rtp.c

			break;
	}
	AST_LIST_UNLOCK(&protos);

	return cur;
}

int ast_rtp_early_media(struct ast_channel *dest, struct ast_channel *src)
{
	struct ast_rtp *destp, *srcp=NULL;		/* Audio RTP Channels */
	struct ast_rtp *vdestp, *vsrcp=NULL;		/* Video RTP channels */
	struct ast_rtp_protocol *destpr, *srcpr=NULL;
	int srccodec;
	/* Lock channels */
	ast_channel_lock(dest);
	if (src) {
		while(ast_channel_trylock(src)) {
			ast_channel_unlock(dest);
			usleep(1);
			ast_channel_lock(dest);
		}
	}

	/* Find channel driver interfaces */
	destpr = get_proto(dest);
	if (src)
		srcpr = get_proto(src);
	if (!destpr) {
		if (option_debug)
			ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
		ast_channel_unlock(dest);
		if (src)
			ast_channel_unlock(src);
		return 0;
	}
	if (!srcpr) {
		if (option_debug)
			ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src ? src->name : "<unspecified>");
		ast_channel_unlock(dest);
		if (src)
			ast_channel_unlock(src);
		return 0;
	}

	/* Get audio and video interface (if native bridge is possible) */
	destp = destpr->get_rtp_info(dest);
	vdestp = (destpr->get_vrtp_info) ? destpr->get_vrtp_info(dest) : NULL;
	if (srcpr) {
		srcp = srcpr->get_rtp_info(src);
		vsrcp = (srcpr->get_vrtp_info) ? srcpr->get_vrtp_info(src) : NULL;
	}

	/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
	if (!destp) {
		/* Somebody doesn't want to play... */
		ast_channel_unlock(dest);
		if (src)
			ast_channel_unlock(src);
		return 0;
	}
	if (srcpr && srcpr->get_codec)
		srccodec = srcpr->get_codec(src);
	else
		srccodec = 0;
	/* Consider empty media as non-existant */
	if (srcp && !srcp->them.sin_addr.s_addr)
		srcp = NULL;
	/* Bridge early media */
	if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, srcp ? ast_test_flag(srcp, FLAG_NAT_ACTIVE) : 0))
		ast_log(LOG_WARNING, "Channel '%s' failed to send early media to '%s'\n", dest->name, src ? src->name : "<unspecified>");
	ast_channel_unlock(dest);
	if (src)
		ast_channel_unlock(src);
	if (option_debug)
		ast_log(LOG_DEBUG, "Setting early  media SDP of '%s' with that of '%s'\n", dest->name, src ? src->name : "<unspecified>");
	return 1;
}

int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, int media)
{
	struct ast_rtp *destp, *srcp;		/* Audio RTP Channels */
	struct ast_rtp *vdestp, *vsrcp;		/* Video RTP channels */
	struct ast_rtp_protocol *destpr, *srcpr;
	int srccodec;
	/* Lock channels */
	ast_channel_lock(dest);
	while(ast_channel_trylock(src)) {
		ast_channel_unlock(dest);
		usleep(1);
		ast_channel_lock(dest);
	}

	/* Find channel driver interfaces */
	destpr = get_proto(dest);
	srcpr = get_proto(src);
	if (!destpr) {
		if (option_debug)
			ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
		ast_channel_unlock(dest);
		ast_channel_unlock(src);
		return 0;
	}
	if (!srcpr) {
		if (option_debug)
			ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
		ast_channel_unlock(dest);
		ast_channel_unlock(src);
		return 0;
	}

	/* Get audio and video interface (if native bridge is possible) */
	destp = destpr->get_rtp_info(dest);
	vdestp = (destpr->get_vrtp_info) ? destpr->get_vrtp_info(dest) : NULL;
	srcp = srcpr->get_rtp_info(src);
	vsrcp = (srcpr->get_vrtp_info) ? srcpr->get_vrtp_info(src) : NULL;

