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review.jami.net / pjproject / e960bb54f90de433209c91d50dac7a288b7a9e95 / . / pjsip-apps / src / samples / siprtp.c
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/* $Id$ */
/*
* 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
*/
/* Usage */
static const char *USAGE =
" PURPOSE: \n"
" This program establishes SIP INVITE session and media, and calculate \n"
" the media quality (packet lost, jitter, rtt, etc.). Unlike normal \n"
" pjmedia applications, this program bypasses all pjmedia stream \n"
" framework and transmit encoded RTP packets manually using own thread. \n"
"\n"
" USAGE:\n"
" siprtp [options] => to start in server mode\n"
" siprtp [options] URL => to start in client mode\n"
"\n"
" Program options:\n"
" --count=N, -c Set number of calls to create (default:1) \n"
" --duration=SEC, -d Set maximum call duration (default:unlimited) \n"
" --auto-quit, -q Quit when calls have been completed (default:no)\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"
" --report-file=FILE Write report to file FILE\n"
"\n"
/* Don't support this anymore, because codec is properly examined in
pjmedia_session_info_from_sdp() function.
" 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=8000Hz)\n"
" --a-bitrate=BPS Set audio codec bitrate to BPS (default=64000bps)\n"
" --a-ptime=MS Set audio frame time to MS msec (default=20ms)\n"
*/
;
/* Include all headers. */
#include <pjsip.h>
#include <pjmedia.h>
#include <pjmedia-codec.h>
#include <pjsip_ua.h>
#include <pjsip_simple.h>
#include <pjlib-util.h>
#include <pjlib.h>
#include <stdlib.h>
#if PJ_HAS_HIGH_RES_TIMER==0
# error "High resolution timer is needed for this sample"
#endif
#define THIS_FILE "siprtp.c"
#define MAX_CALLS 1024
#define RTP_START_PORT 4000
/* Codec descriptor: */
struct codec
{
unsigned pt;
char* name;
unsigned clock_rate;
unsigned bit_rate;
unsigned ptime;
char* description;
};
/* A bidirectional media stream created when the call is active. */
struct media_stream
{
/* Static: */
unsigned call_index; /* Call owner. */
unsigned media_index; /* Media index in call. */
pjmedia_transport *transport; /* To send/recv RTP/RTCP */
/* Active? */
pj_bool_t active; /* Non-zero if is in call. */
/* Current stream info: */
pjmedia_stream_info si; /* Current stream info. */
/* More info: */
unsigned clock_rate; /* clock rate */
unsigned samples_per_frame; /* samples per frame */
unsigned bytes_per_frame; /* frame size. */
/* RTP session: */
pjmedia_rtp_session out_sess; /* outgoing RTP session */
pjmedia_rtp_session in_sess; /* incoming RTP session */
/* RTCP stats: */
pjmedia_rtcp_session rtcp; /* incoming RTCP session. */
/* Thread: */
pj_bool_t thread_quit_flag; /* Stop media thread. */
pj_thread_t *thread; /* Media thread. */
};
/* This is a call structure that is created when the application starts
* and only destroyed when the application quits.
*/
struct call
{
unsigned index;
pjsip_inv_session *inv;
unsigned media_count;
struct media_stream media[1];
pj_time_val start_time;
pj_time_val response_time;
pj_time_val connect_time;
pj_timer_entry d_timer; /**< Disconnect timer. */
};
/* Application's global variables */
static struct app
{
unsigned max_calls;
unsigned uac_calls;
unsigned duration;
pj_bool_t auto_quit;
unsigned thread_count;
int sip_port;
int rtp_start_port;
pj_str_t local_addr;
pj_str_t local_uri;
pj_str_t local_contact;
int app_log_level;
int log_level;
char *log_filename;
char *report_filename;
struct codec audio_codec;
pj_str_t uri_to_call;
pj_caching_pool cp;
pj_pool_t *pool;
pjsip_endpoint *sip_endpt;
pj_bool_t thread_quit;
pj_thread_t *sip_thread[1];
pjmedia_endpt *med_endpt;
struct call call[MAX_CALLS];
} app;
/*
* Prototypes:
*/
/* Callback to be called when SDP negotiation is done in the call: */
static void call_on_media_update( pjsip_inv_session *inv,
pj_status_t status);
/* Callback to be called when invite session's state has changed: */
static void call_on_state_changed( pjsip_inv_session *inv,
pjsip_event *e);
/* Callback to be called when dialog has forked: */
static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e);
/* Callback to be called to handle incoming requests outside dialogs: */
static pj_bool_t on_rx_request( pjsip_rx_data *rdata );
/* Worker thread prototype */
static int sip_worker_thread(void *arg);
/* Create SDP for call */
static pj_status_t create_sdp( pj_pool_t *pool,
struct call *call,
pjmedia_sdp_session **p_sdp);
/* Hangup call */
static void hangup_call(unsigned index);
/* Destroy the call's media */
static void destroy_call_media(unsigned call_index);
/* Destroy media. */
static void destroy_media();
/* This callback is called by media transport on receipt of RTP packet. */
static void on_rx_rtp(void *user_data, const void *pkt, pj_ssize_t size);
/* This callback is called by media transport on receipt of RTCP packet. */
static void on_rx_rtcp(void *user_data, const void *pkt, pj_ssize_t size);
/* Display error */
static void app_perror(const char *sender, const char *title,
pj_status_t status);
/* Print call */
static void print_call(int call_index);
/* 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
* create a dialog and INVITE session for it.
