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review.jami.net / pjproject / 9a0eab59c89f4241916a7aab3ce8714f690b8e1f / . / 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
*/
/* 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 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
{
/* Static: */
pj_uint16_t port; /* RTP port (RTCP is +1) */
/* 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. */
/* Sockets: */
pj_sock_t rtp_sock; /* RTP socket. */
pj_sock_t rtcp_sock; /* RTCP socket. */
/* 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. */
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 */
};
struct call
{
unsigned index;
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;
};
static struct app
{
unsigned max_calls;
unsigned thread_count;
int sip_port;
int rtp_start_port;
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;
pj_caching_pool cp;
pj_pool_t *pool;
pjsip_endpoint *sip_endpt;
pj_bool_t thread_quit;
pj_thread_t *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 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);
/* Destroy the call's media */
static void destroy_call_media(unsigned call_index);
/* Display error */
static void app_perror(const char *sender, const char *title,
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
* 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()
{
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 global endpoint: */
{
const pj_str_t *hostname;
const char *endpt_name;
/* Endpoint MUST be assigned a globally unique name.
* The name will be used as the hostname in Warning header.
*/
/* For this implementation, we'll use hostname for simplicity */
hostname = pj_gethostname();
endpt_name = hostname->ptr;
/* Create the endpoint: */
status = pjsip_endpt_create(&app.cp.factory, endpt_name,
&app.sip_endpt);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
}
/* Add UDP transport. */
{
pj_sockaddr_in addr;
addr.sin_family = PJ_AF_INET;
addr.sin_addr.s_addr = 0;
addr.sin_port = pj_htons((pj_uint16_t)app.sip_port);
status = pjsip_udp_transport_start( app.sip_endpt, &addr, NULL,
1, NULL);
if (status != PJ_SUCCESS)
return status;
}
/*
* 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_memset(&inv_cb, 0, 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);
/* 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.thread[i]) {
pj_thread_join(app.thread[i]);
pj_thread_destroy(app.thread[i]);
app.thread[i] = NULL;
}
}
if (app.sip_endpt) {
pjsip_endpt_destroy(app.sip_endpt);
app.sip_endpt = NULL;
}
if (app.pool) {
pj_pool_release(app.pool);
app.pool = NULL;
pj_caching_pool_destroy(&app.cp);
}
}
/*
* Init media stack.
*/
static pj_status_t init_media()
{
pj_ioqueue_t *ioqueue;
unsigned i, count;
pj_uint16_t rtp_port;
pj_str_t temp;
pj_sockaddr_in addr;
pj_status_t status;
/* Get the ioqueue from the SIP endpoint */
ioqueue = pjsip_endpt_get_ioqueue(app.sip_endpt);
/* Initialize media endpoint so that at least error subsystem is properly
* initialized.
*/
status = pjmedia_endpt_create(&app.cp.factory, ioqueue, 1,
&app.med_endpt);
PJ_ASSERT_RETURN(status == PJ_SUCCESS, status);
/* Determine address to bind socket */
pj_memset(&addr, 0, sizeof(addr));
addr.sin_family = PJ_AF_INET;
i = pj_inet_aton(pj_cstr(&temp, app.local_addr), &addr.sin_addr);
if (i == 0) {
PJ_LOG(3,(THIS_FILE,
"Error: invalid local address %s (expecting IP)",
app.local_addr));
return -1;
}
/* RTP port counter */
rtp_port = (pj_uint16_t)(app.rtp_start_port & 0xFFFE);
/* Init media sockets. */
for (i=0, count=0; i<app.max_calls; ++i, ++count) {
int retry;
app.call[i].index = i;
/* Repeat binding media socket to next port when fails to bind
* to current port number.
