt;
+ strm->rx_stat.payload += (size - 12);
+
/* Decode RTP packet. */
status = pjmedia_rtp_decode_rtp(&strm->in_sess,
packet, size,
@@ -887,6 +1053,7 @@
&payload, &payload_len);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "RTP decode error", status);
+ strm->rx_stat.discard++;
continue;
}
@@ -899,6 +1066,7 @@
app_perror(THIS_FILE, "RTP update error", status);
PJ_LOG(3,(THIS_FILE,"RTP packet detail: pt=%d, seq=%d",
hdr->pt, pj_ntohs(hdr->seq)));
+ strm->rx_stat.discard++;
continue;
}
@@ -919,10 +1087,29 @@
if (status != PJ_SUCCESS)
app_perror(THIS_FILE, "Error receiving RTCP packet", status);
else {
- if (size > sizeof(strm->rem_rtcp))
+ if (size > sizeof(strm->rem_rtcp)) {
PJ_LOG(3,(THIS_FILE, "Error: RTCP packet too large"));
- else
+ status = -1;
+ } else {
pj_memcpy(&strm->rem_rtcp, packet, size);
+ status = PJ_SUCCESS;
+ }
+ }
+
+ if (status == PJ_SUCCESS) {
+ /* Process RTCP stats */
+ unsigned jitter;
+
+ jitter = pj_ntohl(strm->rem_rtcp.rr.jitter) * 1000 /
+ strm->clock_rate;
+ if (jitter < strm->tx_stat.jitter_min)
+ strm->tx_stat.jitter_min = jitter;
+ if (jitter > strm->tx_stat.jitter_max)
+ strm->tx_stat.jitter_max = jitter;
+ strm->tx_stat.jitter_avg = (strm->tx_stat.jitter_avg * strm->tx_stat.rtcp_cnt +
+ jitter) / (strm->tx_stat.rtcp_cnt + 1);
+
+ strm->tx_stat.rtcp_cnt++;
}
}
@@ -969,6 +1156,10 @@
/* Schedule next send */
next_rtp.msec += strm->samples_per_frame * 1000 / strm->clock_rate;
pj_time_val_normalize(&next_rtp);
+
+ /* Update stats */
+ strm->tx_stat.pkt++;
+ strm->tx_stat.payload += strm->bytes_per_frame;
}
@@ -1001,6 +1192,22 @@
}
+ /* Process RTCP stats */
+ {
+ unsigned jitter;
+
+ jitter = pj_ntohl(rtcp_pkt->rr.jitter) * 1000 /
+ strm->clock_rate;
+ if (jitter < strm->rx_stat.jitter_min)
+ strm->rx_stat.jitter_min = jitter;
+ if (jitter > strm->rx_stat.jitter_max)
+ strm->rx_stat.jitter_max = jitter;
+ strm->rx_stat.jitter_avg = (strm->rx_stat.jitter_avg * strm->rx_stat.rtcp_cnt +
+ jitter) / (strm->rx_stat.rtcp_cnt + 1);
+
+ strm->rx_stat.rtcp_cnt++;
+ }
+
next_rtcp.sec += 5;
}
@@ -1018,7 +1225,8 @@
pj_pool_t *pool;
struct media_stream *audio;
pjmedia_sdp_session *local_sdp, *remote_sdp;
-
+ struct codec *codec_desc = NULL;
+ unsigned i;
call = inv->mod_data[mod_siprtp.id];
pool = inv->dlg->pool;
@@ -1047,11 +1255,27 @@
return;
}
+ /* Get the remainder of codec information from codec descriptor */
+ if (audio->si.fmt.pt == app.audio_codec.pt)
+ codec_desc = &app.audio_codec;
+ else {
+ /* Find the codec description in codec array */
+ for (i=0; i<PJ_ARRAY_SIZE(audio_codecs); ++i) {
+ if (audio_codecs[i].pt == audio->si.fmt.pt) {
+ codec_desc = &audio_codecs[i];
+ break;
+ }
+ }
+
+ if (codec_desc == NULL) {
+ PJ_LOG(3, (THIS_FILE, "Error: Invalid codec payload type"));
+ return;
+ }
+ }
audio->clock_rate = audio->si.fmt.sample_rate;
- audio->samples_per_frame = audio->clock_rate * 20 / 1000;
- audio->bytes_per_frame = 160;
- PJ_TODO(TAKE_CODEC_INFO_FROM_ARGUMENT);
+ audio->samples_per_frame = audio->clock_rate * codec_desc->ptime / 1000;
+ audio->bytes_per_frame = codec_desc->bit_rate * codec_desc->ptime / 1000 / 8;