*/
static pjsip_module mod_siprtp =
{
NULL, NULL, /* prev, next. */
{ "mod-siprtpapp", 13 }, /* Name. */
-1, /* Id */
PJSIP_MOD_PRIORITY_APPLICATION, /* Priority */
NULL, /* load() */
NULL, /* start() */
NULL, /* stop() */
NULL, /* unload() */
&on_rx_request, /* on_rx_request() */
NULL, /* on_rx_response() */
NULL, /* on_tx_request. */
NULL, /* on_tx_response() */
NULL, /* on_tsx_state() */
};
/* 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: */
status = pjlib_util_init();
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
/* Must create a pool factory before we can allocate any memory. */
pj_caching_pool_init(&app.cp, &pj_pool_factory_default_policy, 0);
/* Create application pool for misc. */
app.pool = pj_pool_create(&app.cp.factory, "app", 1000, 1000, NULL);
/* Create the endpoint: */
status = pjsip_endpt_create(&app.cp.factory, pj_gethostname()->ptr,
&app.sip_endpt);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
/* Add UDP transport. */
{
pj_sockaddr_in addr;
pjsip_host_port addrname;
pjsip_transport *tp;
pj_bzero(&addr, sizeof(addr));
addr.sin_family = PJ_AF_INET;
addr.sin_addr.s_addr = 0;
addr.sin_port = pj_htons((pj_uint16_t)app.sip_port);
if (app.local_addr.slen) {
addrname.host = app.local_addr;
addrname.port = app.sip_port;
status = pj_sockaddr_in_init(&addr, &app.local_addr,
(pj_uint16_t)app.sip_port);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable to resolve IP interface", status);
return status;
}
}
status = pjsip_udp_transport_start( app.sip_endpt, &addr,
(app.local_addr.slen ? &addrname:NULL),
1, &tp);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Unable to start UDP transport", status);
return status;
}
PJ_LOG(3,(THIS_FILE, "SIP UDP listening on %.*s:%d",
(int)tp->local_name.host.slen, tp->local_name.host.ptr,
tp->local_name.port));
}
/*
* Init transaction layer.
* This will create/initialize transaction hash tables etc.
*/
status = pjsip_tsx_layer_init_module(app.sip_endpt);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
/* Initialize UA layer. */
status = pjsip_ua_init_module( app.sip_endpt, NULL );
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
/* Init invite session module. */
{
pjsip_inv_callback inv_cb;
/* Init the callback for INVITE session: */
pj_bzero(&inv_cb, sizeof(inv_cb));
inv_cb.on_state_changed = &call_on_state_changed;
inv_cb.on_new_session = &call_on_forked;
inv_cb.on_media_update = &call_on_media_update;
/* Initialize invite session module: */
status = pjsip_inv_usage_init(app.sip_endpt, &inv_cb);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
}
/* Register our module to receive incoming requests. */
status = pjsip_endpt_register_module( app.sip_endpt, &mod_siprtp);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
/* Init calls */
for (i=0; i<app.max_calls; ++i)
app.call[i].index = i;
/* Done */
return PJ_SUCCESS;
}
/*
* Destroy SIP
*/
static void destroy_sip()
{
unsigned i;
app.thread_quit = 1;
for (i=0; i<app.thread_count; ++i) {
if (app.sip_thread[i]) {
pj_thread_join(app.sip_thread[i]);
pj_thread_destroy(app.sip_thread[i]);
app.sip_thread[i] = NULL;
}
}
if (app.sip_endpt) {
pjsip_endpt_destroy(app.sip_endpt);
app.sip_endpt = NULL;
}
}
/*
* Init media stack.
*/
static pj_status_t init_media()
{
unsigned i, count;
pj_uint16_t rtp_port;
pj_status_t status;
/* Initialize media endpoint so that at least error subsystem is properly
* initialized.
*/
status = pjmedia_endpt_create(&app.cp.factory, NULL, 1, &app.med_endpt);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
/* Must register codecs to be supported */
pjmedia_codec_g711_init(app.med_endpt);
/* RTP port counter */
rtp_port = (pj_uint16_t)(app.rtp_start_port & 0xFFFE);
/* Init media transport for all calls. */
for (i=0, count=0; i<app.max_calls; ++i, ++count) {
unsigned j;
/* Create transport for each media in the call */
for (j=0; j<PJ_ARRAY_SIZE(app.call[0].media); ++j) {
/* Repeat binding media socket to next port when fails to bind
* to current port number.
*/
int retry;
app.call[i].media[j].call_index = i;
app.call[i].media[j].media_index = j;
status = -1;
for (retry=0; retry<100; ++retry,rtp_port+=2) {
struct media_stream *m = &app.call[i].media[j];
status = pjmedia_transport_udp_create2(app.med_endpt,
"siprtp",
&app.local_addr,
rtp_port, 0,
&m->transport);
if (status == PJ_SUCCESS) {
rtp_port += 2;
break;
}
}
}
if (status != PJ_SUCCESS)
goto on_error;
}
/* Done */
return PJ_SUCCESS;
on_error:
destroy_media();
return status;
}
/*
* Destroy media.
*/
static void destroy_media()
{
unsigned i;
for (i=0; i<app.max_calls; ++i) {
unsigned j;
for (j=0; j<PJ_ARRAY_SIZE(app.call[0].media); ++j) {
struct media_stream *m = &app.call[i].media[j];
if (m->transport) {
pjmedia_transport_close(m->transport);
m->transport = NULL;
}
}
}
if (app.med_endpt) {
pjmedia_endpt_destroy(app.med_endpt);
app.med_endpt = NULL;
}
}
/*
* Make outgoing call.
*/
static pj_status_t make_call(const pj_str_t *dst_uri)
{
unsigned i;
struct call *call;
pjsip_dialog *dlg;
pjmedia_sdp_session *sdp;
pjsip_tx_data *tdata;
pj_status_t status;
/* Find unused call slot */
for (i=0; i<app.max_calls; ++i) {
if (app.call[i].inv == NULL)
break;
}
if (i == app.max_calls)
return PJ_ETOOMANY;
call = &app.call[i];
/* Create UAC dialog */
status = pjsip_dlg_create_uac( pjsip_ua_instance(),
&app.local_uri, /* local URI */
&app.local_contact, /* local Contact */
dst_uri, /* remote URI */
dst_uri, /* remote target */
&dlg); /* dialog */
if (status != PJ_SUCCESS) {
++app.uac_calls;
return status;
}
/* Create SDP */
create_sdp( dlg->pool, call, &sdp);
/* Create the INVITE session. */
status = pjsip_inv_create_uac( dlg, sdp, 0, &call->inv);
if (status != PJ_SUCCESS) {
pjsip_dlg_terminate(dlg);
++app.uac_calls;
return status;
}
/* 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
* an SDP body as well.
