/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2006, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*! 
 * \file 
 *
 * \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
 *
 * \author Mark Spencer <markster@digium.com>
 * 
 * \note RTP is defined in RFC 3550.
 */

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <sys/time.h>
#include <signal.h>
#include <errno.h>
#include <unistd.h>
#include <netinet/in.h>
#include <sys/time.h>
#include <sys/socket.h>
#include <arpa/inet.h>
#include <fcntl.h>

#include "asterisk/rtp.h"
#include "asterisk/frame.h"
#include "asterisk/logger.h"
#include "asterisk/options.h"
#include "asterisk/channel.h"
#include "asterisk/acl.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/lock.h"
#include "asterisk/utils.h"
#include "asterisk/cli.h"
#include "asterisk/unaligned.h"
#include "asterisk/utils.h"

#define MAX_TIMESTAMP_SKEW	640

#define RTP_SEQ_MOD     (1<<16) 	/*!< A sequence number can't be more than 16 bits */
#define RTCP_DEFAULT_INTERVALMS   5000	/*!< Default milli-seconds between RTCP reports we send */
#define RTCP_MIN_INTERVALMS       500	/*!< Min milli-seconds between RTCP reports we send */
#define RTCP_MAX_INTERVALMS       60000	/*!< Max milli-seconds between RTCP reports we send */

#define RTCP_PT_FUR     192
#define RTCP_PT_SR      200
#define RTCP_PT_RR      201
#define RTCP_PT_SDES    202
#define RTCP_PT_BYE     203
#define RTCP_PT_APP     204

#define RTP_MTU		1200

#define DEFAULT_DTMF_TIMEOUT 3000	/*!< samples */

static int dtmftimeout = DEFAULT_DTMF_TIMEOUT;

static int rtpstart = 0;		/*!< First port for RTP sessions (set in rtp.conf) */
static int rtpend = 0;			/*!< Last port for RTP sessions (set in rtp.conf) */
static int rtpdebug = 0;		/*!< Are we debugging? */
static int rtcpdebug = 0;		/*!< Are we debugging RTCP? */
static int rtcpstats = 0;		/*!< Are we debugging RTCP? */
static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
static int stundebug = 0;		/*!< Are we debugging stun? */
static struct sockaddr_in rtpdebugaddr;	/*!< Debug packets to/from this host */
static struct sockaddr_in rtcpdebugaddr;	/*!< Debug RTCP packets to/from this host */
#ifdef SO_NO_CHECK
static int nochecksums = 0;
#endif

/*!
 * \brief Structure representing a RTP session.
 *
 * RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP.  A participant may be involved in multiple RTP sessions at the same time [...]"
 *
 */
/*! \brief The value of each payload format mapping: */
struct rtpPayloadType {
	int isAstFormat; 	/*!< whether the following code is an AST_FORMAT */
	int code;
};


/*! \brief RTP session description */
struct ast_rtp {
	int s;
	char resp;
	struct ast_frame f;
	unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
	unsigned int ssrc;		/*!< Synchronization source, RFC 3550, page 10. */
	unsigned int themssrc;		/*!< Their SSRC */
	unsigned int rxssrc;
	unsigned int lastts;
	unsigned int lastdigitts;
	unsigned int lastrxts;
	unsigned int lastividtimestamp;
	unsigned int lastovidtimestamp;
	unsigned int lasteventseqn;
	int lastrxseqno;                /*!< Last received sequence number */
	unsigned short seedrxseqno;     /*!< What sequence number did they start with?*/
	unsigned int seedrxts;          /*!< What RTP timestamp did they start with? */
	unsigned int rxcount;           /*!< How many packets have we received? */
	unsigned int rxoctetcount;      /*!< How many octets have we received? should be rxcount *160*/
	unsigned int txcount;           /*!< How many packets have we sent? */
	unsigned int txoctetcount;      /*!< How many octets have we sent? (txcount*160)*/
	unsigned int cycles;            /*!< Shifted count of sequence number cycles */
	double rxjitter;                /*!< Interarrival jitter at the moment */
	double rxtransit;               /*!< Relative transit time for previous packet */
	unsigned int lasteventendseqn;
	int lasttxformat;
	int lastrxformat;
	int dtmfcount;
	unsigned int dtmfduration;
	int nat;
	unsigned int flags;
	struct sockaddr_in us;		/*!< Socket representation of the local endpoint. */
	struct sockaddr_in them;	/*!< Socket representation of the remote endpoint. */
	struct timeval rxcore;
	struct timeval txcore;
	double drxcore;                 /*!< The double representation of the first received packet */
	struct timeval lastrx;          /*!< timeval when we last received a packet */
	struct timeval dtmfmute;
	struct ast_smoother *smoother;
	int *ioid;
	unsigned short seqno;		/*!< Sequence number, RFC 3550, page 13. */
	unsigned short rxseqno;
	struct sched_context *sched;
	struct io_context *io;
	void *data;
	ast_rtp_callback callback;
	struct rtpPayloadType current_RTP_PT[MAX_RTP_PT];
	int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */
	int rtp_lookup_code_cache_code;
	int rtp_lookup_code_cache_result;
	struct ast_rtcp *rtcp;
};

/* Forward declarations */
static int ast_rtcp_write(void *data);
static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw);
static int ast_rtcp_write_sr(void *data);
static int ast_rtcp_write_rr(void *data);
static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp);

#define FLAG_3389_WARNING		(1 << 0)
#define FLAG_NAT_ACTIVE			(3 << 1)
#define FLAG_NAT_INACTIVE		(0 << 1)
#define FLAG_NAT_INACTIVE_NOWARN	(1 << 1)
#define FLAG_HAS_DTMF			(1 << 3)

/*!
 * \brief Structure defining an RTCP session.
 * 
 * The concept "RTCP session" is not defined in RFC 3550, but since 
 * this structure is analogous to ast_rtp, which tracks a RTP session, 
 * it is logical to think of this as a RTCP session.
 *
 * RTCP packet is defined on page 9 of RFC 3550.
 * 
 */
struct ast_rtcp {
	int s;				/*!< Socket */
	struct sockaddr_in us;		/*!< Socket representation of the local endpoint. */
	struct sockaddr_in them;	/*!< Socket representation of the remote endpoint. */
	unsigned int soc;		/*!< What they told us */
	unsigned int spc;		/*!< What they told us */
	unsigned int themrxlsr;		/*!< The middle 32 bits of the NTP timestamp in the last received SR*/
	struct timeval rxlsr;		/*!< Time when we got their last SR */
	struct timeval txlsr;		/*!< Time when we sent or last SR*/
	unsigned int expected_prior;	/*!< no. packets in previous interval */
	unsigned int received_prior;	/*!< no. packets received in previous interval */
	int schedid;			/*!< Schedid returned from ast_sched_add() to schedule RTCP-transmissions*/
	unsigned int rr_count;		/*!< number of RRs we've sent, not including report blocks in SR's */
	unsigned int sr_count;		/*!< number of SRs we've sent */
	unsigned int lastsrtxcount;     /*!< Transmit packet count when last SR sent */
	double accumulated_transit;	/*!< accumulated a-dlsr-lsr */
	double rtt;			/*!< Last reported rtt */
	unsigned int reported_jitter;	/*!< The contents of their last jitter entry in the RR */
	unsigned int reported_lost;	/*!< Reported lost packets in their RR */
	char quality[AST_MAX_USER_FIELD];
	double maxrxjitter;
	double minrxjitter;
	double maxrtt;
	double minrtt;
	int sendfur;
};


typedef struct { unsigned int id[4]; } __attribute__((packed)) stun_trans_id;

/* XXX Maybe stun belongs in another file if it ever has use outside of RTP */
struct stun_header {
	unsigned short msgtype;
	unsigned short msglen;
	stun_trans_id  id;
	unsigned char ies[0];
} __attribute__((packed));

struct stun_attr {
	unsigned short attr;
	unsigned short len;
	unsigned char value[0];
} __attribute__((packed));

struct stun_addr {
	unsigned char unused;
	unsigned char family;
	unsigned short port;
	unsigned int addr;
} __attribute__((packed));

#define STUN_IGNORE		(0)
#define STUN_ACCEPT		(1)

#define STUN_BINDREQ	0x0001
#define STUN_BINDRESP	0x0101
#define STUN_BINDERR	0x0111
#define STUN_SECREQ	0x0002
#define STUN_SECRESP	0x0102
#define STUN_SECERR	0x0112