	/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
	if (!destp || !srcp) {
		/* Somebody doesn't want to play... */
		ast_channel_unlock(dest);
		ast_channel_unlock(src);
		return 0;
	}
	ast_rtp_pt_copy(destp, srcp);
	if (vdestp && vsrcp)
		ast_rtp_pt_copy(vdestp, vsrcp);
	if (srcpr->get_codec)
		srccodec = srcpr->get_codec(src);
	else
		srccodec = 0;
	if (media) {
		/* Bridge early media */
		if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
			ast_log(LOG_WARNING, "Channel '%s' failed to send early media to '%s'\n", dest->name, src->name);
	}
	ast_channel_unlock(dest);
	ast_channel_unlock(src);
	if (option_debug)
		ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
	return 1;
}

/*! \brief  Make a note of a RTP paymoad type that was seen in a SDP "m=" line.
 * By default, use the well-known value for this type (although it may 
 * still be set to a different value by a subsequent "a=rtpmap:" line)
 */
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) 
{
	if (pt < 0 || pt > MAX_RTP_PT) 
		return; /* bogus payload type */

	if (static_RTP_PT[pt].code != 0) {
		rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
	}
} 

/*! \brief Make a note of a RTP payload type (with MIME type) that was seen in
 	a SDP "a=rtpmap:" line. */
void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
			 char* mimeType, char* mimeSubtype) 
{
	int i;

	if (pt < 0 || pt > MAX_RTP_PT) 
		return; /* bogus payload type */

	for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
		if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
		     strcasecmp(mimeType, mimeTypes[i].type) == 0) {
			rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
			return;
		}
	}
} 

/*! \brief Return the union of all of the codecs that were set by rtp_set...() calls 
 * They're returned as two distinct sets: AST_FORMATs, and AST_RTPs */
void ast_rtp_get_current_formats(struct ast_rtp* rtp,
			     int* astFormats, int* nonAstFormats) {
	int pt;

	*astFormats = *nonAstFormats = 0;
	for (pt = 0; pt < MAX_RTP_PT; ++pt) {
		if (rtp->current_RTP_PT[pt].isAstFormat) {
			*astFormats |= rtp->current_RTP_PT[pt].code;
		} else {
			*nonAstFormats |= rtp->current_RTP_PT[pt].code;
		}
	}
}

struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt) 
{
	struct rtpPayloadType result;

	result.isAstFormat = result.code = 0;
	if (pt < 0 || pt > MAX_RTP_PT) 
		return result; /* bogus payload type */

	/* Start with the negotiated codecs */
	result = rtp->current_RTP_PT[pt];

	/* If it doesn't exist, check our static RTP type list, just in case */
	if (!result.code) 
		result = static_RTP_PT[pt];
	return result;
}

/*! \brief Looks up an RTP code out of our *static* outbound list */
int ast_rtp_lookup_code(struct ast_rtp* rtp, const int isAstFormat, const int code) {

	int pt;

	if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
		code == rtp->rtp_lookup_code_cache_code) {

		/* Use our cached mapping, to avoid the overhead of the loop below */
		return rtp->rtp_lookup_code_cache_result;
	}

	/* Check the dynamic list first */
	for (pt = 0; pt < MAX_RTP_PT; ++pt) {
  		if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
			rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
			rtp->rtp_lookup_code_cache_code = code;
			rtp->rtp_lookup_code_cache_result = pt;
			return pt;
		}
	}

	/* Then the static list */
	for (pt = 0; pt < MAX_RTP_PT; ++pt) {
		if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
			rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
  			rtp->rtp_lookup_code_cache_code = code;
			rtp->rtp_lookup_code_cache_result = pt;
			return pt;
		}
	}
	return -1;
}

char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code) 
{

	int i;

	for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
		if (mimeTypes[i].payloadType.code == code && mimeTypes[i].payloadType.isAstFormat == isAstFormat)
			return mimeTypes[i].subtype;
	}
	return "";
}

char *ast_rtp_lookup_mime_multiple(char *buf, int size, const int capability, const int isAstFormat)
{
	int format;
	unsigned len;
	char *end = buf;
	char *start = buf;

	if (!buf || !size)
		return NULL;

	snprintf(end, size, "0x%x (", capability);

	len = strlen(end);
	end += len;
	size -= len;
	start = end;

	for (format = 1; format < AST_RTP_MAX; format <<= 1) {
		if (capability & format) {
			const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format);
			snprintf(end, size, "%s|", name);
			len = strlen(end);
			end += len;
			size -= len;
		}
	}

	if (start == end)
		snprintf(start, size, "nothing)"); 
	else if (size > 1)
		*(end -1) = ')';
	
	return buf;
}

static int rtp_socket(void)
{
	int s;
	long flags;
	s = socket(AF_INET, SOCK_DGRAM, 0);
	if (s > -1) {
		flags = fcntl(s, F_GETFL);
		fcntl(s, F_SETFL, flags | O_NONBLOCK);
#ifdef SO_NO_CHECK
		if (nochecksums)
			setsockopt(s, SOL_SOCKET, SO_NO_CHECK, &nochecksums, sizeof(nochecksums));
#endif
	}
	return s;
}