*/
status = pjsip_inv_invite(call->inv, &tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
/* Send initial INVITE request.
* From now on, the invite session's state will be reported to us
* via the invite session callbacks.
*/
status = pjsip_inv_send_msg(call->inv, tdata);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
return PJ_SUCCESS;
}
/*
* Receive incoming call
*/
static void process_incoming_call(pjsip_rx_data *rdata)
{
unsigned i, options;
struct call *call;
pjsip_dialog *dlg;
pjmedia_sdp_session *sdp;
pjsip_tx_data *tdata;
pj_status_t status;
/* Find free call slot */
for (i=0; i<app.max_calls; ++i) {
if (app.call[i].inv == NULL)
break;
}
if (i == app.max_calls) {
const pj_str_t reason = pj_str("Too many calls");
pjsip_endpt_respond_stateless( app.sip_endpt, rdata,
500, &reason,
NULL, NULL);
return;
}
call = &app.call[i];
/* Verify that we can handle the request. */
options = 0;
status = pjsip_inv_verify_request(rdata, &options, NULL, NULL,
app.sip_endpt, &tdata);
if (status != PJ_SUCCESS) {
/*
* No we can't handle the incoming INVITE request.
*/
if (tdata) {
pjsip_response_addr res_addr;
pjsip_get_response_addr(tdata->pool, rdata, &res_addr);
pjsip_endpt_send_response(app.sip_endpt, &res_addr, tdata,
NULL, NULL);
} else {
/* Respond with 500 (Internal Server Error) */
pjsip_endpt_respond_stateless(app.sip_endpt, rdata, 500, NULL,
NULL, NULL);
}
return;
}
/* Create UAS dialog */
status = pjsip_dlg_create_uas( pjsip_ua_instance(), rdata,
&app.local_contact, &dlg);
if (status != PJ_SUCCESS) {
const pj_str_t reason = pj_str("Unable to create dialog");
pjsip_endpt_respond_stateless( app.sip_endpt, rdata,
500, &reason,
NULL, NULL);
return;
}
/* Create SDP */
create_sdp( dlg->pool, call, &sdp);
/* Create UAS invite session */
status = pjsip_inv_create_uas( dlg, rdata, sdp, 0, &call->inv);
if (status != PJ_SUCCESS) {
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);
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);
PJ_ASSERT_ON_FAIL(status == PJ_SUCCESS, return);
/* Done */
}
/* Callback to be called when dialog has forked: */
static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e)
{
PJ_UNUSED_ARG(inv);
PJ_UNUSED_ARG(e);
PJ_TODO( HANDLE_FORKING );
}
/* 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");
pjsip_endpt_respond_stateless( app.sip_endpt, rdata,
500, &reason,
NULL, NULL);
return PJ_TRUE;
}
/* Handle incoming INVITE */
process_incoming_call(rdata);
/* Done */
return PJ_TRUE;
}
/* Callback timer to disconnect call (limiting call duration) */
static void timer_disconnect_call( pj_timer_heap_t *timer_heap,
struct pj_timer_entry *entry)
{
struct call *call = entry->user_data;
PJ_UNUSED_ARG(timer_heap);
entry->id = 0;
hangup_call(call->index);
}
/* Callback to be called when invite session's state has changed: */
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 (!call)
return;
if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {
pj_time_val null_time = {0, 0};
if (call->d_timer.id != 0) {
pjsip_endpt_cancel_timer(app.sip_endpt, &call->d_timer);
call->d_timer.id = 0;
}
PJ_LOG(3,(THIS_FILE, "Call #%d disconnected. Reason=%d (%.*s)",
call->index,
inv->cause,
(int)inv->cause_text.slen,
inv->cause_text.ptr));
PJ_LOG(3,(THIS_FILE, "Call #%d statistics:", call->index));
print_call(call->index);
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;
++app.uac_calls;
} 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)));
if (app.duration != 0) {
call->d_timer.id = 1;
call->d_timer.user_data = call;
call->d_timer.cb = &timer_disconnect_call;
t.sec = app.duration;
t.msec = 0;
pjsip_endpt_schedule_timer(app.sip_endpt, &call->d_timer, &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);
}
}
/* Utility */
static void app_perror(const char *sender, const char *title,
pj_status_t status)
{
char errmsg[PJ_ERR_MSG_SIZE];
pj_strerror(status, errmsg, sizeof(errmsg));
PJ_LOG(3,(sender, "%s: %s [status=%d]", title, errmsg, status));
}
/* Worker thread for SIP */
static int sip_worker_thread(void *arg)
{
PJ_UNUSED_ARG(arg);
while (!app.thread_quit) {
pj_time_val timeout = {0, 10};
pjsip_endpt_handle_events(app.sip_endpt, &timeout);
}
return 0;
}
/* Init application options */
static pj_status_t init_options(int argc, char *argv[])
{
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,
OPT_REPORT_FILE };
struct pj_getopt_option long_options[] = {
{ "count", 1, 0, 'c' },
{ "duration", 1, 0, 'd' },
{ "auto-quit", 0, 0, 'q' },
{ "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 },
{ "report-file", 1, 0, OPT_REPORT_FILE },
/* Don't support this anymore, see comments in USAGE above.