#define STUN_MAPPED_ADDRESS	0x0001
#define STUN_RESPONSE_ADDRESS	0x0002
#define STUN_CHANGE_REQUEST	0x0003
#define STUN_SOURCE_ADDRESS	0x0004
#define STUN_CHANGED_ADDRESS	0x0005
#define STUN_USERNAME		0x0006
#define STUN_PASSWORD		0x0007
#define STUN_MESSAGE_INTEGRITY	0x0008
#define STUN_ERROR_CODE		0x0009
#define STUN_UNKNOWN_ATTRIBUTES	0x000a
#define STUN_REFLECTED_FROM	0x000b

static const char *stun_msg2str(int msg)
{
	switch(msg) {
	case STUN_BINDREQ:
		return "Binding Request";
	case STUN_BINDRESP:
		return "Binding Response";
	case STUN_BINDERR:
		return "Binding Error Response";
	case STUN_SECREQ:
		return "Shared Secret Request";
	case STUN_SECRESP:
		return "Shared Secret Response";
	case STUN_SECERR:
		return "Shared Secret Error Response";
	}
	return "Non-RFC3489 Message";
}

static const char *stun_attr2str(int msg)
{
	switch(msg) {
	case STUN_MAPPED_ADDRESS:
		return "Mapped Address";
	case STUN_RESPONSE_ADDRESS:
		return "Response Address";
	case STUN_CHANGE_REQUEST:
		return "Change Request";
	case STUN_SOURCE_ADDRESS:
		return "Source Address";
	case STUN_CHANGED_ADDRESS:
		return "Changed Address";
	case STUN_USERNAME:
		return "Username";
	case STUN_PASSWORD:
		return "Password";
	case STUN_MESSAGE_INTEGRITY:
		return "Message Integrity";
	case STUN_ERROR_CODE:
		return "Error Code";
	case STUN_UNKNOWN_ATTRIBUTES:
		return "Unknown Attributes";
	case STUN_REFLECTED_FROM:
		return "Reflected From";
	}
	return "Non-RFC3489 Attribute";
}

struct stun_state {
	const char *username;
	const char *password;
};

static int stun_process_attr(struct stun_state *state, struct stun_attr *attr)
{
	if (stundebug)
		ast_verbose("Found STUN Attribute %s (%04x), length %d\n",
			stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len));
	switch(ntohs(attr->attr)) {
	case STUN_USERNAME:
		state->username = (const char *) (attr->value);
		break;
	case STUN_PASSWORD:
		state->password = (const char *) (attr->value);
		break;
	default:
		if (stundebug)
			ast_verbose("Ignoring STUN attribute %s (%04x), length %d\n", 
				stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr), ntohs(attr->len));
	}
	return 0;
}

static void append_attr_string(struct stun_attr **attr, int attrval, const char *s, int *len, int *left)
{
	int size = sizeof(**attr) + strlen(s);
	if (*left > size) {
		(*attr)->attr = htons(attrval);
		(*attr)->len = htons(strlen(s));
		memcpy((*attr)->value, s, strlen(s));
		(*attr) = (struct stun_attr *)((*attr)->value + strlen(s));
		*len += size;
		*left -= size;
	}
}

static void append_attr_address(struct stun_attr **attr, int attrval, struct sockaddr_in *sin, int *len, int *left)
{
	int size = sizeof(**attr) + 8;
	struct stun_addr *addr;
	if (*left > size) {
		(*attr)->attr = htons(attrval);
		(*attr)->len = htons(8);
		addr = (struct stun_addr *)((*attr)->value);
		addr->unused = 0;
		addr->family = 0x01;
		addr->port = sin->sin_port;
		addr->addr = sin->sin_addr.s_addr;
		(*attr) = (struct stun_attr *)((*attr)->value + 8);
		*len += size;
		*left -= size;
	}
}

static int stun_send(int s, struct sockaddr_in *dst, struct stun_header *resp)
{
	return sendto(s, resp, ntohs(resp->msglen) + sizeof(*resp), 0,
		(struct sockaddr *)dst, sizeof(*dst));
}

static void stun_req_id(struct stun_header *req)
{
	int x;
	for (x=0;x<4;x++)
		req->id.id[x] = ast_random();
}

size_t ast_rtp_alloc_size(void)
{
	return sizeof(struct ast_rtp);
}

void ast_rtp_stun_request(struct ast_rtp *rtp, struct sockaddr_in *suggestion, const char *username)
{
	struct stun_header *req;
	unsigned char reqdata[1024];
	int reqlen, reqleft;
	struct stun_attr *attr;

	req = (struct stun_header *)reqdata;
	stun_req_id(req);
	reqlen = 0;
	reqleft = sizeof(reqdata) - sizeof(struct stun_header);
	req->msgtype = 0;
	req->msglen = 0;
	attr = (struct stun_attr *)req->ies;
	if (username)
		append_attr_string(&attr, STUN_USERNAME, username, &reqlen, &reqleft);
	req->msglen = htons(reqlen);
	req->msgtype = htons(STUN_BINDREQ);
	stun_send(rtp->s, suggestion, req);
}

static int stun_handle_packet(int s, struct sockaddr_in *src, unsigned char *data, size_t len)
{
	struct stun_header *resp, *hdr = (struct stun_header *)data;
	struct stun_attr *attr;
	struct stun_state st;
	int ret = STUN_IGNORE;	
	unsigned char respdata[1024];
	int resplen, respleft;
	
	if (len < sizeof(struct stun_header)) {
		if (option_debug)
			ast_log(LOG_DEBUG, "Runt STUN packet (only %zd, wanting at least %zd)\n", len, sizeof(struct stun_header));
		return -1;
	}
	if (stundebug)
		ast_verbose("STUN Packet, msg %s (%04x), length: %d\n", stun_msg2str(ntohs(hdr->msgtype)), ntohs(hdr->msgtype), ntohs(hdr->msglen));
	if (ntohs(hdr->msglen) > len - sizeof(struct stun_header)) {
		if (option_debug)
			ast_log(LOG_DEBUG, "Scrambled STUN packet length (got %d, expecting %zd)\n", ntohs(hdr->msglen), len - sizeof(struct stun_header));
	} else
		len = ntohs(hdr->msglen);
	data += sizeof(struct stun_header);
	memset(&st, 0, sizeof(st));
	while(len) {
		if (len < sizeof(struct stun_attr)) {
			if (option_debug)
				ast_log(LOG_DEBUG, "Runt Attribute (got %zd, expecting %zd)\n", len, sizeof(struct stun_attr));
			break;
		}
		attr = (struct stun_attr *)data;
		if (ntohs(attr->len) > len) {
			if (option_debug)
				ast_log(LOG_DEBUG, "Inconsistent Attribute (length %d exceeds remaining msg len %zd)\n", ntohs(attr->len), len);
			break;
		}
		if (stun_process_attr(&st, attr)) {
			if (option_debug)
				ast_log(LOG_DEBUG, "Failed to handle attribute %s (%04x)\n", stun_attr2str(ntohs(attr->attr)), ntohs(attr->attr));
			break;
		}
		/* Clear attribute in case previous entry was a string */
		attr->attr = 0;
		data += ntohs(attr->len) + sizeof(struct stun_attr);
		len -= ntohs(attr->len) + sizeof(struct stun_attr);
	}
	/* Null terminate any string */
	*data = '\0';
	resp = (struct stun_header *)respdata;
	resplen = 0;
	respleft = sizeof(respdata) - sizeof(struct stun_header);
	resp->id = hdr->id;
	resp->msgtype = 0;
	resp->msglen = 0;
	attr = (struct stun_attr *)resp->ies;
	if (!len) {
		switch(ntohs(hdr->msgtype)) {
		case STUN_BINDREQ:
			if (stundebug)
				ast_verbose("STUN Bind Request, username: %s\n", 
					st.username ? st.username : "<none>");
			if (st.username)
				append_attr_string(&attr, STUN_USERNAME, st.username, &resplen, &respleft);
			append_attr_address(&attr, STUN_MAPPED_ADDRESS, src, &resplen, &respleft);
			resp->msglen = htons(resplen);
			resp->msgtype = htons(STUN_BINDRESP);
			stun_send(s, src, resp);
			ret = STUN_ACCEPT;
			break;
		default:
			if (stundebug)
				ast_verbose("Dunno what to do with STUN message %04x (%s)\n", ntohs(hdr->msgtype), stun_msg2str(ntohs(hdr->msgtype)));
		}
	}
	return ret;
}

/*! \brief List of current sessions */
static AST_LIST_HEAD_STATIC(protos, ast_rtp_protocol);

static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw)
{
	unsigned int sec, usec, frac;
	sec = tv.tv_sec + 2208988800u; /* Sec between 1900 and 1970 */
	usec = tv.tv_usec;
	frac = (usec << 12) + (usec << 8) - ((usec * 3650) >> 6);
	*msw = sec;
	*lsw = frac;
}

int ast_rtp_fd(struct ast_rtp *rtp)
{
	return rtp->s;
}

int ast_rtcp_fd(struct ast_rtp *rtp)
{
	if (rtp->rtcp)
		return rtp->rtcp->s;
	return -1;
}

unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
{
	unsigned int interval;
	/*! \todo XXX Do a more reasonable calculation on this one
	* Look in RFC 3550 Section A.7 for an example*/
	interval = rtcpinterval;
	return interval;
}

void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
{
	rtp->data = data;
}

void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback)
{
	rtp->callback = callback;
}

void ast_rtp_setnat(struct ast_rtp *rtp, int nat)
{
	rtp->nat = nat;
}

void ast_rtp_setdtmf(struct ast_rtp *rtp, int dtmf)
{
	ast_set2_flag(rtp, dtmf ? 1 : 0, FLAG_HAS_DTMF);
}

static struct ast_frame *send_dtmf(struct ast_rtp *rtp)
{
	if (ast_tvcmp(ast_tvnow(), rtp->dtmfmute) < 0) {
		if (option_debug)
			ast_log(LOG_DEBUG, "Ignore potential DTMF echo from '%s'\n", ast_inet_ntoa(rtp->them.sin_addr));
		rtp->resp = 0;
		rtp->dtmfduration = 0;
		return &ast_null_frame;
	}
	if (option_debug)
		ast_log(LOG_DEBUG, "Sending dtmf: %d (%c), at %s\n", rtp->resp, rtp->resp, ast_inet_ntoa(rtp->them.sin_addr));
	if (rtp->resp == 'X') {
		rtp->f.frametype = AST_FRAME_CONTROL;
		rtp->f.subclass = AST_CONTROL_FLASH;
	} else {
		rtp->f.frametype = AST_FRAME_DTMF;
		rtp->f.subclass = rtp->resp;
	}
	rtp->f.datalen = 0;
	rtp->f.samples = 0;
	rtp->f.mallocd = 0;
	rtp->f.src = "RTP";
	rtp->resp = 0;
	rtp->dtmfduration = 0;
	return &rtp->f;
	
