rtp.c

			   predict we should be, use that */
			pred = rtp->lastts + f->datalen;
			break;
		case AST_FORMAT_ADPCM:
		case AST_FORMAT_G726:
			/* If we're within +/- 20ms from when where we
			   predict we should be, use that */
			pred = rtp->lastts + f->datalen * 2;
			break;
		case AST_FORMAT_G729A:
			pred = rtp->lastts + f->datalen * 8;
			break;
		case AST_FORMAT_GSM:
			pred = rtp->lastts + (f->datalen * 160 / 33);
			break;
		case AST_FORMAT_ILBC:
			pred = rtp->lastts + (f->datalen * 240 / 50);
			break;
		case AST_FORMAT_G723_1:
			pred = rtp->lastts + g723_samples(f->data, f->datalen);
			break;
		case AST_FORMAT_SPEEX:
		    pred = rtp->lastts + 160;
			// assumes that the RTP packet contains one Speex frame
			break;
		default:
			ast_log(LOG_WARNING, "Not sure about timestamp format for codec format %s\n", ast_getformatname(f->subclass));
		}
		/* Re-calculate last TS */
		rtp->lastts = rtp->lastts + ms * 8;
		if (!f->delivery.tv_sec && !f->delivery.tv_usec) {
			/* If this isn't an absolute delivery time, Check if it is close to our prediction, 
			   and if so, go with our prediction */
			if (abs(rtp->lastts - pred) < 640)
				rtp->lastts = pred;
			else {
				ast_log(LOG_DEBUG, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
				mark = 1;
			}
		}
	} else {
		mark = f->subclass & 0x1;
		pred = rtp->lastovidtimestamp + f->samples;
		/* Re-calculate last TS */
		rtp->lastts = rtp->lastts + ms * 90;
		/* If it's close to our prediction, go for it */
		if (abs(rtp->lastts - pred) < 7200) {
			rtp->lastts = pred;
			rtp->lastovidtimestamp += f->samples;
		} else {
			ast_log(LOG_DEBUG, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
			rtp->lastovidtimestamp = rtp->lastts;
		}
	}
	/* Get a pointer to the header */
	rtpheader = (unsigned int *)(f->data - hdrlen);
	rtpheader[0] = htonl((2 << 30) | (codec << 16) | (rtp->seqno++) | (mark << 23));
	rtpheader[1] = htonl(rtp->lastts);
	rtpheader[2] = htonl(rtp->ssrc); 
	if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
		res = sendto(rtp->s, (void *)rtpheader, f->datalen + hdrlen, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
		if (res <0) 
			ast_log(LOG_NOTICE, "RTP Transmission error to %s:%d: %s\n", inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
#if 0
		printf("Sent %d bytes of RTP data to %s:%d\n", res, inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
#endif		
	}
	return 0;
}

int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
{
	struct ast_frame *f;
	int codec;
	int hdrlen = 12;
	int subclass;
	

	/* If we have no peer, return immediately */	
	if (!rtp->them.sin_addr.s_addr)
		return 0;

	/* If there is no data length, return immediately */
	if (!_f->datalen) 
		return 0;
	
	/* Make sure we have enough space for RTP header */
	if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
		ast_log(LOG_WARNING, "RTP can only send voice\n");
		return -1;
	}

	subclass = _f->subclass;
	if (_f->frametype == AST_FRAME_VIDEO)
		subclass &= ~0x1;

	codec = ast_rtp_lookup_code(rtp, 1, subclass);
	if (codec < 0) {
		ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
		return -1;
	}

	if (rtp->lasttxformat != subclass) {
		/* New format, reset the smoother */
		ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
		rtp->lasttxformat = subclass;
		if (rtp->smoother)
			ast_smoother_free(rtp->smoother);
		rtp->smoother = NULL;
	}


	switch(subclass) {
	case AST_FORMAT_ULAW:
	case AST_FORMAT_ALAW:
		if (!rtp->smoother) {
			rtp->smoother = ast_smoother_new(160);
		}
		if (!rtp->smoother) {
			ast_log(LOG_WARNING, "Unable to create smoother :(\n");
			return -1;
		}
		ast_smoother_feed(rtp->smoother, _f);
		
		while((f = ast_smoother_read(rtp->smoother)))
			ast_rtp_raw_write(rtp, f, codec);
		break;
	case AST_FORMAT_ADPCM:
	case AST_FORMAT_G726:
		if (!rtp->smoother) {
			rtp->smoother = ast_smoother_new(80);
		}
		if (!rtp->smoother) {
			ast_log(LOG_WARNING, "Unable to create smoother :(\n");
			return -1;
		}
		ast_smoother_feed(rtp->smoother, _f);
		
		while((f = ast_smoother_read(rtp->smoother)))
			ast_rtp_raw_write(rtp, f, codec);
		break;
	case AST_FORMAT_G729A:
		if (!rtp->smoother) {
			rtp->smoother = ast_smoother_new(20);
			if (rtp->smoother)
				ast_smoother_set_flags(rtp->smoother, AST_SMOOTHER_FLAG_G729);
		}
		if (!rtp->smoother) {
			ast_log(LOG_WARNING, "Unable to create g729 smoother :(\n");
			return -1;
		}
		ast_smoother_feed(rtp->smoother, _f);
		