*/
retry = 0;
do {
struct media_stream *m = &app.call[i].media[0];
++retry;
rtp_port += 2;
m->port = rtp_port;
/* Create and bind RTP socket */
status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0,
&m->rtp_sock);
if (status != PJ_SUCCESS)
goto on_error;
addr.sin_port = pj_htons(rtp_port);
status = pj_sock_bind(m->rtp_sock, &addr, sizeof(addr));
if (status != PJ_SUCCESS) {
pj_sock_close(m->rtp_sock), m->rtp_sock=0;
continue;
}
/* Create and bind RTCP socket */
status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0,
&m->rtcp_sock);
if (status != PJ_SUCCESS)
goto on_error;
addr.sin_port = pj_htons((pj_uint16_t)(rtp_port+1));
status = pj_sock_bind(m->rtcp_sock, &addr, sizeof(addr));
if (status != PJ_SUCCESS) {
pj_sock_close(m->rtp_sock), m->rtp_sock=0;
pj_sock_close(m->rtcp_sock), m->rtcp_sock=0;
continue;
}
} while (status != PJ_SUCCESS && retry < 100);
if (status != PJ_SUCCESS)
goto on_error;
}
/* Done */
return PJ_SUCCESS;
on_error:
for (i=0; i<count; ++i) {
struct media_stream *m = &app.call[i].media[0];
pj_sock_close(m->rtp_sock), m->rtp_sock=0;
pj_sock_close(m->rtcp_sock), m->rtcp_sock=0;
}
return status;
}
/*
* Destroy media.
*/
static void destroy_media()
{
unsigned i;
for (i=0; i<app.max_calls; ++i) {
struct media_stream *m = &app.call[i].media[0];
if (m->rtp_sock)
pj_sock_close(m->rtp_sock), m->rtp_sock = 0;
if (m->rtcp_sock)
pj_sock_close(m->rtcp_sock), m->rtcp_sock = 0;
}
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)
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);
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;
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];
/* 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 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};
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);
}
}
/* 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 */
static int 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;
}
/* Usage */
static const char *USAGE =
"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"
"\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"
;
/* 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 };
struct pj_getopt_option long_options[] = {
{ "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;
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 = 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:l:",
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 'p':
app.sip_port = atoi(pj_optarg);
break;
case 'r':
app.rtp_start_port = atoi(pj_optarg);
break;
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;
}
}
/* 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, 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;
struct media_stream *audio = &call->media[0];
PJ_ASSERT_RETURN(pool && p_sdp, PJ_EINVAL);
/* 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 = pj_str(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 = audio->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;
}
/*
* Media thread
*
* This is the thread to send and receive both RTP and RTCP packets.
*/
static int media_thread(void *arg)
{
enum { RTCP_INTERVAL = 5 };
struct media_stream *strm = arg;
char packet[1500];
unsigned msec_interval;
pj_timestamp freq, next_rtp, next_rtcp;
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);
while (!strm->thread_quit_flag) {
pj_fd_set_t set;
pj_timestamp now, lesser;
pj_time_val timeout;
int rc;
/* Determine how long to sleep */
if (next_rtp.u64 < next_rtcp.u64)
lesser = next_rtp;
else
lesser = next_rtcp;
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);
}
PJ_FD_ZERO(&set);
PJ_FD_SET(strm->rtp_sock, &set);
PJ_FD_SET(strm->rtcp_sock, &set);
rc = pj_sock_select(FD_SETSIZE, &set, NULL, NULL, &timeout);
if (rc > 0 && PJ_FD_ISSET(strm->rtp_sock, &set)) {
/*
* Process incoming RTP packet.