}

static inline int rtp_debug_test_addr(struct sockaddr_in *addr)
{
	if (rtpdebug == 0)
		return 0;
	if (rtpdebugaddr.sin_addr.s_addr) {
		if (((ntohs(rtpdebugaddr.sin_port) != 0)
			&& (rtpdebugaddr.sin_port != addr->sin_port))
			|| (rtpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
		return 0;
	}
	return 1;
}

static inline int rtcp_debug_test_addr(struct sockaddr_in *addr)
{
	if (rtcpdebug == 0)
		return 0;
	if (rtcpdebugaddr.sin_addr.s_addr) {
		if (((ntohs(rtcpdebugaddr.sin_port) != 0)
			&& (rtcpdebugaddr.sin_port != addr->sin_port))
			|| (rtcpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
		return 0;
	}
	return 1;
}


static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *data, int len)
{
	unsigned int event;
	char resp = 0;
	struct ast_frame *f = NULL;
	event = ntohl(*((unsigned int *)(data)));
	event &= 0x001F;
	if (option_debug > 2 || rtpdebug)
		ast_log(LOG_DEBUG, "Cisco DTMF Digit: %08x (len = %d)\n", event, len);
	if (event < 10) {
		resp = '0' + event;
	} else if (event < 11) {
		resp = '*';
	} else if (event < 12) {
		resp = '#';
	} else if (event < 16) {
		resp = 'A' + (event - 12);
	} else if (event < 17) {
		resp = 'X';
	}
	if (rtp->resp && (rtp->resp != resp)) {
		f = send_dtmf(rtp);
	}
	rtp->resp = resp;
	rtp->dtmfcount = dtmftimeout;
	return f;
}

/*! 
 * \brief Process RTP DTMF and events according to RFC 2833.
 * 
 * RFC 2833 is "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals".
 * 
 * \param rtp
 * \param data
 * \param len
 * \param seqno
 * \returns
 */
static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len, unsigned int seqno)
{
	unsigned int event;
	unsigned int event_end;
	unsigned int duration;
	char resp = 0;
	struct ast_frame *f = NULL;

	event = ntohl(*((unsigned int *)(data)));
	event >>= 24;
	event_end = ntohl(*((unsigned int *)(data)));
	event_end <<= 8;
	event_end >>= 24;
	duration = ntohl(*((unsigned int *)(data)));
	duration &= 0xFFFF;
	if (rtpdebug || option_debug > 2)
		ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
	if (event < 10) {
		resp = '0' + event;
	} else if (event < 11) {
		resp = '*';
	} else if (event < 12) {
		resp = '#';
	} else if (event < 16) {
		resp = 'A' + (event - 12);
	} else if (event < 17) {	/* Event 16: Hook flash */
		resp = 'X';	
	}
	if (rtp->resp && (rtp->resp != resp)) {
		f = send_dtmf(rtp);
	} else if (event_end & 0x80) {
		if (rtp->resp) {
			if (rtp->lasteventendseqn != seqno) {
				f = send_dtmf(rtp);
				rtp->lasteventendseqn = seqno;
			}
			rtp->resp = 0;
		}
		resp = 0;
		duration = 0;
	} else if (rtp->resp && rtp->dtmfduration && (duration < rtp->dtmfduration)) {
		f = send_dtmf(rtp);
	}
	if (!(event_end & 0x80))
		rtp->resp = resp;
	rtp->dtmfcount = dtmftimeout;
	rtp->dtmfduration = duration;
	return f;
}

/*!
 * \brief Process Comfort Noise RTP.
 * 
 * This is incomplete at the moment.
 * 
*/
static struct ast_frame *process_rfc3389(struct ast_rtp *rtp, unsigned char *data, int len)
{
	struct ast_frame *f = NULL;
	/* Convert comfort noise into audio with various codecs.  Unfortunately this doesn't
	   totally help us out becuase we don't have an engine to keep it going and we are not
	   guaranteed to have it every 20ms or anything */
	if (rtpdebug)
		ast_log(LOG_DEBUG, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", rtp->lastrxformat, len);

	if (!(ast_test_flag(rtp, FLAG_3389_WARNING))) {
		ast_log(LOG_NOTICE, "Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: %s\n",
			ast_inet_ntoa(rtp->them.sin_addr));
		ast_set_flag(rtp, FLAG_3389_WARNING);
	}

	/* Must have at least one byte */
	if (!len)
		return NULL;
	if (len < 24) {
		rtp->f.data = rtp->rawdata + AST_FRIENDLY_OFFSET;
		rtp->f.datalen = len - 1;
		rtp->f.offset = AST_FRIENDLY_OFFSET;
		memcpy(rtp->f.data, data + 1, len - 1);
	} else {
		rtp->f.data = NULL;
		rtp->f.offset = 0;
		rtp->f.datalen = 0;
	}
	rtp->f.frametype = AST_FRAME_CNG;
	rtp->f.subclass = data[0] & 0x7f;
	rtp->f.datalen = len - 1;
	rtp->f.samples = 0;
	rtp->f.delivery.tv_usec = rtp->f.delivery.tv_sec = 0;
	f = &rtp->f;
	return f;
}

static int rtpread(int *id, int fd, short events, void *cbdata)
{
	struct ast_rtp *rtp = cbdata;
	struct ast_frame *f;
	f = ast_rtp_read(rtp);
	if (f) {
		if (rtp->callback)
			rtp->callback(rtp, f, rtp->data);
	}
	return 1;
}

struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp)
{
	socklen_t len;
	int position, i, packetwords;
	int res;
	struct sockaddr_in sin;
	unsigned int rtcpdata[8192 + AST_FRIENDLY_OFFSET];
	unsigned int *rtcpheader;
	int pt;
	struct timeval now;
	unsigned int length;
	int rc;
	double rtt = 0;
	double a;
	double dlsr;
	double lsr;
	unsigned int msw;
	unsigned int lsw;
	unsigned int comp;
	struct ast_frame *f = &ast_null_frame;
	
	if (!rtp || !rtp->rtcp)
		return &ast_null_frame;

	len = sizeof(sin);
	
	res = recvfrom(rtp->rtcp->s, rtcpdata + AST_FRIENDLY_OFFSET, sizeof(rtcpdata) - sizeof(unsigned int) * AST_FRIENDLY_OFFSET,
					0, (struct sockaddr *)&sin, &len);
	rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
	
	if (res < 0) {
		if (errno != EAGAIN)
			ast_log(LOG_WARNING, "RTCP Read error: %s\n", strerror(errno));
		if (errno == EBADF)
			CRASH;
		return &ast_null_frame;
	}

	packetwords = res / 4;
	
	if (rtp->nat) {
		/* Send to whoever sent to us */
		if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
		    (rtp->rtcp->them.sin_port != sin.sin_port)) {
			memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
			if (option_debug || rtpdebug)
				ast_log(LOG_DEBUG, "RTCP NAT: Got RTCP from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
		}
	}
	if (option_debug)
		ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);

	/* Process a compound packet */
	position = 0;
	while (position < packetwords) {
		i = position;
		length = ntohl(rtcpheader[i]);
		pt = (length & 0xff0000) >> 16;
		rc = (length & 0x1f000000) >> 24;
		length &= 0xffff;
    
		if ((i + length) > packetwords) {
			ast_log(LOG_WARNING, "RTCP Read too short\n");
			return &ast_null_frame;
		}
		
		if (rtcp_debug_test_addr(&sin)) {
		  	ast_verbose("\n\nGot RTCP from %s:%d\n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port));
		  	ast_verbose("PT: %d(%s)\n", pt, (pt == 200) ? "Sender Report" : (pt == 201) ? "Receiver Report" : (pt == 192) ? "H.261 FUR" : "Unknown");
		  	ast_verbose("Reception reports: %d\n", rc);
		  	ast_verbose("SSRC of sender: %u\n", rtcpheader[i + 1]);
		}
    
		i += 2; /* Advance past header and ssrc */
		
		switch (pt) {
		case RTCP_PT_SR:
			gettimeofday(&rtp->rtcp->rxlsr,NULL); /* To be able to populate the dlsr */
			rtp->rtcp->spc = ntohl(rtcpheader[i+3]);
			rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
			rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff) >> 16); /* Going to LSR in RR*/
    