pjmedia_rtp_session_init(&audio->out_sess, audio->si.tx_pt,
@@ -1059,6 +1283,12 @@
pjmedia_rtp_session_init(&audio->in_sess, audio->si.fmt.pt, 0);
pjmedia_rtcp_init(&audio->rtcp, 0);
+
+ /* Clear media statistics */
+ pj_memset(&audio->rx_stat, 0, sizeof(audio->rx_stat));
+ pj_memset(&audio->tx_stat, 0, sizeof(audio->tx_stat));
+
+
/* Start media thread. */
audio->thread_quit_flag = 0;
status = pj_thread_create( inv->pool, "media", &media_thread, audio,
@@ -1081,12 +1311,39 @@
pj_thread_destroy(audio->thread);
audio->thread = NULL;
audio->thread_quit_flag = 0;
+
+ /* Flush RTP/RTCP packets */
+ {
+ pj_fd_set_t set;
+ pj_time_val timeout = {0, 0};
+ char packet[1500];
+ pj_ssize_t size;
+ pj_status_t status;
+ int rc;
+
+ do {
+ PJ_FD_ZERO(&set);
+ PJ_FD_SET(audio->rtp_sock, &set);
+ PJ_FD_SET(audio->rtcp_sock, &set);
+
+ rc = pj_sock_select(FD_SETSIZE, &set, NULL, NULL, &timeout);
+ if (rc > 0 && PJ_FD_ISSET(audio->rtp_sock, &set)) {
+ size = sizeof(packet);
+ status = pj_sock_recv(audio->rtp_sock, packet, &size, 0);
+
+ }
+ if (rc > 0 && PJ_FD_ISSET(audio->rtcp_sock, &set)) {
+ size = sizeof(packet);
+ status = pj_sock_recv(audio->rtcp_sock, packet, &size, 0);
+ }
+
+ } while (rc > 0);
+ }
}
}
-/////////////////////////////////////////////////////////////////////////////
-/*
+/*****************************************************************************
* USER INTERFACE STUFFS
*/
@@ -1110,41 +1367,144 @@
static void print_call(int call_index)
{
+ struct call *call = &app.call[call_index];
int len;
- pjsip_inv_session *inv = app.call[call_index].inv;
+ pjsip_inv_session *inv = call->inv;
pjsip_dialog *dlg = inv->dlg;
- struct media_stream *audio = &app.call[call_index].media[0];
+ struct media_stream *audio = &call->media[0];
char userinfo[128];
- char packets[16];
+ char duration[80];
+ char bps[16], ipbps[16], packets[16], bytes[16], ipbytes[16];
+ pj_uint32_t total_loss;
- /* Dump invite sesion info. */
+ /* Print duration */
+ if (inv->state == PJSIP_INV_STATE_CONFIRMED) {
+ pj_time_val now;
+
+ pj_gettimeofday(&now);
+ PJ_TIME_VAL_SUB(now, call->connect_time);
+
+ sprintf(duration, " [duration: %02d:%02d:%02d.%03d]",
+ 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("Call #%d: %s\n", call_index, pjsip_inv_state_name(inv->state));
+
printf(" %s\n", userinfo);
- if (app.call[call_index].media[0].thread == NULL) {
+
+ /* 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 %d 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: %d ms", PJ_TIME_VAL_MSEC(t));
+ } else {
+ connectdelay[0] = '\0';
+ }
+
+ printf(" Signaling quality: %s%s\n", pdd, connectdelay);
+ }
+
+
+ if (call->media[0].thread == NULL) {
return;
}
- printf(" Stream #0: audio %.*s@%dHz, %d bytes/sec\n",
+ printf(" Stream #0: audio %.*s@%dHz, %dms/frame, %sbps (%sbps +IP hdr)\n",
(int)audio->si.fmt.encoding_name.slen,
audio->si.fmt.encoding_name.ptr,
audio->clock_rate,
- audio->bytes_per_frame * audio->clock_rate / audio->samples_per_frame);
- printf(" RX pkt=%s, fraction lost=%5.2f%%, jitter=%dms\n",
- good_number(packets, audio->rtcp.received),
- audio->rtcp.rtcp_pkt.rr.fract_lost/255.0,
- pj_ntohl(audio->rtcp.rtcp_pkt.rr.jitter) * 1000 / audio->clock_rate);