{ "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;
int option_index;
/* Get local IP address for the default IP address */
{
const pj_str_t *hostname;
pj_sockaddr_in tmp_addr;
char *addr;
hostname = pj_gethostname();
pj_sockaddr_in_init(&tmp_addr, hostname, 0);
addr = pj_inet_ntoa(tmp_addr.sin_addr);
pj_ansi_strcpy(ip_addr, addr);
}
/* Init defaults */
app.max_calls = 1;
app.thread_count = 1;
app.sip_port = 5060;
app.rtp_start_port = RTP_START_PORT;
app.local_addr = pj_str(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:d:p:r:i:l:q",
long_options, &option_index))!=-1)
{
switch (c) {
case 'c':
app.max_calls = atoi(pj_optarg);
if (app.max_calls < 0 || app.max_calls > MAX_CALLS) {
PJ_LOG(3,(THIS_FILE, "Invalid max calls value %s", pj_optarg));
return 1;
}
break;
case 'd':
app.duration = atoi(pj_optarg);
break;
case 'q':
app.auto_quit = 1;
break;
case 'p':
app.sip_port = atoi(pj_optarg);
break;
case 'r':
app.rtp_start_port = atoi(pj_optarg);
break;
case 'i':
app.local_addr = pj_str(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;
case OPT_REPORT_FILE:
app.report_filename = pj_optarg;
break;
default:
puts(USAGE);
return 1;
}
}
/* Check if URL is specified */
if (pj_optind < argc)
app.uri_to_call = pj_str(argv[pj_optind]);
/* Build local URI and contact */
pj_ansi_sprintf( local_uri, "sip:%s:%d", app.local_addr.ptr, app.sip_port);
app.local_uri = pj_str(local_uri);
app.local_contact = app.local_uri;
return PJ_SUCCESS;
}
/*****************************************************************************
* MEDIA STUFFS
*/
/*
* Create SDP session for a call.
*/
static pj_status_t create_sdp( pj_pool_t *pool,
struct call *call,
pjmedia_sdp_session **p_sdp)
{
pj_time_val tv;
pjmedia_sdp_session *sdp;
pjmedia_sdp_media *m;
pjmedia_sdp_attr *attr;
pjmedia_transport_udp_info tpinfo;
struct media_stream *audio = &call->media[0];
PJ_ASSERT_RETURN(pool && p_sdp, PJ_EINVAL);
/* Get transport info */
pjmedia_transport_udp_get_info(audio->transport, &tpinfo);
/* Create and initialize basic SDP session */
sdp = pj_pool_zalloc (pool, sizeof(pjmedia_sdp_session));
pj_gettimeofday(&tv);
sdp->origin.user = pj_str("pjsip-siprtp");
sdp->origin.version = sdp->origin.id = tv.sec + 2208988800UL;
sdp->origin.net_type = pj_str("IN");
sdp->origin.addr_type = pj_str("IP4");
sdp->origin.addr = *pj_gethostname();
sdp->name = pj_str("pjsip");
/* Since we only support one media stream at present, put the
* SDP connection line in the session level.
*/
sdp->conn = pj_pool_zalloc (pool, sizeof(pjmedia_sdp_conn));
sdp->conn->net_type = pj_str("IN");
sdp->conn->addr_type = pj_str("IP4");
sdp->conn->addr = app.local_addr;
/* SDP time and attributes. */
sdp->time.start = sdp->time.stop = 0;
sdp->attr_count = 0;
/* Create media stream 0: */
sdp->media_count = 1;
m = pj_pool_zalloc (pool, sizeof(pjmedia_sdp_media));
sdp->media[0] = m;
/* Standard media info: */
m->desc.media = pj_str("audio");
m->desc.port = pj_ntohs(tpinfo.skinfo.rtp_addr_name.sin_port);
m->desc.port_count = 1;
m->desc.transport = pj_str("RTP/AVP");
/* Add format and rtpmap for each codec. */
m->desc.fmt_count = 1;
m->attr_count = 0;
{
pjmedia_sdp_rtpmap rtpmap;
pjmedia_sdp_attr *attr;
char ptstr[10];
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);
m->attr[m->attr_count++] = attr;
}
/* Add sendrecv attribute. */
attr = pj_pool_zalloc(pool, sizeof(pjmedia_sdp_attr));
attr->name = pj_str("sendrecv");
m->attr[m->attr_count++] = attr;
#if 1
/*
* Add support telephony event
*/
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("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("121 0-15");
m->attr[m->attr_count++] = attr;
#endif
/* Done */
*p_sdp = sdp;
return PJ_SUCCESS;
}
#if defined(PJ_WIN32) && PJ_WIN32 != 0
#include <windows.h>
static void boost_priority(void)
{
SetPriorityClass( GetCurrentProcess(), REALTIME_PRIORITY_CLASS);
SetThreadPriority(GetCurrentThread(), THREAD_PRIORITY_HIGHEST);
}
#elif defined(PJ_LINUX) && PJ_LINUX != 0
#include <pthread.h>
static void boost_priority(void)
{
#define POLICY SCHED_FIFO
struct sched_param tp;
int max_prio;
int policy;
int rc;
if (sched_get_priority_min(POLICY) < sched_get_priority_max(POLICY))
max_prio = sched_get_priority_max(POLICY)-1;
else
max_prio = sched_get_priority_max(POLICY)+1;
/*
* Adjust process scheduling algorithm and priority
*/
rc = sched_getparam(0, &tp);
if (rc != 0) {
app_perror( THIS_FILE, "sched_getparam error",
PJ_RETURN_OS_ERROR(rc));
return;
}
tp.__sched_priority = max_prio;
rc = sched_setscheduler(0, POLICY, &tp);
if (rc != 0) {
app_perror( THIS_FILE, "sched_setscheduler error",
PJ_RETURN_OS_ERROR(rc));
}
PJ_LOG(4, (THIS_FILE, "New process policy=%d, priority=%d",
policy, tp.__sched_priority));
/*
* Adjust thread scheduling algorithm and priority
*/
rc = pthread_getschedparam(pthread_self(), &policy, &tp);
if (rc != 0) {
app_perror( THIS_FILE, "pthread_getschedparam error",
PJ_RETURN_OS_ERROR(rc));
return;
}
PJ_LOG(4, (THIS_FILE, "Old thread policy=%d, priority=%d",
policy, tp.__sched_priority));
policy = POLICY;
tp.__sched_priority = max_prio;
rc = pthread_setschedparam(pthread_self(), policy, &tp);
if (rc != 0) {
app_perror( THIS_FILE, "pthread_setschedparam error",
PJ_RETURN_OS_ERROR(rc));
return;
}
PJ_LOG(4, (THIS_FILE, "New thread policy=%d, priority=%d",
policy, tp.__sched_priority));
}
#else
# define boost_priority()
#endif
/*
* This callback is called by media transport on receipt of RTP packet.