			if (rtcp_debug_test_addr(&sin)) {
				ast_verbose("NTP timestamp: %lu.%010lu\n", (unsigned long) ntohl(rtcpheader[i]), (unsigned long) ntohl(rtcpheader[i + 1]) * 4096);
				ast_verbose("RTP timestamp: %lu\n", (unsigned long) ntohl(rtcpheader[i + 2]));
				ast_verbose("SPC: %lu\tSOC: %lu\n", (unsigned long) ntohl(rtcpheader[i + 3]), (unsigned long) ntohl(rtcpheader[i + 4]));
			}
			i += 5;
			if (rc < 1)
				break;
			/* Intentional fall through */
		case RTCP_PT_RR:
			/* This is the place to calculate RTT */
			/* Don't handle multiple reception reports (rc > 1) yet */
			gettimeofday(&now, NULL);
			timeval2ntp(now, &msw, &lsw);
			/* Use the one we sent them in our SR instead, rtcp->txlsr could have been rewritten if the dlsr is large */
			if (ntohl(rtcpheader[i + 4])) { /* We must have the LSR */
				comp = ((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16);
				a = (double)((comp & 0xffff0000) >> 16) + (double)((double)(comp & 0xffff)/1000000.);
				lsr = (double)((ntohl(rtcpheader[i + 4]) & 0xffff0000) >> 16) + (double)((double)(ntohl(rtcpheader[i + 4]) & 0xffff) / 1000000.);
				dlsr = (double)(ntohl(rtcpheader[i + 5])/65536.);
				rtt = a - dlsr - lsr;
				rtp->rtcp->accumulated_transit += rtt;
				rtp->rtcp->rtt = rtt;
				if (rtp->rtcp->maxrtt<rtt)
					rtp->rtcp->maxrtt = rtt;
				if (rtp->rtcp->minrtt>rtt)
				rtp->rtcp->minrtt = rtt;
			}
			rtp->rtcp->reported_jitter = ntohl(rtcpheader[i + 3]);
			rtp->rtcp->reported_lost = ntohl(rtcpheader[i + 1]) & 0xffffff;
			if (rtcp_debug_test_addr(&sin)) {
				ast_verbose("Fraction lost: %ld\n", (((long) ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24));
				ast_verbose("Packets lost so far: %d\n", rtp->rtcp->reported_lost);
				ast_verbose("Highest sequence number: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff));
				ast_verbose("Sequence number cycles: %ld\n", (long) (ntohl(rtcpheader[i + 2]) & 0xffff) >> 16);
				ast_verbose("Interarrival jitter: %u\n", rtp->rtcp->reported_jitter);
				ast_verbose("Last SR(our NTP): %lu.%010lu\n",(unsigned long) ntohl(rtcpheader[i + 4]) >> 16,((unsigned long) ntohl(rtcpheader[i + 4]) << 16) * 4096);
				ast_verbose("DLSR: %4.4f (sec)\n",ntohl(rtcpheader[i + 5])/65536.0);
				if (rtt)
					ast_verbose("RTT: %f(sec)\n", rtt);
			}
			break;
		case RTCP_PT_FUR:
			if (rtcp_debug_test_addr(&sin))
				ast_verbose("Received an RTCP Fast Update Request\n");
			rtp->f.frametype = AST_FRAME_CONTROL;
			rtp->f.subclass = AST_CONTROL_VIDUPDATE;
			rtp->f.datalen = 0;
			rtp->f.samples = 0;
			rtp->f.mallocd = 0;
			rtp->f.src = "RTP";
			f = &rtp->f;
			break;
		case RTCP_PT_SDES:
			if (rtcp_debug_test_addr(&sin))
				ast_verbose("Received an SDES from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
			break;
		case RTCP_PT_BYE:
			if (rtcp_debug_test_addr(&sin))
				ast_verbose("Received a BYE from %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
			break;
		default:
			ast_log(LOG_NOTICE, "Unknown RTCP packet (pt=%d) received from %s:%d\n", pt, ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
			break;
		}
		position += (length + 1);
	}
			
	return f;
}

static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int timestamp, int mark)
{
	struct timeval now;
	double transit;
	double current_time;
	double d;
	double dtv;
	double prog;
	
	if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
		gettimeofday(&rtp->rxcore, NULL);
		rtp->drxcore = (double) rtp->rxcore.tv_sec + (double) rtp->rxcore.tv_usec / 1000000;
		/* map timestamp to a real time */
		rtp->seedrxts = timestamp; /* Their RTP timestamp started with this */
		rtp->rxcore.tv_sec -= timestamp / 8000;
		rtp->rxcore.tv_usec -= (timestamp % 8000) * 125;
		/* Round to 0.1ms for nice, pretty timestamps */
		rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100;
		if (rtp->rxcore.tv_usec < 0) {
			/* Adjust appropriately if necessary */
			rtp->rxcore.tv_usec += 1000000;
			rtp->rxcore.tv_sec -= 1;
		}
	}

	gettimeofday(&now,NULL);
	/* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */
	tv->tv_sec = rtp->rxcore.tv_sec + timestamp / 8000;
	tv->tv_usec = rtp->rxcore.tv_usec + (timestamp % 8000) * 125;
	if (tv->tv_usec >= 1000000) {
		tv->tv_usec -= 1000000;
		tv->tv_sec += 1;
	}
	prog = (double)((timestamp-rtp->seedrxts)/8000.);
	dtv = (double)rtp->drxcore + (double)(prog);
	current_time = (double)now.tv_sec + (double)now.tv_usec/1000000;
	transit = current_time - dtv;
	d = transit - rtp->rxtransit;
	rtp->rxtransit = transit;
	if (d<0)
		d=-d;
	rtp->rxjitter += (1./16.) * (d - rtp->rxjitter);
	if (rtp->rxjitter > rtp->rtcp->maxrxjitter)
		rtp->rtcp->maxrxjitter = rtp->rxjitter;
	if (rtp->rxjitter < rtp->rtcp->minrxjitter)
		rtp->rtcp->minrxjitter = rtp->rxjitter;
}

struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
{
	int res;
	struct sockaddr_in sin;
	socklen_t len;
	unsigned int seqno;
	int version;
	int payloadtype;
	int tseqno;
	int hdrlen = 12;
	int padding;
	int mark;
	int ext;
	unsigned int ssrc;
	unsigned int timestamp;
	unsigned int *rtpheader;
	struct rtpPayloadType rtpPT;
	
	len = sizeof(sin);
	
	/* Cache where the header will go */
	res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
					0, (struct sockaddr *)&sin, &len);

	rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
	if (res < 0) {
		if (errno != EAGAIN)
			ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
		if (errno == EBADF)
			CRASH;
		return &ast_null_frame;
	}
	
	if (res < hdrlen) {
		ast_log(LOG_WARNING, "RTP Read too short\n");
		return &ast_null_frame;
	}

	/* Get fields */
	seqno = ntohl(rtpheader[0]);

	/* Check RTP version */
	version = (seqno & 0xC0000000) >> 30;
	if (!version) {
		if ((stun_handle_packet(rtp->s, &sin, rtp->rawdata + AST_FRIENDLY_OFFSET, res) == STUN_ACCEPT) &&
			(!rtp->them.sin_port && !rtp->them.sin_addr.s_addr)) {
			memcpy(&rtp->them, &sin, sizeof(rtp->them));
		}
		return &ast_null_frame;
	}

	if (version != 2)
		return &ast_null_frame;
	/* Ignore if the other side hasn't been given an address
	   yet.  */
	if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
		return &ast_null_frame;

	if (rtp->nat) {
		/* Send to whoever sent to us */
		if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
		    (rtp->them.sin_port != sin.sin_port)) {
			rtp->them = sin;
			if (rtp->rtcp) {
				memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
				rtp->rtcp->them.sin_port = htons(ntohs(rtp->them.sin_port)+1);
			}
			rtp->rxseqno = 0;
			ast_set_flag(rtp, FLAG_NAT_ACTIVE);
			if (option_debug || rtpdebug)
				ast_log(LOG_DEBUG, "RTP NAT: Got audio from other end. Now sending to address %s:%d\n", ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
		}
	}

	payloadtype = (seqno & 0x7f0000) >> 16;
	padding = seqno & (1 << 29);
	mark = seqno & (1 << 23);
	ext = seqno & (1 << 28);
	seqno &= 0xffff;
	timestamp = ntohl(rtpheader[1]);
	ssrc = ntohl(rtpheader[2]);
	
	if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
		if (option_debug || rtpdebug)
			ast_log(LOG_DEBUG, "Forcing Marker bit, because SSRC has changed\n");
		mark = 1;
	}

	rtp->rxssrc = ssrc;
	
	if (padding) {
		/* Remove padding bytes */
		res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
	}
	
	if (ext) {
		/* RTP Extension present */
		hdrlen += 4;
		hdrlen += (ntohl(rtpheader[3]) & 0xffff) << 2;
	}

	if (res < hdrlen) {
		ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
		return &ast_null_frame;
	}

	rtp->rxcount++; /* Only count reasonably valid packets, this'll make the rtcp stats more accurate */

	tseqno = rtp->lastrxseqno +1;

	if (rtp->rxcount==1) {
		/* This is the first RTP packet successfully received from source */
		rtp->seedrxseqno = seqno;
	}

	if (rtp->rtcp && rtp->rtcp->schedid < 1) {
		/* Schedule transmission of Receiver Report */
		rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
	}

	if (tseqno > RTP_SEQ_MOD) { /* if tseqno is greater than RTP_SEQ_MOD it would indicate that the sender cycled */
		rtp->cycles += RTP_SEQ_MOD;
		ast_verbose("SEQNO cycled: %u\t%d\n", rtp->cycles, seqno);
	}

	rtp->lastrxseqno = seqno;
	
	if (rtp->themssrc==0)
		rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
	
	if (rtp_debug_test_addr(&sin))
		ast_verbose("Got  RTP packet from    %s:%d (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
			ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);

	rtpPT = ast_rtp_lookup_pt(rtp, payloadtype);
	if (!rtpPT.isAstFormat) {
		struct ast_frame *f = NULL;