*/
pj_status_t status;
pj_ssize_t size;
const pjmedia_rtp_hdr *hdr;
const void *payload;
unsigned payload_len;
size = sizeof(packet);
status = pj_sock_recv(strm->rtp_sock, packet, &size, 0);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "RTP recv() error", status);
continue;
}
++strm->rx_stat.pkt;
strm->rx_stat.payload += (size - 12);
/* Decode RTP packet. */
status = pjmedia_rtp_decode_rtp(&strm->in_sess,
packet, size,
&hdr,
&payload, &payload_len);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "RTP decode error", status);
strm->rx_stat.discard++;
continue;
}
/* Update RTP session */
status = pjmedia_rtp_session_update(&strm->in_sess, hdr);
if (status != PJ_SUCCESS &&
status != PJMEDIA_RTP_ESESSPROBATION &&
status != PJMEDIA_RTP_ESESSRESTART)
{
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;
}
/* Update the RTCP session. */
pjmedia_rtcp_rx_rtp(&strm->rtcp, pj_ntohs(hdr->seq),
pj_ntohl(hdr->ts));
}
if (rc > 0 && PJ_FD_ISSET(strm->rtcp_sock, &set)) {
/*
* Process incoming RTCP
*/
pj_status_t status;
pj_ssize_t size;
size = sizeof(packet);
status = pj_sock_recv( strm->rtcp_sock, packet, &size, 0);
if (status != PJ_SUCCESS)
app_perror(THIS_FILE, "Error receiving RTCP packet", status);
else {
if (size > sizeof(strm->rem_rtcp)) {
PJ_LOG(3,(THIS_FILE, "Error: RTCP packet too large"));
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++;
}
}
pj_get_timestamp(&now);
if (next_rtp.u64 <= now.u64) {
/*
* Time to send RTP packet.
*/
pj_status_t status;
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,
&hdr, &hdrlen);
if (status == PJ_SUCCESS) {
/* Copy RTP header to packet */
pj_memcpy(packet, hdr, hdrlen);
/* Zero the payload */
pj_memset(packet+hdrlen, 0, strm->bytes_per_frame);
/* Send RTP packet */
size = hdrlen + strm->bytes_per_frame;
status = pj_sock_sendto( strm->rtp_sock, packet, &size, 0,
&strm->si.rem_addr,
sizeof(strm->si.rem_addr));
if (status != PJ_SUCCESS)
app_perror(THIS_FILE, "Error sending RTP packet", status);
}
/* 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);
/* Update stats */
strm->tx_stat.pkt++;
strm->tx_stat.payload += strm->bytes_per_frame;
}
if (next_rtcp.u64 <= now.u64) {
/*
* Time to send RTCP packet.
*/
pjmedia_rtcp_pkt *rtcp_pkt;
int rtcp_len;
pj_sockaddr_in rem_addr;
pj_ssize_t size;
int port;
pj_status_t status;
/* Build RTCP packet */
pjmedia_rtcp_build_rtcp(&strm->rtcp, &rtcp_pkt, &rtcp_len);
/* Calculate address based on RTP address */
rem_addr = strm->si.rem_addr;
port = pj_ntohs(strm->si.rem_addr.sin_port) + 1;
rem_addr.sin_port = pj_htons((pj_uint16_t)port);
/* Send packet */
size = rtcp_len;
status = pj_sock_sendto(strm->rtcp_sock, rtcp_pkt, &size, 0,
&rem_addr, sizeof(rem_addr));
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Error sending RTCP packet", status);
}
/* 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.u64 += (freq.u64 * RTCP_INTERVAL);
}
}
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;
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.sample_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, audio->clock_rate, 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,
0, 0, &audio->thread);
if (status != PJ_SUCCESS) {
app_perror(THIS_FILE, "Error creating media thread", status);
}
}
/* Destroy call's media */
static void destroy_call_media(unsigned call_index)
{
struct media_stream *audio = &app.call[call_index].media[0];
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;
/* 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
*/
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];
char bps[16], ipbps[16], packets[16], bytes[16], ipbytes[16];
pj_uint32_t total_loss;
/* 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(" %s\n", userinfo);
/* 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, %dms/frame, %sbps (%sbps +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));
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),
""
);
}
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);
}
}
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"
" 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 '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;
/* 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", &worker_thread, NULL,
0, 0, &app.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;
}
}
} else {
PJ_LOG(3,(THIS_FILE, "Ready for incoming calls (max=%d)",
app.max_calls));
}
/* Start user interface loop */
console_main();
/* Shutting down... */
destroy_media();
destroy_sip();
app_logging_shutdown();
return 0;
}