*/
static void on_rx_rtp(void *user_data, const void *pkt, pj_ssize_t size)
{
struct media_stream *strm;
pj_status_t status;
const pjmedia_rtp_hdr *hdr;
const void *payload;
unsigned payload_len;
strm = user_data;
/* Discard packet if media is inactive */
if (!strm->active)
return;
/* Check for errors */
if (size < 0) {
app_perror(THIS_FILE, "RTP recv() error", -size);
return;
}
/* Decode RTP packet. */
status = pjmedia_rtp_decode_rtp(&strm->in_sess,
pkt, size,
&hdr, &payload, &payload_len);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "RTP decode error", status);
return;
}
//PJ_LOG(4,(THIS_FILE, "Rx seq=%d", pj_ntohs(hdr->seq)));
/* Update the RTCP session. */
pjmedia_rtcp_rx_rtp(&strm->rtcp, pj_ntohs(hdr->seq),
pj_ntohl(hdr->ts), payload_len);
/* Update RTP session */
pjmedia_rtp_session_update(&strm->in_sess, hdr, NULL);
}
/*
* This callback is called by media transport on receipt of RTCP packet.
*/
static void on_rx_rtcp(void *user_data, const void *pkt, pj_ssize_t size)
{
struct media_stream *strm;
strm = user_data;
/* Discard packet if media is inactive */
if (!strm->active)
return;
/* Check for errors */
if (size < 0) {
app_perror(THIS_FILE, "Error receiving RTCP packet", -size);
return;
}
/* Update RTCP session */
pjmedia_rtcp_rx_rtcp(&strm->rtcp, pkt, size);
}
/*
* Media thread
*
* This is the thread to send and receive both RTP and RTCP packets.
*/
static int media_thread(void *arg)
{
enum { RTCP_INTERVAL = 5000, RTCP_RAND = 2000 };
struct media_stream *strm = arg;
char packet[1500];
unsigned msec_interval;
pj_timestamp freq, next_rtp, next_rtcp;
/* Boost thread priority if necessary */
boost_priority();
/* Let things settle */
pj_thread_sleep(100);
msec_interval = strm->samples_per_frame * 1000 / strm->clock_rate;
pj_get_timestamp_freq(&freq);
pj_get_timestamp(&next_rtp);
next_rtp.u64 += (freq.u64 * msec_interval / 1000);
next_rtcp = next_rtp;
next_rtcp.u64 += (freq.u64 * (RTCP_INTERVAL+(pj_rand()%RTCP_RAND)) / 1000);
while (!strm->thread_quit_flag) {
pj_timestamp now, lesser;
pj_time_val timeout;
pj_bool_t send_rtp, send_rtcp;
send_rtp = send_rtcp = PJ_FALSE;
/* Determine how long to sleep */
if (next_rtp.u64 < next_rtcp.u64) {
lesser = next_rtp;
send_rtp = PJ_TRUE;
} else {
lesser = next_rtcp;
send_rtcp = PJ_TRUE;
}
pj_get_timestamp(&now);
if (lesser.u64 <= now.u64) {
timeout.sec = timeout.msec = 0;
//printf("immediate "); fflush(stdout);
} else {
pj_uint64_t tick_delay;
tick_delay = lesser.u64 - now.u64;
timeout.sec = 0;
timeout.msec = (pj_uint32_t)(tick_delay * 1000 / freq.u64);
pj_time_val_normalize(&timeout);
//printf("%d:%03d ", timeout.sec, timeout.msec); fflush(stdout);
}
/* Wait for next interval */
//if (timeout.sec!=0 && timeout.msec!=0) {
pj_thread_sleep(PJ_TIME_VAL_MSEC(timeout));
if (strm->thread_quit_flag)
break;
//}
pj_get_timestamp(&now);
if (send_rtp || next_rtp.u64 <= now.u64) {
/*
* Time to send RTP packet.
*/
pj_status_t status;
const void *p_hdr;
const pjmedia_rtp_hdr *hdr;
pj_ssize_t size;
int hdrlen;
/* Format RTP header */
status = pjmedia_rtp_encode_rtp( &strm->out_sess, strm->si.tx_pt,
0, /* marker bit */
strm->bytes_per_frame,
strm->samples_per_frame,
&p_hdr, &hdrlen);
if (status == PJ_SUCCESS) {
//PJ_LOG(4,(THIS_FILE, "\t\tTx seq=%d", pj_ntohs(hdr->seq)));
hdr = (const pjmedia_rtp_hdr*) p_hdr;
/* Copy RTP header to packet */
pj_memcpy(packet, hdr, hdrlen);
/* Zero the payload */
pj_bzero(packet+hdrlen, strm->bytes_per_frame);
/* Send RTP packet */
size = hdrlen + strm->bytes_per_frame;
status = pjmedia_transport_send_rtp(strm->transport,
packet, size);
if (status != PJ_SUCCESS)
app_perror(THIS_FILE, "Error sending RTP packet", status);
} else {
pj_assert(!"RTP encode() error");
}
/* Update RTCP SR */
pjmedia_rtcp_tx_rtp( &strm->rtcp, (pj_uint16_t)strm->bytes_per_frame);
/* Schedule next send */
next_rtp.u64 += (msec_interval * freq.u64 / 1000);
}
if (send_rtcp || next_rtcp.u64 <= now.u64) {
/*
* Time to send RTCP packet.