/*!
 * \brief Initialize a new RTCP session.
 * 
 * \returns The newly initialized RTCP session.
 */
static struct ast_rtcp *ast_rtcp_new(void)
{
	struct ast_rtcp *rtcp;

	if (!(rtcp = ast_calloc(1, sizeof(*rtcp))))
		return NULL;
	rtcp->s = rtp_socket();
	rtcp->us.sin_family = AF_INET;
	if (rtcp->s < 0) {
		free(rtcp);
		ast_log(LOG_WARNING, "Unable to allocate socket: %s\n", strerror(errno));
		return NULL;
	}

	return rtcp;
}

struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr addr)
{
	struct ast_rtp *rtp;
	int x;
	int first;
	int startplace;
	
	if (!(rtp = ast_calloc(1, sizeof(*rtp))))
		return NULL;
	rtp->them.sin_family = AF_INET;
	rtp->us.sin_family = AF_INET;
	rtp->s = rtp_socket();
	rtp->ssrc = ast_random();
	rtp->seqno = ast_random() & 0xffff;
	if (rtp->s < 0) {
		free(rtp);
		ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));
		return NULL;
	}
	if (sched && rtcpenable) {
		rtp->sched = sched;
		rtp->rtcp = ast_rtcp_new();
	}
	
	/* Select a random port number in the range of possible RTP */
	x = (ast_random() % (rtpend-rtpstart)) + rtpstart;
	x = x & ~1;
	/* Save it for future references. */
	startplace = x;
	/* Iterate tring to bind that port and incrementing it otherwise untill a port was found or no ports are available. */
	for (;;) {
		/* Must be an even port number by RTP spec */
		rtp->us.sin_port = htons(x);
		rtp->us.sin_addr = addr;
		/* If there's rtcp, initialize it as well. */
		if (rtp->rtcp)
			rtp->rtcp->us.sin_port = htons(x + 1);
		/* Try to bind it/them. */
		if (!(first = bind(rtp->s, (struct sockaddr *)&rtp->us, sizeof(rtp->us))) &&
			(!rtp->rtcp || !bind(rtp->rtcp->s, (struct sockaddr *)&rtp->rtcp->us, sizeof(rtp->rtcp->us))))
			break;
		if (!first) {
			/* Primary bind succeeded! Gotta recreate it */
			close(rtp->s);
			rtp->s = rtp_socket();
		}
		if (errno != EADDRINUSE) {
			/* We got an error that wasn't expected, abort! */
			ast_log(LOG_ERROR, "Unexpected bind error: %s\n", strerror(errno));
			close(rtp->s);
			if (rtp->rtcp) {
				close(rtp->rtcp->s);
				free(rtp->rtcp);
			}
			free(rtp);
			return NULL;
		}
		/* The port was used, increment it (by two). */
		x += 2;
		/* Did we go over the limit ? */
		if (x > rtpend)
			/* then, start from the begingig. */
			x = (rtpstart + 1) & ~1;
		/* Check if we reached the place were we started. */
		if (x == startplace) {
			/* If so, there's no ports available. */
			ast_log(LOG_ERROR, "No RTP ports remaining. Can't setup media stream for this call.\n");
			close(rtp->s);
			if (rtp->rtcp) {
				close(rtp->rtcp->s);
				free(rtp->rtcp);
			}
			free(rtp);
			return NULL;
		}
	}
	if (io && sched && callbackmode) {
		/* Operate this one in a callback mode */
		rtp->sched = sched;
		rtp->io = io;
		rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
	}
	ast_rtp_pt_default(rtp);
	return rtp;
}

struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode)
{
	struct in_addr ia;

	memset(&ia, 0, sizeof(ia));
	return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
}

