PeerStreamer-ng

											var offerlessInvite = false;

											} else {

												Janus.log("This call doesn't contain an offer... we'll need to provide one ourselves");

												offerlessInvite = true;

												// In case you want to offer video when reacting to an offerless call, set this to true

												doVideo = false;

											var extra = "";

											if(offerlessInvite)

												extra = " (no SDP offer provided)";

												message: "Incoming call from " + result["displayname"] + " (" + result["username"] + ")!" + rtpType + extra,

															// Notice that we can only answer if we got an offer: if this was

															// an offerless call, we'll need to create an offer ourselves

															var sipcallAction = (offerlessInvite ? sipcall.createOffer : sipcall.createAnswer);

															sipcallAction(

																		Janus.debug("Got SDP " + jsep.type + "! audio=" + doAudio + ", video=" + doVideo);

											Janus.log(jsep);

											// Call can start, now: handle the remote answer

		if((video && session->media.has_video && session->media.video_rtp_fd != -1) ||

				(!video && session->media.has_audio && session->media.audio_rtp_fd != -1)) {

		if((video && session->media.has_video && session->media.video_rtcp_fd != -1) ||

				(!video && session->media.has_audio && session->media.audio_rtcp_fd != -1)) {

			gboolean answer = !strcasecmp(msg_sdp_type, "answer");

			if(!answer) {

				JANUS_LOG(LOG_VERB, "This is a response to an offerless INVITE\n");

			}

				json_object_set_new(info, "event", json_string(answer ? "accepted" : "accepting"));

			if(!answer) {

				if(session->transaction)

					g_free(session->transaction);

				session->transaction = msg->transaction ? g_strdup(msg->transaction) : NULL;

			}

			json_object_set_new(result, "event", json_string(answer ? "accepted" : "accepting"));

			if(answer) {

				/* Start the media */

				session->media.ready = TRUE;	/* FIXME Maybe we need a better way to signal this */

				GError *error = NULL;

				char tname[16];

				g_snprintf(tname, sizeof(tname), "siprertp %s", session->account.username);

				g_thread_try_new(tname, janus_sipre_relay_thread, session, &error);

				if(error != NULL) {

					JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the RTP/RTCP thread...\n", error->code, error->message ? error->message : "??");

				}

				/* If we just updated the session, let's wait until things have calmed down */

				/* Check the socket error */

				int error = 0;

				socklen_t errlen = sizeof(error);

				getsockopt(fds[i].fd, SOL_SOCKET, SO_ERROR, (void *)&error, &errlen);

				if(error == 0) {

					/* Maybe not a breaking error after all? */

					continue;

				} else if(error == 111) {

					/* ICMP error? If it's related to RTCP, let's just close the RTCP socket and move on */

					if(fds[i].fd == session->media.audio_rtcp_fd) {

						JANUS_LOG(LOG_WARN, "[SIPre-%s] Got a '%s' on the audio RTCP socket, closing it\n",

							session->account.username, strerror(error));

						close(session->media.audio_rtcp_fd);

						session->media.audio_rtcp_fd = -1;

					} else if(fds[i].fd == session->media.video_rtcp_fd) {

						JANUS_LOG(LOG_WARN, "[SIPre-%s] Got a '%s' on the video RTCP socket, closing it\n",

							session->account.username, strerror(error));

						close(session->media.video_rtcp_fd);

						session->media.video_rtcp_fd = -1;

					}

					/* FIXME Should we do the same with the RTP sockets as well? We may risk overreacting, there... */

					continue;

				}

				JANUS_LOG(LOG_ERR, "[SIPre-%s] Error polling %d (socket #%d): %s...\n", session->account.username,

					fds[i].fd, i, fds[i].revents & POLLERR ? "POLLERR" : "POLLHUP");

				JANUS_LOG(LOG_ERR, "[SIPre-%s]   -- %d (%s)\n", session->account.username, error, strerror(error));

				/* Can we assume it's pretty much over, after a POLLERR? */

				goon = FALSE;

	/* New incoming call, check if there's an SDP to process */

	janus_sdp *sdp = NULL;

	const char *offer = (const char *)mbuf_buf(msg->mb);

	if(offer == NULL || mbuf_get_left(msg->mb) == 0) {

		JANUS_LOG(LOG_WARN, "[SIPre-%s] Received offerless INVITE\n", session->account.username);

	} else {

		g_snprintf(sdp_offer, sizeof(sdp_offer), "%.*s", (int)mbuf_get_left(msg->mb), offer);

		JANUS_LOG(LOG_WARN, "[SIPre-%s]   -- Offer: %s\n", session->account.username, sdp_offer);

		/* Parse the remote SDP */

		char sdperror[100];

		sdp = janus_sdp_parse(sdp_offer, sdperror, sizeof(sdperror));

		if(!sdp) {

			JANUS_LOG(LOG_ERR, "Error parsing SDP! %s\n", sdperror);

			mqueue_push(mq, janus_sipre_mqueue_event_do_rcode, janus_sipre_mqueue_payload_create(session, msg, 488, NULL));

			return;

		}

	if(sdp) {

		gboolean changed = FALSE;

		janus_sipre_sdp_process(session, sdp, FALSE, FALSE, &changed);

		/* Check if offer has neither audio nor video, fail with 488 */

		if(!session->media.has_audio && !session->media.has_video) {

			mqueue_push(mq, janus_sipre_mqueue_event_do_rcode, janus_sipre_mqueue_payload_create(session, msg, 488, NULL));

			janus_sdp_free(sdp);

			return;

		}

		/* Also fail with 488 if there's no remote IP address that can be used for RTP */

		if(!session->media.remote_ip) {

			mqueue_push(mq, janus_sipre_mqueue_event_do_rcode, janus_sipre_mqueue_payload_create(session, msg, 488, NULL));

			janus_sdp_free(sdp);

			return;

		}

	/* Notify the browser about the call */

	json_t *jsep = NULL;

	if(sdp)

		jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp_offer);

	if(sdp && session->media.has_srtp_remote) {

	if(jsep)

		json_decref(jsep);