*/
pjmedia_rtcp_pkt *rtcp_pkt;
int rtcp_len;
pj_ssize_t size;
pj_status_t status;
/* Build RTCP packet */
pjmedia_rtcp_build_rtcp(&strm->rtcp, &rtcp_pkt, &rtcp_len);
/* Send packet */
size = rtcp_len;
status = pjmedia_transport_send_rtcp(strm->transport,
rtcp_pkt, size);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Error sending RTCP packet", status);
}
/* Schedule next send */
next_rtcp.u64 += (freq.u64 * (RTCP_INTERVAL+(pj_rand()%RTCP_RAND)) /
1000);
}
}
return 0;
}
/* Callback to be called when SDP negotiation is done in the call: */
static void call_on_media_update( pjsip_inv_session *inv,
pj_status_t status)
{
struct call *call;
pj_pool_t *pool;
struct media_stream *audio;
const 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;
audio = &call->media[0];
/* If this is a mid-call media update, then destroy existing media */
if (audio->thread != NULL)
destroy_call_media(call->index);
/* Do nothing if media negotiation has failed */
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "SDP negotiation failed", status);
return;
}
/* Capture stream definition from the SDP */
pjmedia_sdp_neg_get_active_local(inv->neg, &local_sdp);
pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp);
status = pjmedia_stream_info_from_sdp(&audio->si, inv->pool, app.med_endpt,
local_sdp, remote_sdp, 0);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Error creating stream info from SDP", status);
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.clock_rate;
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,
pj_rand());
pjmedia_rtp_session_init(&audio->in_sess, audio->si.fmt.pt, 0);
pjmedia_rtcp_init(&audio->rtcp, "rtcp", audio->clock_rate,
audio->samples_per_frame, 0);
/* Attach media to transport */
status = pjmedia_transport_attach(audio->transport, audio,
&audio->si.rem_addr,
&audio->si.rem_rtcp,
sizeof(pj_sockaddr_in),
&on_rx_rtp,
&on_rx_rtcp);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Error on pjmedia_transport_attach()", status);
return;
}
/* Start media thread. */
audio->thread_quit_flag = 0;
status = pj_thread_create( inv->pool, "media", &media_thread, audio,
0, 0, &audio->thread);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Error creating media thread", status);
return;
}
/* Set the media as active */
audio->active = PJ_TRUE;
}
/* Destroy call's media */
static void destroy_call_media(unsigned call_index)
{
struct media_stream *audio = &app.call[call_index].media[0];
if (audio) {
audio->active = PJ_FALSE;
if (audio->thread) {
audio->thread_quit_flag = 1;
pj_thread_join(audio->thread);
pj_thread_destroy(audio->thread);
audio->thread = NULL;
audio->thread_quit_flag = 0;
}
pjmedia_transport_detach(audio->transport, audio);
}
}
/*****************************************************************************
* USER INTERFACE STUFFS
*/
static void call_get_duration(int call_index, pj_time_val *dur)
{
struct call *call = &app.call[call_index];
pjsip_inv_session *inv;
dur->sec = dur->msec = 0;
if (!call)
return;
inv = call->inv;
if (!inv)
return;
if (inv->state >= PJSIP_INV_STATE_CONFIRMED && call->connect_time.sec) {
pj_gettimeofday(dur);
PJ_TIME_VAL_SUB((*dur), call->connect_time);
}
}
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.%02dK",
val / 1000,
(val % 1000) / 100);
} else {
pj_ansi_sprintf(buf, "%d.%02dM",
val / 1000000,
(val % 1000000) / 10000);
}
return buf;
}
static void print_avg_stat(void)
{
#define MIN_(var,val) if ((int)val < (int)var) var = val
#define MAX_(var,val) if ((int)val > (int)var) var = val
#define AVG_(var,val) var = ( ((var * count) + val) / (count+1) )
#define BIGVAL 0x7FFFFFFFL
struct stat_entry
{
int min, avg, max;
};
struct stat_entry call_dur, call_pdd;
pjmedia_rtcp_stat min_stat, avg_stat, max_stat;
char srx_min[16], srx_avg[16], srx_max[16];
char brx_min[16], brx_avg[16], brx_max[16];
char stx_min[16], stx_avg[16], stx_max[16];
char btx_min[16], btx_avg[16], btx_max[16];
unsigned i, count;
pj_bzero(&call_dur, sizeof(call_dur));
call_dur.min = BIGVAL;
pj_bzero(&call_pdd, sizeof(call_pdd));
call_pdd.min = BIGVAL;
pj_bzero(&min_stat, sizeof(min_stat));
min_stat.rx.pkt = min_stat.tx.pkt = BIGVAL;
min_stat.rx.bytes = min_stat.tx.bytes = BIGVAL;
min_stat.rx.loss = min_stat.tx.loss = BIGVAL;
min_stat.rx.dup = min_stat.tx.dup = BIGVAL;
min_stat.rx.reorder = min_stat.tx.reorder = BIGVAL;
min_stat.rx.jitter.min = min_stat.tx.jitter.min = BIGVAL;
min_stat.rtt.min = BIGVAL;
pj_bzero(&avg_stat, sizeof(avg_stat));
pj_bzero(&max_stat, sizeof(max_stat));
for (i=0, count=0; i<app.max_calls; ++i) {
struct call *call = &app.call[i];
struct media_stream *audio = &call->media[0];
pj_time_val dur;
unsigned msec_dur;
if (call->inv == NULL ||
call->inv->state < PJSIP_INV_STATE_CONFIRMED ||