int ast_rtp_settos(struct ast_rtp *rtp, int tos)
{
	int res;

	if ((res = setsockopt(rtp->s, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)))) 
		ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
	return res;
}

void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
{
	rtp->them.sin_port = them->sin_port;
	rtp->them.sin_addr = them->sin_addr;
	if (rtp->rtcp) {
		rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1);
		rtp->rtcp->them.sin_addr = them->sin_addr;
	}
	rtp->rxseqno = 0;
}

int ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
{
	if ((them->sin_family != AF_INET) ||
		(them->sin_port != rtp->them.sin_port) ||
		(them->sin_addr.s_addr != rtp->them.sin_addr.s_addr)) {
		them->sin_family = AF_INET;
		them->sin_port = rtp->them.sin_port;
		them->sin_addr = rtp->them.sin_addr;
		return 1;
	}
	return 0;
}

void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
{
	*us = rtp->us;
}

void ast_rtp_stop(struct ast_rtp *rtp)
{
	memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
	memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
	if (rtp->rtcp) {
		memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
		memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->them.sin_port));
	}
}

void ast_rtp_reset(struct ast_rtp *rtp)
{
	memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
	memset(&rtp->txcore, 0, sizeof(rtp->txcore));
	memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
	rtp->lastts = 0;
	rtp->lastdigitts = 0;
	rtp->lastrxts = 0;
	rtp->lastividtimestamp = 0;
	rtp->lastovidtimestamp = 0;
	rtp->lasteventseqn = 0;
	rtp->lasteventendseqn = 0;
	rtp->lasttxformat = 0;
	rtp->lastrxformat = 0;
	rtp->dtmfcount = 0;
	rtp->dtmfduration = 0;
	rtp->seqno = 0;
	rtp->rxseqno = 0;
}

void ast_rtp_destroy(struct ast_rtp *rtp)
{
	if (rtp->smoother)
		ast_smoother_free(rtp->smoother);
	if (rtp->ioid)
		ast_io_remove(rtp->io, rtp->ioid);
	if (rtp->s > -1)
		close(rtp->s);
	if (rtp->rtcp) {
		close(rtp->rtcp->s);
		free(rtp->rtcp);
	}
	free(rtp);
}

static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval *delivery)
{
	struct timeval t;
	long ms;
	if (ast_tvzero(rtp->txcore)) {
		rtp->txcore = ast_tvnow();
		/* Round to 20ms for nice, pretty timestamps */
		rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
	}
	/* Use previous txcore if available */
	t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
	ms = ast_tvdiff_ms(t, rtp->txcore);
	if (ms < 0)
		ms = 0;
	/* Use what we just got for next time */
	rtp->txcore = t;
	return (unsigned int) ms;
}

int ast_rtp_senddigit(struct ast_rtp *rtp, char digit)
{
	unsigned int *rtpheader;
	int hdrlen = 12;
	int res;
	int x;
	int payload;
	char data[256];
	char iabuf[INET_ADDRSTRLEN];

	if ((digit <= '9') && (digit >= '0'))
		digit -= '0';
	else if (digit == '*')
		digit = 10;
	else if (digit == '#')
		digit = 11;
	else if ((digit >= 'A') && (digit <= 'D')) 
		digit = digit - 'A' + 12;
	else if ((digit >= 'a') && (digit <= 'd')) 
		digit = digit - 'a' + 12;
	else {
		ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
		return -1;
	}
	payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);