		while((f = ast_smoother_read(rtp->smoother)))
			ast_rtp_raw_write(rtp, f, codec);
		break;
	case AST_FORMAT_GSM:
		if (!rtp->smoother) {
			rtp->smoother = ast_smoother_new(33);
		}
		if (!rtp->smoother) {
			ast_log(LOG_WARNING, "Unable to create GSM smoother :(\n");
			return -1;
		}
		ast_smoother_feed(rtp->smoother, _f);
		while((f = ast_smoother_read(rtp->smoother)))
			ast_rtp_raw_write(rtp, f, codec);
		break;
	case AST_FORMAT_ILBC:
		if (!rtp->smoother) {
			rtp->smoother = ast_smoother_new(50);
		}
		if (!rtp->smoother) {
			ast_log(LOG_WARNING, "Unable to create ILBC smoother :(\n");
			return -1;
		}
		ast_smoother_feed(rtp->smoother, _f);
		while((f = ast_smoother_read(rtp->smoother)))
			ast_rtp_raw_write(rtp, f, codec);
		break;
	default:	
		ast_log(LOG_WARNING, "Not sure about sending format %s packets\n", ast_getformatname(subclass));
		// fall through to...
	case AST_FORMAT_H261:
	case AST_FORMAT_H263:
	case AST_FORMAT_G723_1:
	case AST_FORMAT_SPEEX:
	        // Don't buffer outgoing frames; send them one-per-packet:
		if (_f->offset < hdrlen) {
			f = ast_frdup(_f);
		} else {
			f = _f;
		}
		ast_rtp_raw_write(rtp, f, codec);
	}
		
	return 0;
}

void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
{
	struct ast_rtp_protocol *cur, *prev;
	cur = protos;
	prev = NULL;
	while(cur) {
		if (cur == proto) {
			if (prev)
				prev->next = proto->next;
			else
				protos = proto->next;
			return;
		}
		prev = cur;
		cur = cur->next;
	}
}

int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
{
	struct ast_rtp_protocol *cur;
	cur = protos;
	while(cur) {
		if (cur->type == proto->type) {
			ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
			return -1;
		}
		cur = cur->next;
	}
	proto->next = protos;
	protos = proto;
	return 0;
}

static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
{
	struct ast_rtp_protocol *cur;
	cur = protos;
	while(cur) {
		if (cur->type == chan->type) {
			return cur;
		}
		cur = cur->next;
	}
	return NULL;
}

int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc)
{
	struct ast_frame *f;
	struct ast_channel *who, *cs[3];
	struct ast_rtp *p0, *p1;
	struct ast_rtp *vp0, *vp1;
	struct ast_rtp_protocol *pr0, *pr1;
	struct sockaddr_in ac0, ac1;
	struct sockaddr_in vac0, vac1;
	struct sockaddr_in t0, t1;
	struct sockaddr_in vt0, vt1;
	
	void *pvt0, *pvt1;
	int to;
	memset(&vt0, 0, sizeof(vt0));
	memset(&vt1, 0, sizeof(vt1));
	memset(&vac0, 0, sizeof(vac0));
	memset(&vac1, 0, sizeof(vac1));

	/* XXX Wait a half a second for things to settle up 
			this really should be fixed XXX */
	ast_autoservice_start(c0);
	ast_autoservice_start(c1);
	usleep(500000);
	ast_autoservice_stop(c0);
	ast_autoservice_stop(c1);