call->connect_time.sec == 0)
{
continue;
}
/* Duration */
call_get_duration(i, &dur);
msec_dur = PJ_TIME_VAL_MSEC(dur);
MIN_(call_dur.min, msec_dur);
MAX_(call_dur.max, msec_dur);
AVG_(call_dur.avg, msec_dur);
/* Connect delay */
if (call->connect_time.sec) {
pj_time_val t = call->connect_time;
PJ_TIME_VAL_SUB(t, call->start_time);
msec_dur = PJ_TIME_VAL_MSEC(t);
} else {
msec_dur = 10;
}
MIN_(call_pdd.min, msec_dur);
MAX_(call_pdd.max, msec_dur);
AVG_(call_pdd.avg, msec_dur);
/* RX Statistisc: */
/* Packets */
MIN_(min_stat.rx.pkt, audio->rtcp.stat.rx.pkt);
MAX_(max_stat.rx.pkt, audio->rtcp.stat.rx.pkt);
AVG_(avg_stat.rx.pkt, audio->rtcp.stat.rx.pkt);
/* Bytes */
MIN_(min_stat.rx.bytes, audio->rtcp.stat.rx.bytes);
MAX_(max_stat.rx.bytes, audio->rtcp.stat.rx.bytes);
AVG_(avg_stat.rx.bytes, audio->rtcp.stat.rx.bytes);
/* Packet loss */
MIN_(min_stat.rx.loss, audio->rtcp.stat.rx.loss);
MAX_(max_stat.rx.loss, audio->rtcp.stat.rx.loss);
AVG_(avg_stat.rx.loss, audio->rtcp.stat.rx.loss);
/* Packet dup */
MIN_(min_stat.rx.dup, audio->rtcp.stat.rx.dup);
MAX_(max_stat.rx.dup, audio->rtcp.stat.rx.dup);
AVG_(avg_stat.rx.dup, audio->rtcp.stat.rx.dup);
/* Packet reorder */
MIN_(min_stat.rx.reorder, audio->rtcp.stat.rx.reorder);
MAX_(max_stat.rx.reorder, audio->rtcp.stat.rx.reorder);
AVG_(avg_stat.rx.reorder, audio->rtcp.stat.rx.reorder);
/* Jitter */
MIN_(min_stat.rx.jitter.min, audio->rtcp.stat.rx.jitter.min);
MAX_(max_stat.rx.jitter.max, audio->rtcp.stat.rx.jitter.max);
AVG_(avg_stat.rx.jitter.avg, audio->rtcp.stat.rx.jitter.avg);
/* TX Statistisc: */
/* Packets */
MIN_(min_stat.tx.pkt, audio->rtcp.stat.tx.pkt);
MAX_(max_stat.tx.pkt, audio->rtcp.stat.tx.pkt);
AVG_(avg_stat.tx.pkt, audio->rtcp.stat.tx.pkt);
/* Bytes */
MIN_(min_stat.tx.bytes, audio->rtcp.stat.tx.bytes);
MAX_(max_stat.tx.bytes, audio->rtcp.stat.tx.bytes);
AVG_(avg_stat.tx.bytes, audio->rtcp.stat.tx.bytes);
/* Packet loss */
MIN_(min_stat.tx.loss, audio->rtcp.stat.tx.loss);
MAX_(max_stat.tx.loss, audio->rtcp.stat.tx.loss);
AVG_(avg_stat.tx.loss, audio->rtcp.stat.tx.loss);
/* Packet dup */
MIN_(min_stat.tx.dup, audio->rtcp.stat.tx.dup);
MAX_(max_stat.tx.dup, audio->rtcp.stat.tx.dup);
AVG_(avg_stat.tx.dup, audio->rtcp.stat.tx.dup);
/* Packet reorder */
MIN_(min_stat.tx.reorder, audio->rtcp.stat.tx.reorder);
MAX_(max_stat.tx.reorder, audio->rtcp.stat.tx.reorder);
AVG_(avg_stat.tx.reorder, audio->rtcp.stat.tx.reorder);
/* Jitter */
MIN_(min_stat.tx.jitter.min, audio->rtcp.stat.tx.jitter.min);
MAX_(max_stat.tx.jitter.max, audio->rtcp.stat.tx.jitter.max);
AVG_(avg_stat.tx.jitter.avg, audio->rtcp.stat.tx.jitter.avg);
/* RTT */
MIN_(min_stat.rtt.min, audio->rtcp.stat.rtt.min);
MAX_(max_stat.rtt.max, audio->rtcp.stat.rtt.max);
AVG_(avg_stat.rtt.avg, audio->rtcp.stat.rtt.avg);
++count;
}
if (count == 0) {
puts("No active calls");
return;
}
printf("Total %d call(s) active.\n"
" Average Statistics\n"
" min avg max \n"
" -----------------------\n"
" call duration: %7d %7d %7d %s\n"
" connect delay: %7d %7d %7d %s\n"
" RX stat:\n"
" packets: %7s %7s %7s %s\n"
" payload: %7s %7s %7s %s\n"
" loss: %7d %7d %7d %s\n"
" percent loss: %7.3f %7.3f %7.3f %s\n"
" dup: %7d %7d %7d %s\n"
" reorder: %7d %7d %7d %s\n"
" jitter: %7.3f %7.3f %7.3f %s\n"
" TX stat:\n"
" packets: %7s %7s %7s %s\n"
" payload: %7s %7s %7s %s\n"
" loss: %7d %7d %7d %s\n"
" percent loss: %7.3f %7.3f %7.3f %s\n"
" dup: %7d %7d %7d %s\n"
" reorder: %7d %7d %7d %s\n"
" jitter: %7.3f %7.3f %7.3f %s\n"
" RTT : %7.3f %7.3f %7.3f %s\n"
,
count,
call_dur.min/1000, call_dur.avg/1000, call_dur.max/1000,
"seconds",
call_pdd.min, call_pdd.avg, call_pdd.max,
"ms",
/* rx */
good_number(srx_min, min_stat.rx.pkt),
good_number(srx_avg, avg_stat.rx.pkt),
good_number(srx_max, max_stat.rx.pkt),
"packets",
good_number(brx_min, min_stat.rx.bytes),
good_number(brx_avg, avg_stat.rx.bytes),
good_number(brx_max, max_stat.rx.bytes),
"bytes",
min_stat.rx.loss, avg_stat.rx.loss, max_stat.rx.loss,
"packets",
min_stat.rx.loss*100.0/(min_stat.rx.pkt+min_stat.rx.loss),
avg_stat.rx.loss*100.0/(avg_stat.rx.pkt+avg_stat.rx.loss),
max_stat.rx.loss*100.0/(max_stat.rx.pkt+max_stat.rx.loss),
"%",
min_stat.rx.dup, avg_stat.rx.dup, max_stat.rx.dup,
"packets",
min_stat.rx.reorder, avg_stat.rx.reorder, max_stat.rx.reorder,
"packets",
min_stat.rx.jitter.min/1000.0,
avg_stat.rx.jitter.avg/1000.0,
max_stat.rx.jitter.max/1000.0,
"ms",
/* tx */
good_number(stx_min, min_stat.tx.pkt),
good_number(stx_avg, avg_stat.tx.pkt),
good_number(stx_max, max_stat.tx.pkt),
"packets",
good_number(btx_min, min_stat.tx.bytes),
good_number(btx_avg, avg_stat.tx.bytes),
good_number(btx_max, max_stat.tx.bytes),
"bytes",
min_stat.tx.loss, avg_stat.tx.loss, max_stat.tx.loss,
"packets",