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

	rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
	
	/* Get a pointer to the header */
	rtpheader = (unsigned int *)data;
	rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
	rtpheader[1] = htonl(rtp->lastdigitts);
	rtpheader[2] = htonl(rtp->ssrc); 
	rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0));
	for (x = 0; x < 6; x++) {
		if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
			res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
			if (res < 0) 
				ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
					ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr),
					ntohs(rtp->them.sin_port), strerror(errno));
			if (rtp_debug_test_addr(&rtp->them))
				ast_verbose("Sent RTP packet to %s:%d (type %d, seq %u, ts %u, len %u)\n",
					    ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr),
					    ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
		}
		/* Sequence number of last two end packets does not get incremented */
		if (x < 3)
			rtp->seqno++;
		/* Clear marker bit and set seqno */
		rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
		/* For the last three packets, set the duration and the end bit */
		if (x == 2) {
#if 0
			/* No, this is wrong...  Do not increment lastdigitts, that's not according
			   to the RFC, as best we can determine */
			rtp->lastdigitts++; /* or else the SPA3000 will click instead of beeping... */
			rtpheader[1] = htonl(rtp->lastdigitts);
#endif			
			/* Make duration 800 (100ms) */
			rtpheader[3] |= htonl((800));
			/* Set the End bit */
			rtpheader[3] |= htonl((1 << 23));
		}
	}
	/* Increment the digit timestamp by 120ms, to ensure that digits
	   sent sequentially with no intervening non-digit packets do not
	   get sent with the same timestamp, and that sequential digits
	   have some 'dead air' in between them
	*/
	rtp->lastdigitts += 960;
	/* Increment the sequence number to reflect the last packet
	   that was sent
	*/
	rtp->seqno++;
	return 0;
}

int ast_rtp_sendcng(struct ast_rtp *rtp, int level)
{
	unsigned int *rtpheader;
	int hdrlen = 12;
	int res;
	int payload;
	char data[256];
	char iabuf[INET_ADDRSTRLEN];
	level = 127 - (level & 0x7f);
	payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN);

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

	rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));

	/* Get a pointer to the header */
	rtpheader = (unsigned int *)data;
	rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
	rtpheader[1] = htonl(rtp->lastts);
	rtpheader[2] = htonl(rtp->ssrc); 
	data[12] = level;
	if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
		res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
		if (res <0) 
			ast_log(LOG_ERROR, "RTP Comfort Noise Transmission error to %s:%d: %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
		if(rtp_debug_test_addr(&rtp->them))
			ast_verbose("Sent Comfort Noise RTP packet to %s:%d (type %d, seq %d, ts %d, len %d)\n"
					, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);		   
		   
	}
	return 0;
}

static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec)
{
	unsigned char *rtpheader;
	char iabuf[INET_ADDRSTRLEN];
	int hdrlen = 12;
	int res;
	unsigned int ms;
	int pred;
	int mark = 0;

	ms = calc_txstamp(rtp, &f->delivery);
	/* Default prediction */
	if (f->subclass < AST_FORMAT_MAX_AUDIO) {
		pred = rtp->lastts + f->samples;

		/* Re-calculate last TS */
		rtp->lastts = rtp->lastts + ms * 8;
		if (ast_tvzero(f->delivery)) {
			/* 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) < MAX_TIMESTAMP_SKEW)
				rtp->lastts = pred;
			else {
				if (option_debug > 2)
					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 (ast_tvzero(f->delivery)) {
			if (abs(rtp->lastts - pred) < 7200) {
				rtp->lastts = pred;
				rtp->lastovidtimestamp += f->samples;
			} else {
				if (option_debug > 2)
					ast_log(LOG_DEBUG, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, f->samples);
				rtp->lastovidtimestamp = rtp->lastts;
			}
		}
	}
	/* If the timestamp for non-digit packets has moved beyond the timestamp
	   for digits, update the digit timestamp.
	*/
	if (rtp->lastts > rtp->lastdigitts)
		rtp->lastdigitts = rtp->lastts;

	/* Get a pointer to the header */
	rtpheader = (unsigned char *)(f->data - hdrlen);

	put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (rtp->seqno) | (mark << 23)));
	put_unaligned_uint32(rtpheader + 4, htonl(rtp->lastts));
	put_unaligned_uint32(rtpheader + 8, 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) {
			if (!rtp->nat || (rtp->nat && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
				ast_log(LOG_DEBUG, "RTP Transmission error of packet %d to %s:%d: %s\n", rtp->seqno, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
			} else if ((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) {
				/* Only give this error message once if we are not RTP debugging */
				if (option_debug || rtpdebug)
					ast_log(LOG_DEBUG, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port));
				ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
			}
		}
				
		if(rtp_debug_test_addr(&rtp->them))
			ast_verbose("Sent RTP packet to %s:%d (type %d, seq %u, ts %u, len %u)\n"
					, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), codec, rtp->seqno, rtp->lastts,res - hdrlen);
	}

	rtp->seqno++;

