예제 #1
0
파일: call.c 프로젝트: sealaunch/baresip
static void sipsub_notify_handler(struct sip *sip, const struct sip_msg *msg,
				  void *arg)
{
	struct call *call = arg;
	struct pl scode, reason;
	uint32_t sc;

	if (re_regex((char *)mbuf_buf(msg->mb), mbuf_get_left(msg->mb),
		     "SIP/2.0 [0-9]+ [^\r\n]+", &scode, &reason)) {
		(void)sip_reply(sip, msg, 400, "Bad sipfrag");
		return;
	}

	(void)sip_reply(sip, msg, 200, "OK");

	sc = pl_u32(&scode);

	if (sc >= 300) {
		warning("call: transfer failed: %u %r\n", sc, &reason);
		call_event_handler(call, CALL_EVENT_TRANSFER_FAILED,
				   "%u %r", sc, &reason);
	}
	else if (sc >= 200) {
		call_event_handler(call, CALL_EVENT_CLOSED, "Call transfered");
	}
}
예제 #2
0
파일: call.c 프로젝트: sealaunch/baresip
static void sipsess_progr_handler(const struct sip_msg *msg, void *arg)
{
	struct call *call = arg;
	bool media;

	MAGIC_CHECK(call);

	info("call: SIP Progress: %u %r (%r/%r)\n",
	     msg->scode, &msg->reason, &msg->ctyp.type, &msg->ctyp.subtype);

	if (msg->scode <= 100)
		return;

	/* check for 18x and content-type
	 *
	 * 1. start media-stream if application/sdp
	 * 2. play local ringback tone if not
	 *
	 * we must also handle changes to/from 180 and 183,
	 * so we reset the media-stream/ringback each time.
	 */
	if (msg_ctype_cmp(&msg->ctyp, "application", "sdp")
	    && mbuf_get_left(msg->mb)
	    && !sdp_decode(call->sdp, msg->mb, false)) {
		media = true;
	}
	else if (msg_ctype_cmp(&msg->ctyp, "multipart", "mixed") &&
		 !sdp_decode_multipart(&msg->ctyp.params, msg->mb) &&
		 !sdp_decode(call->sdp, msg->mb, false)) {
		media = true;
	}
	else
		media = false;

	switch (msg->scode) {

	case 180:
		set_state(call, STATE_RINGING);
		break;

	case 183:
		set_state(call, STATE_EARLY);
		break;
	}

	if (media)
		call_event_handler(call, CALL_EVENT_PROGRESS, call->peer_uri);
	else
		call_event_handler(call, CALL_EVENT_RINGING, call->peer_uri);

	call_stream_stop(call);

	if (media)
		call_stream_start(call, false);
}
예제 #3
0
파일: call.c 프로젝트: sealaunch/baresip
static void sipsess_refer_handler(struct sip *sip, const struct sip_msg *msg,
				  void *arg)
{
	struct call *call = arg;
	const struct sip_hdr *hdr;
	int err;

	/* get the transfer target */
	hdr = sip_msg_hdr(msg, SIP_HDR_REFER_TO);
	if (!hdr) {
		warning("call: bad REFER request from %r\n", &msg->from.auri);
		(void)sip_reply(sip, msg, 400, "Missing Refer-To header");
		return;
	}

	/* The REFER creates an implicit subscription.
	 * Reply 202 to the REFER request
	 */
	call->not = mem_deref(call->not);
	err = sipevent_accept(&call->not, uag_sipevent_sock(), msg,
			      sipsess_dialog(call->sess), NULL,
			      202, "Accepted", 60, 60, 60,
			      ua_cuser(call->ua), "message/sipfrag",
			      auth_handler, call->acc, true,
			      sipnot_close_handler, call,
			      "Allow: %s\r\n", uag_allowed_methods());
	if (err) {
		warning("call: refer: sipevent_accept failed: %m\n", err);
		return;
	}

	(void)call_notify_sipfrag(call, 100, "Trying");

	call_event_handler(call, CALL_EVENT_TRANSFER, "%r", &hdr->val);
}
예제 #4
0
BOOL
FooControl::handle_message(Rollout *ro, UINT message, WPARAM wParam, LPARAM lParam)
{
	if (message == CC_SPINNER_CHANGE)
	{
		one_value_local(arg);
		/* handle CC_SPINNER_CHANGE message received for this control - call my _T('changed') event handler with
		 * with the current spinner value as the argument */
		if (spin_type == EDITTYPE_INT)
		{
			value = (float)((ISpinnerControl *)lParam)->GetIVal();
			vl.arg = Integer::intern((int)value);
		}
		else
			vl.arg = Float::intern(value = ((ISpinnerControl *)lParam)->GetFVal()); 
		try
		{
			call_event_handler(ro, n_changed, &vl.arg, 1);
		}
		catch (...)
		{
   			SendMessage(GetDlgItem(ro->page, control_ID), WM_LBUTTONUP, 0, 0); // on error, force a buttonup to release the spinner
			throw;
		}
		pop_value_locals();
		return TRUE;
	}
	return FALSE;
}
예제 #5
0
파일: call.c 프로젝트: sealaunch/baresip
static void invite_timeout(void *arg)
{
	struct call *call = arg;