		/* This is special in-band data that's not one of our codecs */
		if (rtpPT.code == AST_RTP_DTMF) {
			/* It's special -- rfc2833 process it */
			if (rtp_debug_test_addr(&sin)) {
				unsigned char *data;
				unsigned int event;
				unsigned int event_end;
				unsigned int duration;
				data = rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen;
				event = ntohl(*((unsigned int *)(data)));
				event >>= 24;
				event_end = ntohl(*((unsigned int *)(data)));
				event_end <<= 8;
				event_end >>= 24;
				duration = ntohl(*((unsigned int *)(data)));
				duration &= 0xFFFF;
				ast_verbose("Got  RTP RFC2833 from   %s:%d (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u, mark %d, event %08x, end %d, duration %-5.5d) \n", ast_inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp, res - hdrlen, (mark?1:0), event, ((event_end & 0x80)?1:0), duration);
			}
			if (rtp->lasteventseqn <= seqno || rtp->resp == 0 || (rtp->lasteventseqn >= 65530 && seqno <= 6)) {
				f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno);
				rtp->lasteventseqn = seqno;
			}
		} else if (rtpPT.code == AST_RTP_CISCO_DTMF) {
			/* It's really special -- process it the Cisco way */
			if (rtp->lasteventseqn <= seqno || rtp->resp == 0 || (rtp->lasteventseqn >= 65530 && seqno <= 6)) {
				f = process_cisco_dtmf(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
				rtp->lasteventseqn = seqno;
			}
		} else if (rtpPT.code == AST_RTP_CN) {
			/* Comfort Noise */
			f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
		} else {
			ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n", payloadtype, ast_inet_ntoa(rtp->them.sin_addr));
		}
		return f ? f : &ast_null_frame;
	}
	rtp->lastrxformat = rtp->f.subclass = rtpPT.code;
	rtp->f.frametype = (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) ? AST_FRAME_VOICE : AST_FRAME_VIDEO;

	if (!rtp->lastrxts)
		rtp->lastrxts = timestamp;

	rtp->rxseqno = seqno;

	if (rtp->dtmfcount) {
#if 0
		printf("dtmfcount was %d\n", rtp->dtmfcount);
#endif		
		rtp->dtmfcount -= (timestamp - rtp->lastrxts);
		if (rtp->dtmfcount < 0)
			rtp->dtmfcount = 0;
#if 0
		if (dtmftimeout != rtp->dtmfcount)
			printf("dtmfcount is %d\n", rtp->dtmfcount);
#endif
	}
	rtp->lastrxts = timestamp;

	/* Send any pending DTMF */
	if (rtp->resp && !rtp->dtmfcount) {
		if (option_debug)
			ast_log(LOG_DEBUG, "Sending pending DTMF\n");
		return send_dtmf(rtp);
	}
	rtp->f.mallocd = 0;
	rtp->f.datalen = res - hdrlen;
	rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
	rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
	if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO) {
		rtp->f.samples = ast_codec_get_samples(&rtp->f);
		if (rtp->f.subclass == AST_FORMAT_SLINEAR) 
			ast_frame_byteswap_be(&rtp->f);
		calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
		/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
		rtp->f.has_timing_info = 1;
		rtp->f.ts = timestamp / 8;
		rtp->f.len = rtp->f.samples / 8;
		rtp->f.seqno = seqno;
	} else {
		/* Video -- samples is # of samples vs. 90000 */
		if (!rtp->lastividtimestamp)
			rtp->lastividtimestamp = timestamp;
		rtp->f.samples = timestamp - rtp->lastividtimestamp;
		rtp->lastividtimestamp = timestamp;
		rtp->f.delivery.tv_sec = 0;
		rtp->f.delivery.tv_usec = 0;
		if (mark)
			rtp->f.subclass |= 0x1;
		
	}
	rtp->f.src = "RTP";
	return &rtp->f;
}

/* The following array defines the MIME Media type (and subtype) for each
   of our codecs, or RTP-specific data type. */
static struct {
	struct rtpPayloadType payloadType;
	char* type;
	char* subtype;
} mimeTypes[] = {
	{{1, AST_FORMAT_G723_1}, "audio", "G723"},
	{{1, AST_FORMAT_GSM}, "audio", "GSM"},
	{{1, AST_FORMAT_ULAW}, "audio", "PCMU"},
	{{1, AST_FORMAT_ALAW}, "audio", "PCMA"},
	{{1, AST_FORMAT_G726}, "audio", "G726-32"},
	{{1, AST_FORMAT_ADPCM}, "audio", "DVI4"},
	{{1, AST_FORMAT_SLINEAR}, "audio", "L16"},
	{{1, AST_FORMAT_LPC10}, "audio", "LPC"},
	{{1, AST_FORMAT_G729A}, "audio", "G729"},
	{{1, AST_FORMAT_SPEEX}, "audio", "speex"},
	{{1, AST_FORMAT_ILBC}, "audio", "iLBC"},
	{{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32"},
	{{0, AST_RTP_DTMF}, "audio", "telephone-event"},
	{{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"},
	{{0, AST_RTP_CN}, "audio", "CN"},
	{{1, AST_FORMAT_JPEG}, "video", "JPEG"},
	{{1, AST_FORMAT_PNG}, "video", "PNG"},
	{{1, AST_FORMAT_H261}, "video", "H261"},
	{{1, AST_FORMAT_H263}, "video", "H263"},
	{{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998"},
	{{1, AST_FORMAT_H264}, "video", "H264"},
};

/* Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
   also, our own choices for dynamic payload types.  This is our master
   table for transmission */
static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
	[0] = {1, AST_FORMAT_ULAW},
#ifdef USE_DEPRECATED_G726
	[2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
#endif
	[3] = {1, AST_FORMAT_GSM},
	[4] = {1, AST_FORMAT_G723_1},
	[5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
	[6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
	[7] = {1, AST_FORMAT_LPC10},
	[8] = {1, AST_FORMAT_ALAW},
	[10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
	[11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
	[13] = {0, AST_RTP_CN},
	[16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
	[17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
	[18] = {1, AST_FORMAT_G729A},
	[19] = {0, AST_RTP_CN},		/* Also used for CN */
	[26] = {1, AST_FORMAT_JPEG},
	[31] = {1, AST_FORMAT_H261},
	[34] = {1, AST_FORMAT_H263},
	[103] = {1, AST_FORMAT_H263_PLUS},
	[97] = {1, AST_FORMAT_ILBC},
	[99] = {1, AST_FORMAT_H264},
	[101] = {0, AST_RTP_DTMF},
	[110] = {1, AST_FORMAT_SPEEX},
	[111] = {1, AST_FORMAT_G726},
	[112] = {1, AST_FORMAT_G726_AAL2},
	[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
};

void ast_rtp_pt_clear(struct ast_rtp* rtp) 
{
	int i;
	if (!rtp)
		return;

	for (i = 0; i < MAX_RTP_PT; ++i) {
		rtp->current_RTP_PT[i].isAstFormat = 0;
		rtp->current_RTP_PT[i].code = 0;
	}

	rtp->rtp_lookup_code_cache_isAstFormat = 0;
	rtp->rtp_lookup_code_cache_code = 0;
	rtp->rtp_lookup_code_cache_result = 0;
}

void ast_rtp_pt_default(struct ast_rtp* rtp) 
{
	int i;

	/* Initialize to default payload types */
	for (i = 0; i < MAX_RTP_PT; ++i) {
		rtp->current_RTP_PT[i].isAstFormat = static_RTP_PT[i].isAstFormat;
		rtp->current_RTP_PT[i].code = static_RTP_PT[i].code;
	}

	rtp->rtp_lookup_code_cache_isAstFormat = 0;
	rtp->rtp_lookup_code_cache_code = 0;
	rtp->rtp_lookup_code_cache_result = 0;
}

void ast_rtp_pt_copy(struct ast_rtp *dest, const struct ast_rtp *src)
{
	unsigned int i;

	for (i=0; i < MAX_RTP_PT; ++i) {
		dest->current_RTP_PT[i].isAstFormat = 
			src->current_RTP_PT[i].isAstFormat;
		dest->current_RTP_PT[i].code = 
			src->current_RTP_PT[i].code; 
	}
	dest->rtp_lookup_code_cache_isAstFormat = 0;
	dest->rtp_lookup_code_cache_code = 0;
	dest->rtp_lookup_code_cache_result = 0;
}

/*! \brief Get channel driver interface structure */
static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
{
	struct ast_rtp_protocol *cur = NULL;

	AST_LIST_LOCK(&protos);
	AST_LIST_TRAVERSE(&protos, cur, list) {
		if (cur->type == chan->tech->type)
			break;
	}
	AST_LIST_UNLOCK(&protos);

	return cur;
}

int ast_rtp_early_bridge(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 media early */
	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 setup early bridge 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 bridge 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 */
		if (destpr->set_rtp_peer(dest, srcp, vsrcp, srccodec, ast_test_flag(srcp, FLAG_NAT_ACTIVE)))
			ast_log(LOG_WARNING, "Channel '%s' failed to setup early bridge 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 payload 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
 * an SDP "a=rtpmap:" line.
 */
void ast_rtp_set_rtpmap_type(struct ast_rtp *rtp, int pt,
			     char *mimeType, char *mimeSubtype,
			     enum ast_rtp_options options)
{
	unsigned 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;
			if ((mimeTypes[i].payloadType.code == AST_FORMAT_G726) &&
			    mimeTypes[i].payloadType.isAstFormat &&
			    (options & AST_RTP_OPT_G726_NONSTANDARD))
				rtp->current_RTP_PT[pt].code = AST_FORMAT_G726_AAL2;
			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 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;
}

const char *ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code,
				  enum ast_rtp_options options)
{
	unsigned int i;

	for (i = 0; i < sizeof(mimeTypes)/sizeof(mimeTypes[0]); ++i) {
		if ((mimeTypes[i].payloadType.code == code) && (mimeTypes[i].payloadType.isAstFormat == isAstFormat)) {
			if (isAstFormat &&
			    (code == AST_FORMAT_G726_AAL2) &&
			    (options & AST_RTP_OPT_G726_NONSTANDARD))
				return "AAL2-G726-32";
			else
				return mimeTypes[i].subtype;
		}
	}

	return "";
}

char *ast_rtp_lookup_mime_multiple(char *buf, size_t size, const int capability,
				   const int isAstFormat, enum ast_rtp_options options)
{
	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, options);

