simpleua.c

基于sip协议的网络电话源码

	 * Note that Route URI SHOULD have an ";lr" parameter!
	 */


	/* Get the SDP body to be put in the outgoing INVITE, by asking
	 * media endpoint to create one for us. The SDP will contain all
	 * codecs that have been registered to it (in this case, only
	 * PCMA and PCMU), plus telephony event.
	 */
	status = pjmedia_endpt_create_sdp( g_med_endpt,	    /* the media endpt	*/
					   dlg->pool,	    /* pool.		*/
					   1,		    /* # of streams	*/
					   &g_med_skinfo,   /* RTP sock info	*/
					   &local_sdp);	    /* the SDP result	*/
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);



	/* Create the INVITE session, and pass the SDP returned earlier
	 * as the session's initial capability.
	 */
	status = pjsip_inv_create_uac( dlg, local_sdp, 0, &g_inv);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);


	/* Create initial INVITE request.
	 * This INVITE request will contain a perfectly good request and 
	 * an SDP body as well.
	 */
	status = pjsip_inv_invite(g_inv, &tdata);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);



	/* 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(g_inv, tdata);
	PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);


    } else {

	/* No URL to make call to */

	PJ_LOG(3,(THIS_FILE, "Ready to accept incoming calls..."));
    }


    /* Loop until one call is completed */
    for (;!g_complete;) {
	pj_time_val timeout = {0, 10};
	pjsip_endpt_handle_events(g_endpt, &timeout);
    }

    /* On exit, dump current memory usage: */
    dump_pool_usage(THIS_FILE, &cp);

    return 0;
}



/*
 * Callback when INVITE session state has changed.
 * This callback is registered when the invite session module is initialized.
 * We mostly want to know when the invite session has been disconnected,
 * so that we can quit the application.
 */
static void call_on_state_changed( pjsip_inv_session *inv, 
				   pjsip_event *e)
{
    PJ_UNUSED_ARG(e);

    if (inv->state == PJSIP_INV_STATE_DISCONNECTED) {

	PJ_LOG(3,(THIS_FILE, "Call DISCONNECTED [reason=%d (%s)]", 
		  inv->cause,
		  pjsip_get_status_text(inv->cause)->ptr));

	PJ_LOG(3,(THIS_FILE, "One call completed, application quitting..."));
	g_complete = 1;

    } else {

	PJ_LOG(3,(THIS_FILE, "Call state changed to %s", 
		  pjsip_inv_state_name(inv->state)));

    }
}


/* This callback is called when dialog has forked. */
static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e)
{
    /* To be done... */
    PJ_UNUSED_ARG(inv);
    PJ_UNUSED_ARG(e);
}


/*
 * Callback when incoming requests outside any transactions and any
 * dialogs are received. We're only interested to hande incoming INVITE
 * request, and we'll reject any other requests with 500 response.
 */
static pj_bool_t on_rx_request( pjsip_rx_data *rdata )
{
    pjsip_dialog *dlg;
    pjmedia_sdp_session *local_sdp;
    pjsip_tx_data *tdata;
    unsigned options = 0;
    pj_status_t status;


    /* 
     * Respond (statelessly) any non-INVITE requests with 500 
     */
    if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) {

	if (rdata->msg_info.msg->line.req.method.id != PJSIP_ACK_METHOD) {
	    pj_str_t reason = pj_str("Simple UA unable to handle "
				     "this request");

	    pjsip_endpt_respond_stateless( g_endpt, rdata, 
					   500, &reason,
					   NULL, NULL);
	}
	return PJ_TRUE;
    }


    /*
     * Reject INVITE if we already have an INVITE session in progress.
     */
    if (g_inv) {

	pj_str_t reason = pj_str("Another call is in progress");

	pjsip_endpt_respond_stateless( g_endpt, rdata, 
				       500, &reason,
				       NULL, NULL);
	return PJ_TRUE;

    }

    /* Verify that we can handle the request. */
    status = pjsip_inv_verify_request(rdata, &options, NULL, NULL,
				      g_endpt, NULL);
    if (status != PJ_SUCCESS) {

	pj_str_t reason = pj_str("Sorry Simple UA can not handle this INVITE");

	pjsip_endpt_respond_stateless( g_endpt, rdata, 
				       500, &reason,
				       NULL, NULL);
	return PJ_TRUE;
    } 

    /*
     * Create UAS dialog.
     */
    status = pjsip_dlg_create_uas( pjsip_ua_instance(), 
				   rdata,
				   NULL, /* contact */
				   &dlg);
    if (status != PJ_SUCCESS) {
	pjsip_endpt_respond_stateless(g_endpt, rdata, 500, NULL,
				      NULL, NULL);
	return PJ_TRUE;
    }

    /* 
     * Get media capability from media endpoint: 
     */

    status = pjmedia_endpt_create_sdp( g_med_endpt, rdata->tp_info.pool, 1,
				       &g_med_skinfo, 
				       &local_sdp);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);