	info("%s: Local timeout after %u seconds\n",
	     call->peer_uri, LOCAL_TIMEOUT);

	call_event_handler(call, CALL_EVENT_CLOSED, "Local timeout");
}
예제 #6
0
파일: call.c 프로젝트: sealaunch/baresip
static void menc_error_handler(int err, void *arg)
{
	struct call *call = arg;
	MAGIC_CHECK(call);

	warning("call: mediaenc '%s' error: %m\n", call->acc->mencid, err);

	call_stream_stop(call);
	call_event_handler(call, CALL_EVENT_CLOSED, "mediaenc failed");
}
예제 #7
0
파일: call.c 프로젝트: sealaunch/baresip
static void video_error_handler(int err, const char *str, void *arg)
{
	struct call *call = arg;
	MAGIC_CHECK(call);

	warning("call: video device error: %m (%s)\n", err, str);

	call_stream_stop(call);
	call_event_handler(call, CALL_EVENT_CLOSED, str);
}
예제 #8
0
파일: call.c 프로젝트: sealaunch/baresip
/** Called when all media streams are established */
static void mnat_handler(int err, uint16_t scode, const char *reason,
			 void *arg)
{
	struct call *call = arg;
	MAGIC_CHECK(call);

	if (err) {
		warning("call: medianat '%s' failed: %m\n",
			call->acc->mnatid, err);
		call_event_handler(call, CALL_EVENT_CLOSED, "%m", err);
		return;
	}
	else if (scode) {
		warning("call: medianat failed: %u %s\n", scode, reason);
		call_event_handler(call, CALL_EVENT_CLOSED, "%u %s",
				   scode, reason);
		return;
	}

	/* Re-INVITE */
	if (!call->mnat_wait) {
		info("call: medianat established -- sending Re-INVITE\n");
		(void)call_modify(call);
		return;
	}

	call->mnat_wait = false;

	switch (call->state) {

	case STATE_OUTGOING:
		(void)send_invite(call);
		break;

	case STATE_INCOMING:
		call_event_handler(call, CALL_EVENT_INCOMING, call->peer_uri);
		break;

	default:
		break;
	}
}
예제 #9
0
파일: call.c 프로젝트: sealaunch/baresip
static void sipsub_close_handler(int err, const struct sip_msg *msg,
				 const struct sipevent_substate *substate,
				 void *arg)
{
	struct call *call = arg;

	(void)substate;

	call->sub = mem_deref(call->sub);

	if (err) {
		info("call: subscription closed: %m\n", err);
	}
	else if (msg && msg->scode >= 300) {
		info("call: transfer failed: %u %r\n",
		     msg->scode, &msg->reason);
		call_event_handler(call, CALL_EVENT_TRANSFER_FAILED,
				   "%u %r", msg->scode, &msg->reason);
	}
}
예제 #10
0
파일: call.c 프로젝트: sealaunch/baresip
static void sipsess_estab_handler(const struct sip_msg *msg, void *arg)
{
	struct call *call = arg;

	MAGIC_CHECK(call);

	(void)msg;

	if (call->state == STATE_ESTABLISHED)
		return;

	set_state(call, STATE_ESTABLISHED);

	call_event_handler(call, CALL_EVENT_ESTABLISHED, call->peer_uri);

	call_stream_start(call, true);

	/* the transferor will hangup this call */
	if (call->not) {
		(void)call_notify_sipfrag(call, 200, "OK");
	}
}
예제 #11
0
파일: call.c 프로젝트: sealaunch/baresip
static void sipsess_close_handler(int err, const struct sip_msg *msg,
				  void *arg)
{
	struct call *call = arg;
	char reason[128] = "";

	MAGIC_CHECK(call);

	if (err) {
		info("%s: session closed: %m\n", call->peer_uri, err);

		if (call->not) {
			(void)call_notify_sipfrag(call, 500, "%m", err);
		}
	}
	else if (msg) {

		call->scode = msg->scode;