			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;
	rtcp->them.sin_family = AF_INET;

	if (rtcp->s < 0) {
		free(rtcp);
		ast_log(LOG_WARNING, "Unable to allocate RTCP 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;
	ast_set_flag(rtp, FLAG_HAS_DTMF);
	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)
{
	if (rtp->rtcp && rtp->rtcp->schedid > 0) {
		ast_sched_del(rtp->sched, rtp->rtcp->schedid);
		rtp->rtcp->schedid = -1;
	}

	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->rtcp->them.sin_addr));
		memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->rtcp->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;
}

char *ast_rtp_get_quality(struct ast_rtp *rtp)
{
	/*
	*ssrc          our ssrc
	*themssrc      their ssrc
	*lp            lost packets
	*rxjitter      our calculated jitter(rx)
	*rxcount       no. received packets
	*txjitter      reported jitter of the other end
	*txcount       transmitted packets
	*rlp           remote lost packets
	*/
	
	snprintf(rtp->rtcp->quality, sizeof(rtp->rtcp->quality), "ssrc=%u;themssrc=%u;lp=%u;rxjitter=%f;rxcount=%u;txjitter=%f;txcount=%u;rlp=%u;rtt=%f", rtp->ssrc, rtp->themssrc, rtp->rtcp->expected_prior - rtp->rtcp->received_prior, rtp->rxjitter, rtp->rxcount, (double)rtp->rtcp->reported_jitter/65536., rtp->txcount, rtp->rtcp->reported_lost, rtp->rtcp->rtt);
	
	return rtp->rtcp->quality;
}

void ast_rtp_destroy(struct ast_rtp *rtp)
{
	if (rtcp_debug_test_addr(&rtp->them) || rtcpstats) {
		/*Print some info on the call here */
		ast_verbose("  RTP-stats\n");
		ast_verbose("* Our Receiver:\n");
		ast_verbose("  SSRC:		 %u\n", rtp->themssrc);
		ast_verbose("  Received packets: %u\n", rtp->rxcount);
		ast_verbose("  Lost packets:	 %u\n", rtp->rtcp->expected_prior - rtp->rtcp->received_prior);
		ast_verbose("  Jitter:		 %.4f\n", rtp->rxjitter);
		ast_verbose("  Transit:		 %.4f\n", rtp->rxtransit);
		ast_verbose("  RR-count:	 %u\n", rtp->rtcp->rr_count);
		ast_verbose("* Our Sender:\n");
		ast_verbose("  SSRC:		 %u\n", rtp->ssrc);
		ast_verbose("  Sent packets:	 %u\n", rtp->txcount);
		ast_verbose("  Lost packets:	 %u\n", rtp->rtcp->reported_lost);
		ast_verbose("  Jitter:		 %u\n", rtp->rtcp->reported_jitter);
		ast_verbose("  SR-count:	 %u\n", rtp->rtcp->sr_count);
		ast_verbose("  RTT:		 %f\n", rtp->rtcp->rtt);
	}

	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) {
		if (rtp->rtcp->schedid > 0)
			ast_sched_del(rtp->sched, rtp->rtcp->schedid);
		close(rtp->rtcp->s);
		free(rtp->rtcp);
		rtp->rtcp=NULL;
	}
	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];

	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(rtp->them.sin_addr),
					ntohs(rtp->them.sin_port), strerror(errno));
			if (rtp_debug_test_addr(&rtp->them))
				ast_verbose("Sent RTP DTMF packet to %s:%d (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
					    ast_inet_ntoa(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));
		}
	}
	/*! \note 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;
}

/* \brief Public function: Send an H.261 fast update request, some devices need this rather than SIP XML */
int ast_rtcp_send_h261fur(void *data)
{
	struct ast_rtp *rtp = data;
	int res;

	rtp->rtcp->sendfur = 1;
	res = ast_rtcp_write(data);
	
	return res;
}

/*! \brief Send RTCP sender's report */
static int ast_rtcp_write_sr(void *data)
{
	struct ast_rtp *rtp = data;
	int res;
	int len = 0;
	struct timeval now;
	unsigned int now_lsw;
	unsigned int now_msw;
	unsigned int *rtcpheader;
	unsigned int lost;
	unsigned int extended;
	unsigned int expected;
	unsigned int expected_interval;
	unsigned int received_interval;
	int lost_interval;
	int fraction;
	struct timeval dlsr;
	char bdata[512];

	if (!rtp || !rtp->rtcp || (&rtp->rtcp->them.sin_addr == 0))
		return 0;
	
	if (!rtp->rtcp->them.sin_addr.s_addr) {  /* This'll stop rtcp for this rtp session */
		ast_verbose("RTCP SR transmission error, rtcp halted %s\n",strerror(errno));
		if (rtp->rtcp->schedid > 0)
			ast_sched_del(rtp->sched, rtp->rtcp->schedid);
		rtp->rtcp->schedid = -1;
		return 0;
	}

	gettimeofday(&now, NULL);
	timeval2ntp(now, &now_msw, &now_lsw); /* fill thses ones in from utils.c*/
	rtcpheader = (unsigned int *)bdata;
	rtcpheader[1] = htonl(rtp->ssrc);               /* Our SSRC */
	rtcpheader[2] = htonl(now_msw);                 /* now, MSW. gettimeofday() + SEC_BETWEEN_1900_AND_1970*/
	rtcpheader[3] = htonl(now_lsw);                 /* now, LSW */
	rtcpheader[4] = htonl(rtp->lastts);             /* FIXME shouldn't be that, it should be now */
	rtcpheader[5] = htonl(rtp->txcount);            /* No. packets sent */
	rtcpheader[6] = htonl(rtp->txoctetcount);       /* No. bytes sent */
	len += 28;
	
	extended = rtp->cycles + rtp->lastrxseqno;
	expected = extended - rtp->seedrxseqno + 1;
	if (rtp->rxcount > expected) 
		expected += rtp->rxcount - expected;
	lost = expected - rtp->rxcount;
	expected_interval = expected - rtp->rtcp->expected_prior;
	rtp->rtcp->expected_prior = expected;
	received_interval = rtp->rxcount - rtp->rtcp->received_prior;
	rtp->rtcp->received_prior = rtp->rxcount;
	lost_interval = expected_interval - received_interval;
	if (expected_interval == 0 || lost_interval <= 0)
		fraction = 0;
	else
		fraction = (lost_interval << 8) / expected_interval;
	timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
	rtcpheader[7] = htonl(rtp->themssrc);
	rtcpheader[8] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
	rtcpheader[9] = htonl((rtp->cycles) | ((rtp->lastrxseqno & 0xffff)));
	rtcpheader[10] = htonl((unsigned int)rtp->rxjitter);
	rtcpheader[11] = htonl(rtp->rtcp->themrxlsr);
	rtcpheader[12] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);
	len += 24;
	
	rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SR << 16) | ((len/4)-1));

	if (rtp->rtcp->sendfur) {
		rtcpheader[13] = htonl((2 << 30) | (0 << 24) | (RTCP_PT_FUR << 16) | 1);
		rtcpheader[14] = htonl(rtp->ssrc);               /* Our SSRC */
		len += 8;
		rtp->rtcp->sendfur = 0;
	}
	
	/* Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos */ 
	/* it can change mid call, and SDES can't) */
	rtcpheader[len/4]     = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
	rtcpheader[(len/4)+1] = htonl(rtp->ssrc);               /* Our SSRC */
	rtcpheader[(len/4)+2] = htonl(0x01 << 24);                    /* Empty for the moment */
	len += 12;
	
	res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));
	if (res < 0) {
		ast_log(LOG_ERROR, "RTCP SR transmission error to %s:%d, rtcp halted %s\n",ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port), strerror(errno));
		if (rtp->rtcp->schedid > 0)
			ast_sched_del(rtp->sched, rtp->rtcp->schedid);
		rtp->rtcp->schedid = -1;
		return 0;
	}
	
	/* FIXME Don't need to get a new one */
	gettimeofday(&rtp->rtcp->txlsr, NULL);
	rtp->rtcp->sr_count++;

	rtp->rtcp->lastsrtxcount = rtp->txcount;	
	
	if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
		ast_verbose("* Sent RTCP SR to %s:%d\n", ast_inet_ntoa(rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
		ast_verbose("  Our SSRC: %u\n", rtp->ssrc);
		ast_verbose("  Sent(NTP): %u.%010u\n", (unsigned int)now.tv_sec, (unsigned int)now.tv_usec*4096);
		ast_verbose("  Sent(RTP): %u\n", rtp->lastts);
		ast_verbose("  Sent packets: %u\n", rtp->txcount);
		ast_verbose("  Sent octets: %u\n", rtp->txoctetcount);
		ast_verbose("  Report block:\n");
		ast_verbose("  Fraction lost: %u\n", fraction);
		ast_verbose("  Cumulative loss: %u\n", lost);
		ast_verbose("  IA jitter: %.4f\n", rtp->rxjitter);
		ast_verbose("  Their last SR: %u\n", rtp->rtcp->themrxlsr);
		ast_verbose("  DLSR: %4.4f (sec)\n\n", (double)(ntohl(rtcpheader[12])/65536.0));
	}
	return res;
}