	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 */
		if (option_debug)
			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_SLINEAR:
		if (!rtp->smoother) {
			rtp->smoother = ast_smoother_new(320);
		}
		if (!rtp->smoother) {
			ast_log(LOG_WARNING, "Unable to create smoother :(\n");
			return -1;
		}
		ast_smoother_feed_be(rtp->smoother, _f);
		
		while((f = ast_smoother_read(rtp->smoother)))
			ast_rtp_raw_write(rtp, f, codec);
		break;
	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_H263_PLUS:
	case AST_FORMAT_H264:
	case AST_FORMAT_G723_1:
	case AST_FORMAT_LPC10:
	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;
}

/*! \brief Unregister interface to channel driver */
void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
{
	AST_LIST_LOCK(&protos);
	AST_LIST_REMOVE(&protos, proto, list);
	AST_LIST_UNLOCK(&protos);
}

/*! \brief Register interface to channel driver */
int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
{
	struct ast_rtp_protocol *cur;

	AST_LIST_LOCK(&protos);
	AST_LIST_TRAVERSE(&protos, cur, list) {	
		if (!strcmp(cur->type, proto->type)) {
			ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
			AST_LIST_UNLOCK(&protos);
			return -1;
		}
	}
	AST_LIST_INSERT_HEAD(&protos, proto, list);
	AST_LIST_UNLOCK(&protos);
	
	return 0;
}

/*! \brief Bridge calls. If possible and allowed, initiate
	re-invite so the peers exchange media directly outside 
	of Asterisk. */
enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
{
	struct ast_frame *f;
	struct ast_channel *who, *other, *cs[3];
	struct ast_rtp *p0, *p1;		/* Audio RTP Channels */
	struct ast_rtp *vp0, *vp1;		/* Video RTP channels */
	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;
	char iabuf[INET_ADDRSTRLEN];
	
	void *pvt0, *pvt1;
	int codec0,codec1, oldcodec0, oldcodec1;
	
	memset(&vt0, 0, sizeof(vt0));
	memset(&vt1, 0, sizeof(vt1));
	memset(&vac0, 0, sizeof(vac0));
	memset(&vac1, 0, sizeof(vac1));

	/* if need DTMF, cant native bridge */
	if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
		return AST_BRIDGE_FAILED_NOWARN;

	/* Lock channels */
	ast_channel_lock(c0);
	while(ast_channel_trylock(c1)) {
		ast_channel_unlock(c0);
		usleep(1);
		ast_channel_lock(c0);
	}

	/* Find channel driver interfaces */
	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_channel_unlock(c0);
		ast_channel_unlock(c1);
		return AST_BRIDGE_FAILED;
	}
	if (!pr1) {
		ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
		ast_channel_unlock(c0);
		ast_channel_unlock(c1);
		return AST_BRIDGE_FAILED;
	}

	/* Get channel specific interface structures */
	pvt0 = c0->tech_pvt;
	pvt1 = c1->tech_pvt;

	/* Get audio and video interface (if native bridge is possible) */
	p0 = pr0->get_rtp_info(c0);
	vp0 = pr0->get_vrtp_info ? pr0->get_vrtp_info(c0) : NULL;
	p1 = pr1->get_rtp_info(c1);
	vp1 = pr1->get_vrtp_info ? pr1->get_vrtp_info(c1) : NULL;

	/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
	if (!p0 || !p1) {
		/* Somebody doesn't want to play... */
		ast_channel_unlock(c0);
		ast_channel_unlock(c1);
		return AST_BRIDGE_FAILED_NOWARN;
	}
	/* Get codecs from both sides */
	codec0 = pr0->get_codec ? pr0->get_codec(c0) : 0;
	codec1 = pr1->get_codec ? pr1->get_codec(c1) : 0;
	if (pr0->get_codec && pr1->get_codec) {
		/* Hey, we can't do reinvite if both parties speak different codecs */
		if (!(codec0 & codec1)) {
			if (option_debug)
				ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
			ast_channel_unlock(c0);
			ast_channel_unlock(c1);
			return AST_BRIDGE_FAILED_NOWARN;
		}
	}

	if (option_verbose > 2) 
		ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);

	/* Ok, we should be able to redirect the media. Start with one channel */
	if (pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE))) 
		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(vp1, &vac1);
	}
	/* Then test the other channel */
	if (pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
		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(vp0, &vac0);
	}
	ast_channel_unlock(c0);
	ast_channel_unlock(c1);