		(void)re_snprintf(reason, sizeof(reason), "%u %r",
				  msg->scode, &msg->reason);

		info("%s: session closed: %u %r\n",
		     call->peer_uri, msg->scode, &msg->reason);

		if (call->not) {
			(void)call_notify_sipfrag(call, msg->scode,
						  "%r", &msg->reason);
		}
	}
	else {
		info("%s: session closed\n", call->peer_uri);
	}

	call_stream_stop(call);
	call_event_handler(call, CALL_EVENT_CLOSED, reason);
}
예제 #12
0
파일: call.c 프로젝트: sealaunch/baresip
int call_accept(struct call *call, struct sipsess_sock *sess_sock,
		const struct sip_msg *msg)
{
	bool got_offer;
	int err;

	if (!call || !msg)
		return EINVAL;

	call->outgoing = false;

	got_offer = (mbuf_get_left(msg->mb) > 0);

	err = pl_strdup(&call->peer_uri, &msg->from.auri);
	if (err)
		return err;

	if (pl_isset(&msg->from.dname)) {
		err = pl_strdup(&call->peer_name, &msg->from.dname);
		if (err)
			return err;
	}

	if (got_offer) {
		struct sdp_media *m;
		const struct sa *raddr;

		err = sdp_decode(call->sdp, msg->mb, true);
		if (err)
			return err;

		call->got_offer = true;

		/*
		 * Each media description in the SDP answer MUST
		 * use the same network type as the corresponding
		 * media description in the offer.
		 *
		 * See RFC 6157
		 */
		m = stream_sdpmedia(audio_strm(call->audio));
		raddr = sdp_media_raddr(m);

		if (sa_af(raddr) != call->af) {
			info("call: incompatible address-family"
			     " (local=%s, remote=%s)\n",
			     net_af2name(call->af),
			     net_af2name(sa_af(raddr)));

			sip_treply(NULL, uag_sip(), msg,
				   488, "Not Acceptable Here");

			call_event_handler(call, CALL_EVENT_CLOSED,
					   "Wrong address family");
			return 0;
		}

		/* Check if we have any common audio codecs, after
		 * the SDP offer has been parsed
		 */
		if (!have_common_audio_codecs(call)) {
			info("call: no common audio codecs - rejected\n");

			sip_treply(NULL, uag_sip(), msg,
				   488, "Not Acceptable Here");

			call_event_handler(call, CALL_EVENT_CLOSED,
					   "No audio codecs");

			return 0;
		}
	}

	err = sipsess_accept(&call->sess, sess_sock, msg, 180, "Ringing",
			     ua_cuser(call->ua), "application/sdp", NULL,
			     auth_handler, call->acc, true,
			     sipsess_offer_handler, sipsess_answer_handler,
			     sipsess_estab_handler, sipsess_info_handler,
			     sipsess_refer_handler, sipsess_close_handler,
			     call, "Allow: %s\r\n", uag_allowed_methods());
	if (err) {
		warning("call: sipsess_accept: %m\n", err);
		return err;
	}

	set_state(call, STATE_INCOMING);

	/* New call */
	tmr_start(&call->tmr_inv, LOCAL_TIMEOUT*1000, invite_timeout, call);

	if (!call->acc->mnat)
		call_event_handler(call, CALL_EVENT_INCOMING, call->peer_uri);

	return err;
}
예제 #13
0
파일: call.c 프로젝트: aKlausKranz/baresip
int call_accept(struct call *call, struct sipsess_sock *sess_sock,
		const struct sip_msg *msg)
{
	bool got_offer;
	int err;

	if (!call || !msg)
		return EINVAL;

	got_offer = (mbuf_get_left(msg->mb) > 0);

	err = pl_strdup(&call->peer_uri, &msg->from.auri);
	if (err)
		return err;

	if (pl_isset(&msg->from.dname)) {
		err = pl_strdup(&call->peer_name, &msg->from.dname);
		if (err)
			return err;
	}

	if (got_offer) {

		err = sdp_decode(call->sdp, msg->mb, true);
		if (err)
			return err;

		call->got_offer = true;

		/* Check if we have any common audio codecs, after
		 * the SDP offer has been parsed
		 */
		if (!have_common_audio_codecs(call)) {
			info("call: no common audio codecs - rejected\n");

			sip_treply(NULL, uag_sip(), msg,
				   488, "Not Acceptable Here");

			call_event_handler(call, CALL_EVENT_CLOSED,
					   "No audio codecs");

			return 0;
		}
	}

	err = sipsess_accept(&call->sess, sess_sock, msg, 180, "Ringing",
			     ua_cuser(call->ua), "application/sdp", NULL,
			     auth_handler, call->acc, true,
			     sipsess_offer_handler, sipsess_answer_handler,
			     sipsess_estab_handler, sipsess_info_handler,
			     sipsess_refer_handler, sipsess_close_handler,
			     call, "Allow: %s\r\n", uag_allowed_methods());
	if (err) {
		warning("call: sipsess_accept: %m\n", err);
		return err;
	}

	set_state(call, STATE_INCOMING);

	/* New call */
	tmr_start(&call->tmr_inv, LOCAL_TIMEOUT*1000, invite_timeout, call);

	if (!call->acc->mnat)
		call_event_handler(call, CALL_EVENT_INCOMING, call->peer_uri);

	return err;
}