/*! \brief Send RTCP recepient's report */
static int ast_rtcp_write_rr(void *data)
{
	struct ast_rtp *rtp = data;
	int res;
	int len = 32;
	unsigned int lost;
	unsigned int extended;
	unsigned int expected;
	unsigned int expected_interval;
	unsigned int received_interval;
	int lost_interval;
	struct timeval now;
	unsigned int *rtcpheader;
	char bdata[1024];
	struct timeval dlsr;
	int fraction;

	if (!rtp || !rtp->rtcp || (&rtp->rtcp->them.sin_addr == 0))
		return 0;
	  
	if (!rtp->rtcp->them.sin_addr.s_addr) {
		ast_log(LOG_ERROR, "RTCP RR transmission error to, rtcp halted %s\n",strerror(errno));
		if (rtp->rtcp->schedid > 0)
			ast_sched_del(rtp->sched, rtp->rtcp->schedid);
		rtp->rtcp->schedid = -1;
		return 0;
	}

	extended = rtp->cycles + rtp->lastrxseqno;
	expected = extended - rtp->seedrxseqno + 1;
	lost = expected - rtp->rxcount;
	expected_interval = expected - rtp->rtcp->expected_prior;
	rtp->rtcp->expected_prior = expected;
	received_interval = rtp->rxcount - rtp->rtcp->received_prior;
	rtp->rtcp->received_prior = rtp->rxcount;
	lost_interval = expected_interval - received_interval;
	if (expected_interval == 0 || lost_interval <= 0)
		fraction = 0;
	else
		fraction = (lost_interval << 8) / expected_interval;
	gettimeofday(&now, NULL);
	timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
	rtcpheader = (unsigned int *)bdata;
	rtcpheader[0] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_RR << 16) | ((len/4)-1));
	rtcpheader[1] = htonl(rtp->ssrc);
	rtcpheader[2] = htonl(rtp->themssrc);
	rtcpheader[3] = htonl(((fraction & 0xff) << 24) | (lost & 0xffffff));
	rtcpheader[4] = htonl((rtp->cycles) | ((rtp->lastrxseqno & 0xffff)));
	rtcpheader[5] = htonl((unsigned int)rtp->rxjitter);
	rtcpheader[6] = htonl(rtp->rtcp->themrxlsr);
	rtcpheader[7] = htonl((((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000);

	if (rtp->rtcp->sendfur) {
		rtcpheader[8] = htonl((2 << 30) | (0 << 24) | (RTCP_PT_FUR << 16) | 1); /* Header from page 36 in RFC 3550 */
		rtcpheader[9] = htonl(rtp->ssrc);               /* Our SSRC */
		len += 8;
		rtp->rtcp->sendfur = 0;
	}

	/*! \note Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos 
	it can change mid call, and SDES can't) */
	rtcpheader[len/4]     = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
	rtcpheader[(len/4)+1] = htonl(rtp->ssrc);               /* Our SSRC */
	rtcpheader[(len/4)+2] = htonl(0x01 << 24);              /* Empty for the moment */
	len += 12;
	
	res = sendto(rtp->rtcp->s, (unsigned int *)rtcpheader, len, 0, (struct sockaddr *)&rtp->rtcp->them, sizeof(rtp->rtcp->them));

	if (res < 0) {
		ast_log(LOG_ERROR, "RTCP RR transmission error, rtcp halted: %s\n",strerror(errno));
		/* Remove the scheduler */
		if (rtp->rtcp->schedid > 0)
			ast_sched_del(rtp->sched, rtp->rtcp->schedid);
		rtp->rtcp->schedid = -1;
		return 0;
	}

	rtp->rtcp->rr_count++;

	if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
		ast_verbose("\n* Sending RTCP RR to %s:%d\n"
			"  Our SSRC: %u\nTheir SSRC: %u\niFraction lost: %d\nCumulative loss: %u\n" 
			"  IA jitter: %.4f\n" 
			"  Their last SR: %u\n" 
			"  DLSR: %4.4f (sec)\n\n",
			ast_inet_ntoa(rtp->rtcp->them.sin_addr),
			ntohs(rtp->rtcp->them.sin_port),
			rtp->ssrc, rtp->themssrc, fraction, lost,
			rtp->rxjitter,
			rtp->rtcp->themrxlsr,
			(double)(ntohl(rtcpheader[7])/65536.0));
	}

	return res;
}

/*! \brief Write and RTCP packet to the far end
 * \note Decide if we are going to send an SR (with Reception Block) or RR 
 * RR is sent if we have not sent any rtp packets in the previous interval */
static int ast_rtcp_write(void *data)
{
	struct ast_rtp *rtp = data;
	int res;
	
	if (rtp->txcount > rtp->rtcp->lastsrtxcount)
		res = ast_rtcp_write_sr(data);
	else
		res = ast_rtcp_write_rr(data);
	
	return res;
}

/*! \brief generate comfort noice (CNG) */
int ast_rtp_sendcng(struct ast_rtp *rtp, int level)
{
	unsigned int *rtpheader;
	int hdrlen = 12;
	int res;
	int payload;
	char data[256];
	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(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 %u, len %d)\n"
					, ast_inet_ntoa(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;
	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;

	if (f->has_timing_info)
		rtp->lastts = f->ts * 8;

	/* 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(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(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
				ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
			}
		} else {
			rtp->txcount++;
			rtp->txoctetcount +=(res - hdrlen);
			
			if (rtp->rtcp && rtp->rtcp->schedid < 1) 
			    rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, rtp);
		}
				
		if (rtp_debug_test_addr(&rtp->them))
			ast_verbose("Sent RTP packet to      %s:%d (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6u)\n",
					ast_inet_ntoa(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 and video\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:
	case AST_FORMAT_G726_AAL2:
		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;
	
	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));

	/* 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;
	}

	if (ast_test_flag(p0, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) {
		/* can't bridge, we are carrying DTMF for this channel and the bridge
		   needs it
		*/
		ast_channel_unlock(c0);
		ast_channel_unlock(c1);
		return AST_BRIDGE_FAILED_NOWARN;
	}

	if (ast_test_flag(p1, FLAG_HAS_DTMF) && (flags & AST_BRIDGE_DTMF_CHANNEL_1)) {
		/* can't bridge, we are carrying DTMF for this channel and the bridge
		   needs it
		*/
		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);
	/* External RTP Bridge up, now loop and see if something happes that force us to take the
		media back to Asterisk */
	cs[0] = c0;
	cs[1] = c1;
	cs[2] = NULL;
	oldcodec0 = codec0;
	oldcodec1 = codec1;
	for (;;) {
		/* Check if something changed... */
		if ((c0->tech_pvt != pvt0)  ||
			(c1->tech_pvt != pvt1) ||
			(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
				ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
				if (c0->tech_pvt == pvt0) {
					if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0)) 
						ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
				}
				if (c1->tech_pvt == pvt1) {
					if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0)) 
						ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
				}
				return AST_BRIDGE_RETRY;
		}
		/* Now check if they have changed address */
		ast_rtp_get_peer(p1, &t1);
		ast_rtp_get_peer(p0, &t0);
		if (pr0->get_codec)
			codec0 = pr0->get_codec(c0);
		if (pr1->get_codec)
			codec1 = pr1->get_codec(c1);
		if (vp1)
			ast_rtp_get_peer(vp1, &vt1);
		if (vp0)
			ast_rtp_get_peer(vp0, &vt0);
		if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1)) || (codec1 != oldcodec1)) {
			if (option_debug > 1) {
				ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n", 
					c1->name, ast_inet_ntoa(t1.sin_addr), ntohs(t1.sin_port), codec1);
				ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n", 
					c1->name, ast_inet_ntoa(vt1.sin_addr), ntohs(vt1.sin_port), codec1);
				ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", 
					c1->name, ast_inet_ntoa(ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
				ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", 
					c1->name, ast_inet_ntoa(vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
			}
			if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE))) 
				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));
			oldcodec1 = codec1;
		}
		if (inaddrcmp(&t0, &ac0) || (vp0 && inaddrcmp(&vt0, &vac0))) {
			if (option_debug) {
				ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n", 
					c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0);
				ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n", 
					c0->name, ast_inet_ntoa(ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
			}
			if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
				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));
			oldcodec0 = codec0;
		}
		who = ast_waitfor_n(cs, 2, &timeoutms);
		if (!who) {
			if (!timeoutms) 
				return AST_BRIDGE_RETRY;
			if (option_debug)
				ast_log(LOG_DEBUG, "Ooh, empty read...\n");
			/* check for hangup / whentohangup */
			if (ast_check_hangup(c0) || ast_check_hangup(c1))
				break;
			continue;
		}
		f = ast_read(who);
		other = (who == c0) ? c1 : c0; /* the other channel */
		if (!f || ((f->frametype == AST_FRAME_DTMF) &&
				   (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) || 
			       ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
			/* breaking out of the bridge. */
			*fo = f;
			*rc = who;
			if (option_debug)
				ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
			if ((c0->tech_pvt == pvt0)) {
				if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0)) 
					ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
			}
			if ((c1->tech_pvt == pvt1)) {
				if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0)) 
					ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
			}
			return AST_BRIDGE_COMPLETE;
		} else if ((f->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
			if ((f->subclass == AST_CONTROL_HOLD) || (f->subclass == AST_CONTROL_UNHOLD) ||
			    (f->subclass == AST_CONTROL_VIDUPDATE)) {
				ast_indicate(other, f->subclass);
				ast_frfree(f);
			} else {
				*fo = f;
				*rc = who;
				ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", f->subclass, who->name);
				return AST_BRIDGE_COMPLETE;
			}
		} 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 */
				ast_write(other, 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 AST_BRIDGE_FAILED;
}

