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    /*
    
     * Asterisk -- An open source telephony toolkit.
    
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     *
    
     * Copyright (C) 1999 - 2006, Digium, Inc.
    
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     *
    
     * Mark Spencer <markster@digium.com>
    
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     *
    
     * 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.
     *
    
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     * 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.
     */
    
    
     * \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.
    
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     */
    
    #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.h"
    
    
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    ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
    
    #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"
    
    #define MAX_TIMESTAMP_SKEW	640
    
    
    #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 stundebug = 0;		/*!< Are we debugging stun? */
    
    static struct sockaddr_in rtpdebugaddr;	/*!< Debug packets to/from this host */
    
    static int nochecksums = 0;
    
    /*! \brief The value of each payload format mapping: */
    
    struct rtpPayloadType {
    
    	int isAstFormat; 	/*!< whether the following code is an AST_FORMAT */
    
    	int code;
    
    #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)
    
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    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 rxssrc;
    
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    	unsigned int lastts;
    
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    	unsigned int lastrxts;
    
    	unsigned int lastividtimestamp;
    	unsigned int lastovidtimestamp;
    
    	unsigned int lasteventseqn;
    
    	unsigned int lasteventendseqn;
    
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    	int lasttxformat;
    
    	int lastrxformat;
    
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    	int dtmfcount;
    
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    	unsigned int dtmfduration;
    
    	int nat;
    
    	struct sockaddr_in us;		/*!< Socket representation of the local endpoint. */
    	struct sockaddr_in them;	/*!< Socket representation of the remote endpoint. */
    
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    	struct timeval rxcore;
    	struct timeval txcore;
    
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    	struct timeval dtmfmute;
    
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    	struct ast_smoother *smoother;
    
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    	int *ioid;
    
    	unsigned short seqno;		/*!< Sequence number, RFC 3550, page 13. */
    
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    	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;
    
    /*!
     * \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.
     * 
     */
    
    	int s;				/*!< Socket */
    	struct sockaddr_in us;		/*!< Socket representation of the local endpoint. */
    	struct sockaddr_in them;	/*!< Socket representation of the remote endpoint. */
    
    
    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
    
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    #define STUN_SECREQ	0x0002
    
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    #define STUN_SECERR	0x0112
    
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    #define STUN_MAPPED_ADDRESS	0x0001
    
    #define STUN_RESPONSE_ADDRESS	0x0002
    
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    #define STUN_CHANGE_REQUEST	0x0003
    #define STUN_SOURCE_ADDRESS	0x0004
    
    #define STUN_CHANGED_ADDRESS	0x0005
    
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    #define STUN_USERNAME		0x0006
    #define STUN_PASSWORD		0x0007
    
    #define STUN_MESSAGE_INTEGRITY	0x0008
    
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    #define STUN_ERROR_CODE		0x0009
    
    #define STUN_UNKNOWN_ATTRIBUTES	0x000a
    
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    #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 {
    	unsigned char *username;
    	unsigned 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 = (unsigned char *)(attr->value);
    		break;
    	case STUN_PASSWORD:
    		state->password = (unsigned 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();
    }
    
    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)) {
    
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    		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)) {
    
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    		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)) {
    
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    			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) {
    
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    			if (option_debug)
    
    				ast_log(LOG_DEBUG, "Inconsistant Attribute (length %d exceeds remaining msg len %zd)\n", ntohs(attr->len), len);
    
    			break;
    		}
    		if (stun_process_attr(&st, attr)) {
    
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    			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 ? (const char *)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);
    
    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;
    }
    
    
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    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)
    
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    {
    
    	char iabuf[INET_ADDRSTRLEN];
    
    
    	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(iabuf, sizeof(iabuf), rtp->them.sin_addr));
    
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    		rtp->resp = 0;
    		rtp->dtmfduration = 0;
    
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    	}
    
    	if (option_debug)
    		ast_log(LOG_DEBUG, "Sending dtmf: %d (%c), at %s\n", rtp->resp, rtp->resp, ast_inet_ntoa(iabuf, sizeof(iabuf), 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;
    	}
    
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    	rtp->f.datalen = 0;
    
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    	rtp->f.samples = 0;
    
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    	rtp->f.mallocd = 0;
    	rtp->f.src = "RTP";
    	rtp->resp = 0;
    
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    	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;
    }
    
    
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    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);
    
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    	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';
    
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    	}
    	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)
    
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    {
    	unsigned int event;
    
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    	unsigned int event_end;
    	unsigned int duration;
    
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    	char resp = 0;
    
    	struct ast_frame *f = NULL;
    
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    	event = ntohl(*((unsigned int *)(data)));
    	event >>= 24;
    
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    	event_end = ntohl(*((unsigned int *)(data)));
    	event_end <<= 8;
    	event_end >>= 24;
    	duration = ntohl(*((unsigned int *)(data)));
    	duration &= 0xFFFF;
    
    		ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
    
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    	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';	
    