    /* 
     * Create invite session, and pass both the UAS dialog and the SDP
     * capability to the session.
     */
    status = pjsip_inv_create_uas( dlg, rdata, local_sdp, 0, &g_inv);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);


    /*
     * Initially send 180 response.
     *
     * The very first response to an INVITE must be created with
     * pjsip_inv_initial_answer(). Subsequent responses to the same
     * transaction MUST use pjsip_inv_answer().
     */
    status = pjsip_inv_initial_answer(g_inv, rdata, 
				      180, 
				      NULL, NULL, &tdata);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);


    /* Send the 180 response. */  
    status = pjsip_inv_send_msg(g_inv, tdata); 
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);


    /*
     * Now create 200 response.
     */
    status = pjsip_inv_answer( g_inv, 
			       200, NULL,	/* st_code and st_text */
			       NULL,		/* SDP already specified */
			       &tdata);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);

    /*
     * Send the 200 response.
     */
    status = pjsip_inv_send_msg(g_inv, tdata);
    PJ_ASSERT_RETURN(status == PJ_SUCCESS, PJ_TRUE);


    /* Done. 
     * When the call is disconnected, it will be reported via the callback.
     */

    return PJ_TRUE;
}

 

/*
 * Callback when SDP negotiation has completed.
 * We are interested with this callback because we want to start media
 * as soon as SDP negotiation is completed.
 */
static void call_on_media_update( pjsip_inv_session *inv,
				  pj_status_t status)
{
    pjmedia_session_info sess_info;
    const pjmedia_sdp_session *local_sdp;
    const pjmedia_sdp_session *remote_sdp;
    pjmedia_port *media_port;

    if (status != PJ_SUCCESS) {

	app_perror(THIS_FILE, "SDP negotiation has failed", status);

	/* Here we should disconnect call if we're not in the middle 
	 * of initializing an UAS dialog and if this is not a re-INVITE.
	 */
	return;
    }

    /* Get local and remote SDP.
     * We need both SDPs to create a media session.
     */
    status = pjmedia_sdp_neg_get_active_local(inv->neg, &local_sdp);

    status = pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp);


    /* Create session info based on the two SDPs. 
     * We only support one stream per session for now.
     */
    status = pjmedia_session_info_from_sdp(inv->dlg->pool, g_med_endpt, 
					   1, &sess_info, 
					   local_sdp, remote_sdp);
    if (status != PJ_SUCCESS) {
	app_perror( THIS_FILE, "Unable to create media session", status);
	return;
    }

    /* If required, we can also change some settings in the session info,
     * (such as jitter buffer settings, codec settings, etc) before we
     * create the session.
     */

    /* Create new media session, passing the two SDPs, and also the
     * media socket that we created earlier.
     * The media session is active immediately.
     */
    status = pjmedia_session_create( g_med_endpt, &sess_info,
				     &g_med_transport, NULL, &g_med_session );
    if (status != PJ_SUCCESS) {
	app_perror( THIS_FILE, "Unable to create media session", status);
	return;
    }


    /* Get the media port interface of the first stream in the session. 
     * Media port interface is basicly a struct containing get_frame() and
     * put_frame() function. With this media port interface, we can attach
     * the port interface to conference bridge, or directly to a sound
     * player/recorder device.
     */
    pjmedia_session_get_port(g_med_session, 0, &media_port);



    /* Create a sound Player device and connect the media port to the
     * sound device.
     */
    status = pjmedia_snd_port_create_player( 
		    inv->pool,				/* pool		    */
		    -1,					/* sound dev id	    */
		    media_port->info.clock_rate,	/* clock rate	    */
		    media_port->info.channel_count,	/* channel count    */
		    media_port->info.samples_per_frame, /* samples per frame*/
		    media_port->info.bits_per_sample,   /* bits per sample  */
		    0,					/* options	    */
		    &g_snd_player);
    if (status != PJ_SUCCESS) {
	app_perror( THIS_FILE, "Unable to create sound player", status);
	PJ_LOG(3,(THIS_FILE, "%d %d %d %d",
	    	    media_port->info.clock_rate,	/* clock rate	    */
		    media_port->info.channel_count,	/* channel count    */
		    media_port->info.samples_per_frame, /* samples per frame*/
		    media_port->info.bits_per_sample    /* bits per sample  */
	    ));
	return;
    }

    status = pjmedia_snd_port_connect(g_snd_player, media_port);


    /* Create a sound recorder device and connect the media port to the
     * sound device.
     */
    status = pjmedia_snd_port_create_rec( 
		    inv->pool,				/* pool		    */
		    -1,					/* sound dev id	    */
		    media_port->info.clock_rate,	/* clock rate	    */
		    media_port->info.channel_count,	/* channel count    */
		    media_port->info.samples_per_frame, /* samples per frame*/
		    media_port->info.bits_per_sample,   /* bits per sample  */
		    0,					/* options	    */
		    &g_snd_rec);
    if (status != PJ_SUCCESS) {
	app_perror( THIS_FILE, "Unable to create sound recorder", status);
	return;
    }

    status = pjmedia_snd_port_connect(g_snd_rec, media_port);

    /* Done with media. */
}