siprtp.c

基于sip协议的网络电话源码

    /* 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;
}