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    	}
    	if (rtp->resp && (rtp->resp != resp)) {
    
    		f = send_dtmf(rtp);
    
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    		if (rtp->resp) {
    
    			if(rtp->lasteventendseqn != seqno) {
    				f = send_dtmf(rtp);
    				rtp->lasteventendseqn = seqno;
    			}
    
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    			rtp->resp = 0;
    		}
    
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    		resp = 0;
    		duration = 0;
    
    	} else if (rtp->resp && rtp->dtmfduration && (duration < rtp->dtmfduration)) {
    
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    		f = send_dtmf(rtp);
    	}
    
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    	if (!(event_end & 0x80))
    		rtp->resp = resp;
    
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    	rtp->dtmfcount = dtmftimeout;
    
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    	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 */
    
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    		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(iabuf, sizeof(iabuf), 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;
    
    		memcpy(rtp->f.data, data + 1, len - 1);
    	} else {
    
    	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;
    
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    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)
    {
    
    	int hdrlen = 8;
    	int res;
    	struct sockaddr_in sin;
    	unsigned int rtcpdata[1024];
    
    	char iabuf[INET_ADDRSTRLEN];
    
    	if (!rtp || !rtp->rtcp)
    
    
    	len = sizeof(sin);
    	
    	res = recvfrom(rtp->rtcp->s, rtcpdata, sizeof(rtcpdata),
    					0, (struct sockaddr *)&sin, &len);
    	
    	if (res < 0) {
    
    		if (errno != EAGAIN)
    
    			ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
    
    	}
    
    	if (res < hdrlen) {
    		ast_log(LOG_WARNING, "RTP Read too short\n");
    
    	}
    
    	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(iabuf, sizeof(iabuf), rtp->rtcp->them.sin_addr), ntohs(rtp->rtcp->them.sin_port));
    
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    	if (option_debug)
    		ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);
    
    static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int timestamp, int mark)
    
    	struct timeval ts = ast_samp2tv( timestamp, 8000);
    	if (ast_tvzero(rtp->rxcore) || mark) {
    		rtp->rxcore = ast_tvsub(ast_tvnow(), ts);
    
    		/* Round to 20ms for nice, pretty timestamps */
    
    		rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 20000;
    
    struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
    {
    
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    	int res;
    	struct sockaddr_in sin;
    
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    	unsigned int seqno;
    
    	int version;
    
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    	int payloadtype;
    	int hdrlen = 12;
    
    	char iabuf[INET_ADDRSTRLEN];
    
    	unsigned int ssrc;
    
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    	unsigned int timestamp;
    	unsigned int *rtpheader;
    
    	struct rtpPayloadType rtpPT;
    
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    	len = sizeof(sin);
    	
    
    	/* Cache where the header will go */
    
    	res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
    
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    					0, (struct sockaddr *)&sin, &len);
    
    
    	rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
    
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    	if (res < 0) {
    
    		if (errno != EAGAIN)
    
    			ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
    
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    		if (errno == EBADF)
    			CRASH;
    
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    	}
    	if (res < hdrlen) {
    		ast_log(LOG_WARNING, "RTP Read too short\n");
    
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    	}
    
    	/* 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)
    
    	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;
    
    			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(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port));
    
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    	payloadtype = (seqno & 0x7f0000) >> 16;
    
    	padding = seqno & (1 << 29);
    
    	ext = seqno & (1 << 28);
    
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    	seqno &= 0xffff;
    	timestamp = ntohl(rtpheader[1]);
    
    	ssrc = ntohl(rtpheader[2]);
    	
    	if (!mark && rtp->rxssrc && rtp->rxssrc != ssrc) {
    		ast_log(LOG_WARNING, "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);
    
    	if(rtp_debug_test_addr(&sin))
    
    		ast_verbose("Got RTP packet from %s:%d (type %d, seq %d, ts %d, len %d)\n"
    
    			, ast_inet_ntoa(iabuf, sizeof(iabuf), 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 rfc2833 RTP packet from %s:%d (type %d, seq %d, ts %d, len %d, mark %d, event %08x, end %d, duration %d) \n", ast_inet_ntoa(iabuf, sizeof(iabuf), 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(iabuf, sizeof(iabuf), 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;
    
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    	if (!rtp->lastrxts)
    		rtp->lastrxts = timestamp;
    
    
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    	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);
    
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    	}
    	rtp->f.mallocd = 0;
    
    	rtp->f.datalen = res - hdrlen;
    
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    	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) 
    
    		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 */
    
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    		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;
    		
    
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    	}
    	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. */
    
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    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"},
    	{{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] = {
    
    #ifdef USE_DEPRECATED_G726
    
    	[2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
    
    	[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},
    	[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
    
    void ast_rtp_pt_clear(struct ast_rtp* rtp) 
    {
    
    	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;
    
    static void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src)
    {
    	int i;
    	/* Copy payload types from source to destination */
    	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;