merge new_loader_completion branch, including (at least):

  - restructured build tree and makefiles to eliminate recursion problems
  - support for embedded modules
  - support for static builds
  - simpler cross-compilation support
  - simpler module/loader interface (no exported symbols)



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b

1 files changed, 0 insertions, 3012 deletions

diff --git a/rtp.c b/rtp.c
deleted file mode 100644
index 373eb102d..000000000
--- a/rtp.c
+++ /dev/null
@@ -1,3012 +0,0 @@
-/*
- * 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();
-}
-