- printf(" TX pkt=%s, fraction lost=%5.2f%%, jitter=%dms\n",
- good_number(packets, pj_ntohl(audio->rtcp.rtcp_pkt.sr.sender_pcount)),
- audio->rem_rtcp.rr.fract_lost/255.0,
- pj_ntohl(audio->rem_rtcp.rr.jitter) * 1000 / 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));
+
+ total_loss = (audio->rtcp.rtcp_pkt.rr.total_lost_2 << 16) +
+ (audio->rtcp.rtcp_pkt.rr.total_lost_1 << 8) +
+ audio->rtcp.rtcp_pkt.rr.total_lost_0;
+
+ printf(" RX total %s packets %sB received (%sB +IP hdr)%s\n"
+ " pkt discards=%d (%3.1f%%), loss=%d (%3.1f%%), reorder=%d (%3.1f%%)%s\n"
+ " loss period min=%d ms, avg=%d ms, max=%d ms%s\n"
+ " jitter min=%d ms, avg=%d ms, max=%d ms, current=%d ms%s\n",
+ good_number(packets, audio->rx_stat.pkt),
+ good_number(bytes, audio->rx_stat.payload),
+ good_number(ipbytes, audio->rx_stat.payload + audio->rx_stat.pkt * 32),
+ "",
+ audio->rx_stat.discard,
+ audio->rx_stat.discard * 100.0 / audio->rx_stat.pkt,
+ total_loss,
+ total_loss * 100.0 / audio->rx_stat.pkt,
+ 0, 0.0,
+ "",
+ -1, -1, -1,
+ "",
+ (audio->rx_stat.rtcp_cnt ? audio->rx_stat.jitter_min : -1),
+ (audio->rx_stat.rtcp_cnt ? audio->rx_stat.jitter_avg : -1),
+ (audio->rx_stat.rtcp_cnt ? audio->rx_stat.jitter_max : -1),
+ (audio->rx_stat.rtcp_cnt ? pj_ntohl(audio->rtcp.rtcp_pkt.rr.jitter)*1000/audio->clock_rate : -1),
+ ""
+ );
+
+
+ total_loss = (audio->rem_rtcp.rr.total_lost_2 << 16) +
+ (audio->rem_rtcp.rr.total_lost_1 << 8) +
+ audio->rem_rtcp.rr.total_lost_0;
+
+ printf(" TX total %s packets %sB sent (%sB +IP hdr)%s\n"
+ " pkt discards=%d (%3.1f%%), loss=%d (%3.1f%%), reorder=%d (%3.1f%%)%s\n"
+ " loss period min=%d ms, avg=%d ms, max=%d ms%s\n"
+ " jitter min=%d ms, avg=%d ms, max=%d ms, current=%d ms%s\n",
+ good_number(packets, audio->tx_stat.pkt),
+ good_number(bytes, audio->tx_stat.payload),
+ good_number(ipbytes, audio->tx_stat.payload + audio->tx_stat.pkt * 32),
+ "",
+ audio->tx_stat.discard,
+ audio->tx_stat.discard * 100.0 / audio->tx_stat.pkt,
+ total_loss,
+ total_loss * 100.0 / audio->tx_stat.pkt,
+ 0, 0.0,
+ "",
+ -1, -1, -1,
+ "",
+ (audio->tx_stat.rtcp_cnt ? audio->tx_stat.jitter_min : -1),
+ (audio->tx_stat.rtcp_cnt ? audio->tx_stat.jitter_avg : -1),
+ (audio->tx_stat.rtcp_cnt ? audio->tx_stat.jitter_max : -1),
+ (audio->tx_stat.rtcp_cnt ? pj_ntohl(audio->rem_rtcp.rr.jitter)*1000/audio->clock_rate : -1),
+ ""
+ );
+
}
@@ -1218,6 +1578,8 @@
char input1[10];
unsigned i;
+ printf("%s", MENU);
+
for (;;) {
printf(">>> "); fflush(stdout);
fgets(input1, sizeof(input1), stdin);
@@ -1243,6 +1605,7 @@
goto on_exit;
default:
+ puts("Invalid command");
printf("%s", MENU);
break;
}
@@ -1251,12 +1614,17 @@
}
on_exit:
- ;
+ hangup_all_calls();
}
+/*****************************************************************************
+ * Below is a simple module to log all incoming and outgoing SIP messages
+ */
+
+
/* Notification on incoming messages */
-static pj_bool_t console_on_rx_msg(pjsip_rx_data *rdata)
+static pj_bool_t logger_on_rx_msg(pjsip_rx_data *rdata)
{
PJ_LOG(4,(THIS_FILE, "RX %d bytes %s from %s:%d:\n"
"%s\n"
@@ -1272,7 +1640,7 @@
}
/* Notification on outgoing messages */