	/* if need DTMF, cant native bridge */
	if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
		return -2;
	ast_mutex_lock(&c0->lock);
	ast_mutex_lock(&c1->lock);
	pr0 = get_proto(c0);
	pr1 = get_proto(c1);
	if (!pr0) {
		ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
		ast_mutex_unlock(&c0->lock);
		ast_mutex_unlock(&c1->lock);
		return -1;
	}
	if (!pr1) {
		ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
		ast_mutex_unlock(&c0->lock);
		ast_mutex_unlock(&c1->lock);
		return -1;
	}
	pvt0 = c0->pvt->pvt;
	pvt1 = c1->pvt->pvt;
	p0 = pr0->get_rtp_info(c0);
	if (pr0->get_vrtp_info)
		vp0 = pr0->get_vrtp_info(c0);
	else
		vp0 = NULL;
	p1 = pr1->get_rtp_info(c1);
	if (pr1->get_vrtp_info)
		vp1 = pr1->get_vrtp_info(c1);
	else
		vp1 = NULL;
	if (!p0 || !p1) {
		/* Somebody doesn't want to play... */
		ast_mutex_unlock(&c0->lock);
		ast_mutex_unlock(&c1->lock);
		return -2;
	}
	if (pr0->get_codec && pr1->get_codec) {
		int codec0,codec1;
		codec0 = pr0->get_codec(c0);
		codec1 = pr1->get_codec(c1);
		/* Hey, we can't do reinvite if both parties speak diffrent codecs */
		if (!(codec0 & codec1)) {
			ast_log(LOG_WARNING, "codec0 = %d is not codec1 = %d, cannot native bridge.\n",codec0,codec1);
			ast_mutex_unlock(&c0->lock);
			ast_mutex_unlock(&c1->lock);
			return -2;
		}
	}
	if (pr0->set_rtp_peer(c0, p1, vp1)) 
		ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
	else {
		/* Store RTP peer */
		ast_rtp_get_peer(p1, &ac1);
		if (vp1)
			ast_rtp_get_peer(p1, &vac1);
	}
	if (pr1->set_rtp_peer(c1, p0, vp0))
		ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name);
	else {
		/* Store RTP peer */
		ast_rtp_get_peer(p0, &ac0);
		if (vp0)
			ast_rtp_get_peer(p0, &vac0);
	}
	ast_mutex_unlock(&c0->lock);
	ast_mutex_unlock(&c1->lock);
	cs[0] = c0;
	cs[1] = c1;
	cs[2] = NULL;
	for (;;) {
		if ((c0->pvt->pvt != pvt0)  ||
			(c1->pvt->pvt != pvt1) ||
			(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
				ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
				if (c0->pvt->pvt == pvt0) {
					if (pr0->set_rtp_peer(c0, NULL, NULL)) 
						ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
				}
				if (c1->pvt->pvt == pvt1) {
					if (pr1->set_rtp_peer(c1, NULL, NULL)) 
						ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
				}
				/* Tell it to try again later */
				return -3;
		}
		to = -1;
		ast_rtp_get_peer(p1, &t1);
		ast_rtp_get_peer(p0, &t0);
		if (vp1)
			ast_rtp_get_peer(vp1, &vt1);
		if (vp0)
			ast_rtp_get_peer(vp0, &vt0);
		if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1))) {
			ast_log(LOG_DEBUG, "Oooh, '%s' changed end address\n", c1->name);
			if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL)) 
				ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
			memcpy(&ac1, &t1, sizeof(ac1));
			memcpy(&vac1, &vt1, sizeof(vac1));
		}
		if (inaddrcmp(&t0, &ac0) || (vp0 && inaddrcmp(&vt0, &vac0))) {
			ast_log(LOG_DEBUG, "Oooh, '%s' changed end address\n", c0->name);
			if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL))
				ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
			memcpy(&ac0, &t0, sizeof(ac0));
			memcpy(&vac0, &vt0, sizeof(vac0));
		}
		who = ast_waitfor_n(cs, 2, &to);
		if (!who) {
			ast_log(LOG_DEBUG, "Ooh, empty read...\n");
			/* check for hagnup / whentohangup */
			if (ast_check_hangup(c0) || ast_check_hangup(c1))
				break;
			continue;
		}
		f = ast_read(who);
		if (!f || ((f->frametype == AST_FRAME_DTMF) &&
				   (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) || 
			       ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
			*fo = f;
			*rc = who;
			ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
			if ((c0->pvt->pvt == pvt0) && (!c0->_softhangup)) {
				if (pr0->set_rtp_peer(c0, NULL, NULL)) 
					ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
			}
			if ((c1->pvt->pvt == pvt1) && (!c1->_softhangup)) {
				if (pr1->set_rtp_peer(c1, NULL, NULL)) 
					ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
			}
			/* That's all we needed */
			return 0;
		} else {
			if ((f->frametype == AST_FRAME_DTMF) || 
				(f->frametype == AST_FRAME_VOICE) || 
				(f->frametype == AST_FRAME_VIDEO)) {
				/* Forward voice or DTMF frames if they happen upon us */
				if (who == c0) {
					ast_write(c1, f);
				} else if (who == c1) {
					ast_write(c0, f);
				}
			}
			ast_frfree(f);
		}
		/* Swap priority not that it's a big deal at this point */
		cs[2] = cs[0];
		cs[0] = cs[1];
		cs[1] = cs[2];
		
	}
	return -1;
}

void ast_rtp_reload(void)
{
	struct ast_config *cfg;
	char *s;
	rtpstart = 5000;
	rtpend = 31000;
	cfg = ast_load("rtp.conf");
	if (cfg) {
		if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
			rtpstart = atoi(s);
			if (rtpstart < 1024)
				rtpstart = 1024;
			if (rtpstart > 65535)
				rtpstart = 65535;
		}
		if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
			rtpend = atoi(s);
			if (rtpend < 1024)
				rtpend = 1024;
			if (rtpend > 65535)
				rtpend = 65535;
		}
		ast_destroy(cfg);
	}
	if (rtpstart >= rtpend) {
		ast_log(LOG_WARNING, "Unreasonable values for RTP start/end\n");
		rtpstart = 5000;
		rtpend = 31000;
	}
	if (option_verbose > 1)
		ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
}

void ast_rtp_init(void)
{
	ast_rtp_reload();
}