min_stat.tx.loss*100.0/(min_stat.tx.pkt+min_stat.tx.loss),
avg_stat.tx.loss*100.0/(avg_stat.tx.pkt+avg_stat.tx.loss),
max_stat.tx.loss*100.0/(max_stat.tx.pkt+max_stat.tx.loss),
"%",
min_stat.tx.dup, avg_stat.tx.dup, max_stat.tx.dup,
"packets",
min_stat.tx.reorder, avg_stat.tx.reorder, max_stat.tx.reorder,
"packets",
min_stat.tx.jitter.min/1000.0,
avg_stat.tx.jitter.avg/1000.0,
max_stat.tx.jitter.max/1000.0,
"ms",
/* rtt */
min_stat.rtt.min/1000.0,
avg_stat.rtt.avg/1000.0,
max_stat.rtt.max/1000.0,
"ms"
);
}
#include "siprtp_report.c"
static void list_calls()
{
unsigned i;
puts("List all calls:");
for (i=0; i<app.max_calls; ++i) {
if (!app.call[i].inv)
continue;
print_call(i);
}
}
static void hangup_call(unsigned index)
{
pjsip_tx_data *tdata;
pj_status_t status;
if (app.call[index].inv == NULL)
return;
status = pjsip_inv_end_session(app.call[index].inv, 603, NULL, &tdata);
if (status==PJ_SUCCESS && tdata!=NULL)
pjsip_inv_send_msg(app.call[index].inv, tdata);
}
static void hangup_all_calls()
{
unsigned i;
for (i=0; i<app.max_calls; ++i) {
if (!app.call[i].inv)
continue;
hangup_call(i);
}
/* Wait until all calls are terminated */
for (i=0; i<app.max_calls; ++i) {
while (app.call[i].inv)
pj_thread_sleep(10);
}
}
static pj_bool_t simple_input(const char *title, char *buf, pj_size_t len)
{
char *p;
printf("%s (empty to cancel): ", title); fflush(stdout);
fgets(buf, len, stdin);
/* Remove trailing newlines. */
for (p=buf; ; ++p) {
if (*p=='\r' || *p=='\n') *p='\0';
else if (!*p) break;
}
if (!*buf)
return PJ_FALSE;
return PJ_TRUE;
}
static const char *MENU =
"\n"
"Enter menu character:\n"
" s Summary\n"
" l List all calls\n"
" h Hangup a call\n"
" H Hangup all calls\n"
" q Quit\n"
"\n";
/* Main screen menu */
static void console_main()
{
char input1[10];
unsigned i;
printf("%s", MENU);
for (;;) {
printf(">>> "); fflush(stdout);
fgets(input1, sizeof(input1), stdin);
switch (input1[0]) {
case 's':
print_avg_stat();
break;
case 'l':
list_calls();
break;
case 'h':
if (!simple_input("Call number to hangup", input1, sizeof(input1)))
break;
i = atoi(input1);
hangup_call(i);
break;
case 'H':
hangup_all_calls();
break;
case 'q':
goto on_exit;
default:
puts("Invalid command");
printf("%s", MENU);
break;
}
fflush(stdout);
}
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 logger_on_rx_msg(pjsip_rx_data *rdata)
{
PJ_LOG(4,(THIS_FILE, "RX %d bytes %s from %s:%d:\n"
"%s\n"
"--end msg--",
rdata->msg_info.len,
pjsip_rx_data_get_info(rdata),
rdata->pkt_info.src_name,
rdata->pkt_info.src_port,
rdata->msg_info.msg_buf));
/* Always return false, otherwise messages will not get processed! */
return PJ_FALSE;
}
/* Notification on outgoing messages */
static pj_status_t logger_on_tx_msg(pjsip_tx_data *tdata)
{
/* Important note:
* tp_info field is only valid after outgoing messages has passed
* transport layer. So don't try to access tp_info when the module
* has lower priority than transport layer.
*/
PJ_LOG(4,(THIS_FILE, "TX %d bytes %s to %s:%d:\n"
"%s\n"
"--end msg--",
(tdata->buf.cur - tdata->buf.start),
pjsip_tx_data_get_info(tdata),
tdata->tp_info.dst_name,
tdata->tp_info.dst_port,
tdata->buf.start));
/* Always return success, otherwise message will not get sent! */
return PJ_SUCCESS;
}
/* The module instance. */
static pjsip_module msg_logger =
{
NULL, NULL, /* prev, next. */
{ "mod-siprtp-log", 14 }, /* Name. */
-1, /* Id */
PJSIP_MOD_PRIORITY_TRANSPORT_LAYER-1,/* Priority */
NULL, /* load() */
NULL, /* start() */
NULL, /* stop() */
NULL, /* unload() */
&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;
/* Verify options: */
/* Auto-quit can not be specified for UAS */
if (app.auto_quit && app.uri_to_call.slen == 0) {
printf("Error: --auto-quit option only valid for outgoing "
"mode (UAC) only\n");
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);
destroy_sip();
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);
destroy_sip();
return 1;
}
/* Start worker threads */
for (i=0; i<app.thread_count; ++i) {
pj_thread_create( app.pool, "app", &sip_worker_thread, NULL,
0, 0, &app.sip_thread[i]);
}
/* 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) {
app_perror(THIS_FILE, "Error making call", status);
break;
}
}
if (app.auto_quit) {
/* Wait for calls to complete */
while (app.uac_calls < app.max_calls)
pj_thread_sleep(100);
pj_thread_sleep(200);
} else {
/* Start user interface loop */
console_main();
}
} else {
PJ_LOG(3,(THIS_FILE, "Ready for incoming calls (max=%d)",
app.max_calls));
/* Start user interface loop */
console_main();
}
/* Shutting down... */
destroy_sip();
destroy_media();
if (app.pool) {
pj_pool_release(app.pool);
app.pool = NULL;
pj_caching_pool_destroy(&app.cp);
}
app_logging_shutdown();
/* Shutdown PJLIB */
pj_shutdown();
return 0;
}