static int rtp_do_debug_ip(int fd, int argc, char *argv[])
{
	struct hostent *hp;
	struct ast_hostent ahp;
	int port = 0;
	char *p, *arg;

	if (argc != 4)
		return RESULT_SHOWUSAGE;
	arg = argv[3];
	p = strstr(arg, ":");
	if (p) {
		*p = '\0';
		p++;
		port = atoi(p);
	}
	hp = ast_gethostbyname(arg, &ahp);
	if (hp == NULL)
		return RESULT_SHOWUSAGE;
	rtpdebugaddr.sin_family = AF_INET;
	memcpy(&rtpdebugaddr.sin_addr, hp->h_addr, sizeof(rtpdebugaddr.sin_addr));
	rtpdebugaddr.sin_port = htons(port);
	if (port == 0)
		ast_cli(fd, "RTP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtpdebugaddr.sin_addr));
	else
		ast_cli(fd, "RTP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtpdebugaddr.sin_addr), port);
	rtpdebug = 1;
	return RESULT_SUCCESS;
}

static int rtcp_do_debug_ip(int fd, int argc, char *argv[])
{
	struct hostent *hp;
	struct ast_hostent ahp;
	int port = 0;
	char *p, *arg;
	if (argc != 5)
		return RESULT_SHOWUSAGE;

	arg = argv[4];
	p = strstr(arg, ":");
	if (p) {
		*p = '\0';
		p++;
		port = atoi(p);
	}
	hp = ast_gethostbyname(arg, &ahp);
	if (hp == NULL)
		return RESULT_SHOWUSAGE;
	rtcpdebugaddr.sin_family = AF_INET;
	memcpy(&rtcpdebugaddr.sin_addr, hp->h_addr, sizeof(rtcpdebugaddr.sin_addr));
	rtcpdebugaddr.sin_port = htons(port);
	if (port == 0)
		ast_cli(fd, "RTCP Debugging Enabled for IP: %s\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr));
	else
		ast_cli(fd, "RTCP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(rtcpdebugaddr.sin_addr), port);
	rtcpdebug = 1;
	return RESULT_SUCCESS;
}

static int rtp_do_debug(int fd, int argc, char *argv[])
{
	if (argc != 2) {
		if (argc != 4)
			return RESULT_SHOWUSAGE;
		return rtp_do_debug_ip(fd, argc, argv);
	}
	rtpdebug = 1;
	memset(&rtpdebugaddr,0,sizeof(rtpdebugaddr));
	ast_cli(fd, "RTP Debugging Enabled\n");
	return RESULT_SUCCESS;
}
   
static int rtcp_do_debug(int fd, int argc, char *argv[]) {
	if (argc != 3) {
		if (argc != 5)
			return RESULT_SHOWUSAGE;
		return rtcp_do_debug_ip(fd, argc, argv);
	}
	rtcpdebug = 1;
	memset(&rtcpdebugaddr,0,sizeof(rtcpdebugaddr));
	ast_cli(fd, "RTCP Debugging Enabled\n");
	return RESULT_SUCCESS;
}

static int rtcp_do_stats(int fd, int argc, char *argv[]) {
	if (argc != 3) {
		return RESULT_SHOWUSAGE;
	}
	rtcpstats = 1;
	ast_cli(fd, "RTCP Stats Enabled\n");
	return RESULT_SUCCESS;
}

static int rtp_no_debug(int fd, int argc, char *argv[])
{
	if (argc != 3)
		return RESULT_SHOWUSAGE;
	rtpdebug = 0;
	ast_cli(fd,"RTP Debugging Disabled\n");
	return RESULT_SUCCESS;
}

static int rtcp_no_debug(int fd, int argc, char *argv[])
{
	if (argc != 4)
		return RESULT_SHOWUSAGE;
	rtcpdebug = 0;
	ast_cli(fd,"RTCP Debugging Disabled\n");
	return RESULT_SUCCESS;
}

static int rtcp_no_stats(int fd, int argc, char *argv[])
{
	if (argc != 4)
		return RESULT_SHOWUSAGE;
	rtcpstats = 0;
	ast_cli(fd,"RTCP Stats Disabled\n");
	return RESULT_SUCCESS;
}


static int stun_do_debug(int fd, int argc, char *argv[])
{
	if (argc != 2) {
		return RESULT_SHOWUSAGE;
	}
	stundebug = 1;
	ast_cli(fd, "STUN Debugging Enabled\n");
	return RESULT_SUCCESS;
}
   
static int stun_no_debug(int fd, int argc, char *argv[])
{
	if (argc != 3)
		return RESULT_SHOWUSAGE;
	stundebug = 0;
	ast_cli(fd,"STUN Debugging Disabled\n");
	return RESULT_SUCCESS;
}


static char debug_usage[] =
  "Usage: rtp debug [ip host[:port]]\n"
  "       Enable dumping of all RTP packets to and from host.\n";

static char no_debug_usage[] =
  "Usage: rtp no debug\n"
  "       Disable all RTP debugging\n";

static char stun_debug_usage[] =
  "Usage: stun debug\n"
  "       Enable STUN (Simple Traversal of UDP through NATs) debugging\n";

static char stun_no_debug_usage[] =
  "Usage: stun no debug\n"
  "       Disable STUN debugging\n";


static struct ast_cli_entry  cli_debug_ip =
{{ "rtp", "debug", "ip", NULL } , rtp_do_debug, "Enable RTP debugging on IP", debug_usage };

static struct ast_cli_entry  cli_debug =
{{ "rtp", "debug", NULL } , rtp_do_debug, "Enable RTP debugging", debug_usage };

static struct ast_cli_entry  cli_no_debug =
{{ "rtp", "no", "debug", NULL } , rtp_no_debug, "Disable RTP debugging", no_debug_usage };

static char rtcp_debug_usage[] =
  "Usage: rtp rtcp debug [ip host[:port]]\n"
  "       Enable dumping of all RTCP packets to and from host.\n";
  
static char rtcp_no_debug_usage[] =
  "Usage: rtp rtcp no debug\n"
  "       Disable all RTCP debugging\n";

static char rtcp_stats_usage[] =
  "Usage: rtp rtcp stats\n"
  "       Enable dumping of RTCP stats.\n";
  
static char rtcp_no_stats_usage[] =
  "Usage: rtp rtcp no stats\n"
  "       Disable all RTCP stats\n";

static struct ast_cli_entry  cli_debug_ip_rtcp =
{{ "rtp", "rtcp", "debug", "ip", NULL } , rtcp_do_debug, "Enable RTCP debugging on IP", rtcp_debug_usage };

static struct ast_cli_entry  cli_debug_rtcp =
{{ "rtp", "rtcp", "debug", NULL } , rtcp_do_debug, "Enable RTCP debugging", rtcp_debug_usage };

static struct ast_cli_entry  cli_no_debug_rtcp =
{{ "rtp", "rtcp", "no", "debug", NULL } , rtcp_no_debug, "Disable RTCP debugging", rtcp_no_debug_usage };

static struct ast_cli_entry  cli_stats_rtcp =
{{ "rtp", "rtcp", "stats", NULL } , rtcp_do_stats, "Enable RTCP stats", rtcp_stats_usage };

static struct ast_cli_entry  cli_no_stats_rtcp =
{{ "rtp", "rtcp", "no", "stats", NULL } , rtcp_no_stats, "Disable RTCP stats", rtcp_no_stats_usage };

static struct ast_cli_entry  cli_stun_debug =
{{ "stun", "debug", NULL } , stun_do_debug, "Enable STUN debugging", stun_debug_usage };

static struct ast_cli_entry  cli_stun_no_debug =
{{ "stun", "no", "debug", NULL } , stun_no_debug, "Disable STUN debugging", stun_no_debug_usage };

int ast_rtp_reload(void)
{
	struct ast_config *cfg;
	char *s;

	rtpstart = 5000;
	rtpend = 31000;
	dtmftimeout = DEFAULT_DTMF_TIMEOUT;
	cfg = ast_config_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;
		}
		if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
			rtcpinterval = atoi(s);
			if (rtcpinterval == 0)
				rtcpinterval = 0; /* Just so we're clear... it's zero */
			if (rtcpinterval < RTCP_MIN_INTERVALMS)
				rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
			if (rtcpinterval > RTCP_MAX_INTERVALMS)
				rtcpinterval = RTCP_MAX_INTERVALMS;
		}
		if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
#ifdef SO_NO_CHECK
			if (ast_false(s))
				nochecksums = 1;
			else
				nochecksums = 0;
#else
			if (ast_false(s))
				ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
#endif
		}
		if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
			dtmftimeout = atoi(s);
			if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
				ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
					dtmftimeout, DEFAULT_DTMF_TIMEOUT);
				dtmftimeout = DEFAULT_DTMF_TIMEOUT;
			};
		}
		ast_config_destroy(cfg);
	}
	if (rtpstart >= rtpend) {
		ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
		rtpstart = 5000;
		rtpend = 31000;
	}
	if (option_verbose > 1)
		ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
	return 0;
}

/*! \brief Initialize the RTP system in Asterisk */
void ast_rtp_init(void)
{
	ast_cli_register(&cli_debug);
	ast_cli_register(&cli_debug_ip);
	ast_cli_register(&cli_no_debug);

	ast_cli_register(&cli_debug_rtcp);
	ast_cli_register(&cli_debug_ip_rtcp);
	ast_cli_register(&cli_no_debug_rtcp);

	ast_cli_register(&cli_stats_rtcp);
	ast_cli_register(&cli_no_stats_rtcp);
	
	ast_cli_register(&cli_stun_debug);
	ast_cli_register(&cli_stun_no_debug);
	ast_rtp_reload();
}