-static pj_status_t console_on_tx_msg(pjsip_tx_data *tdata)
+static pj_status_t logger_on_tx_msg(pjsip_tx_data *tdata)
{
/* Important note:
@@ -1305,32 +1673,94 @@
NULL, /* start() */
NULL, /* stop() */
NULL, /* unload() */
- &console_on_rx_msg, /* on_rx_request() */
- &console_on_rx_msg, /* on_rx_response() */
- &console_on_tx_msg, /* on_tx_request. */
- &console_on_tx_msg, /* on_tx_response() */
+ &logger_on_rx_msg, /* on_rx_request() */
+ &logger_on_rx_msg, /* on_rx_response() */
+ &logger_on_tx_msg, /* on_tx_request. */
+ &logger_on_tx_msg, /* on_tx_response() */
NULL, /* on_tsx_state() */
};
+/*****************************************************************************
+ * Console application custom logging:
+ */
+
+
+static FILE *log_file;
+
+
+static void app_log_writer(int level, const char *buffer, int len)
+{
+ /* Write to both stdout and file. */
+
+ if (level <= app.app_log_level)
+ pj_log_write(level, buffer, len);
+
+ if (log_file) {
+ fwrite(buffer, len, 1, log_file);
+ fflush(log_file);
+ }
+}
+
+
+pj_status_t app_logging_init(void)
+{
+ /* Redirect log function to ours */
+
+ pj_log_set_log_func( &app_log_writer );
+
+ /* If output log file is desired, create the file: */
+
+ if (app.log_filename) {
+ log_file = fopen(app.log_filename, "wt");
+ if (log_file == NULL) {
+ PJ_LOG(1,(THIS_FILE, "Unable to open log file %s",
+ app.log_filename));
+ return -1;
+ }
+ }
+
+ return PJ_SUCCESS;
+}
+
+
+void app_logging_shutdown(void)
+{
+ /* Close logging file, if any: */
+
+ if (log_file) {
+ fclose(log_file);
+ log_file = NULL;
+ }
+}
+
/*
* main()
*/
int main(int argc, char *argv[])
{
+ unsigned i;
pj_status_t status;
+ /* Must init PJLIB first */
status = pj_init();
if (status != PJ_SUCCESS)
return 1;
+ /* Get command line options */
status = init_options(argc, argv);
if (status != PJ_SUCCESS)
return 1;
+ /* Init logging */
+ status = app_logging_init();
+ if (status != PJ_SUCCESS)
+ return 1;
+
+ /* Init SIP etc */
status = init_sip();
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Initialization has failed", status);
@@ -1338,8 +1768,10 @@
return 1;
}
+ /* Register module to log incoming/outgoing messages */
pjsip_endpt_register_module(app.sip_endpt, &msg_logger);
+ /* Init media */
status = init_media();
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Media initialization failed", status);
@@ -1347,9 +1779,13 @@
return 1;
}
+ /* If URL is specified, then make call immediately */
if (app.uri_to_call.slen) {
unsigned i;
+ PJ_LOG(3,(THIS_FILE, "Making %d calls to %s..", app.max_calls,
+ app.uri_to_call.ptr));
+
for (i=0; i<app.max_calls; ++i) {
status = make_call(&app.uri_to_call);
if (status != PJ_SUCCESS) {
@@ -1357,13 +1793,28 @@
break;
}
}
+
+ } else {
+
+ PJ_LOG(3,(THIS_FILE, "Ready for incoming calls (max=%d)",
+ app.max_calls));
}
-
+
+ /* Start worker threads */
+ for (i=0; i<app.thread_count; ++i) {
+ pj_thread_create( app.pool, "app", &worker_thread, NULL,
+ 0, 0, &app.thread[i]);
+ }
+
+ /* Start user interface loop */
console_main();
-
+
+ /* Shutting down... */
destroy_media();
destroy_sip();
+ app_logging_shutdown();
+
return 0;
}