asterisk/rtp.c

/*
 * Asterisk -- A telephony toolkit for Linux.
 *
 * Real-time Protocol Support
 * 
 * Copyright (C) 1999, Mark Spencer
 *
 * Mark Spencer <markster@linux-support.net>
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License
 */

#include <stdio.h>
#include <stdlib.h>
#include <pthread.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/channel_pvt.h>

#define TYPE_SILENCE	 0x2
#define TYPE_HIGH	 0x0
#define TYPE_LOW	 0x1
#define TYPE_MASK	 0x3

static int dtmftimeout = 300;	/* 300 samples */

struct ast_rtp {
	int s;
	char resp;
	struct ast_frame f;
	unsigned char rawdata[1024 + AST_FRIENDLY_OFFSET];
	unsigned int ssrc;
	unsigned int lastts;
	unsigned int lastrxts;
	int lasttxformat;
	int lastrxformat;
	int dtmfcount;
	struct sockaddr_in us;
	struct sockaddr_in them;
	struct timeval rxcore;
	struct timeval txcore;
	struct ast_smoother *smoother;
	int *ioid;
	unsigned short seqno;
	struct sched_context *sched;
	struct io_context *io;
	void *data;
	ast_rtp_callback callback;
};

static struct ast_rtp_protocol *protos = NULL;

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

static int g723_len(unsigned char buf)
{
	switch(buf & TYPE_MASK) {
	case TYPE_MASK:
	case TYPE_SILENCE:
		return 4;
		break;
	case TYPE_HIGH:
		return 24;
		break;
	case TYPE_LOW:
		return 20;
		break;
	default:
		ast_log(LOG_WARNING, "Badly encoded frame (%d)\n", buf & TYPE_MASK);
	}
	return -1;
}

static int g723_samples(unsigned char *buf, int maxlen)
{
	int pos = 0;
	int samples = 0;
	int res;
	while(pos < maxlen) {
		res = g723_len(buf[pos]);
		if (res < 0)
			break;
		samples += 240;
		pos += res;
	}
	return samples;
}

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;
}

static struct ast_frame *send_dtmf(struct ast_rtp *rtp)
{
	printf("Sending dtmf: %d (%c)\n", rtp->resp, rtp->resp);
	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;
	return &rtp->f;
	
}

static struct ast_frame *process_rfc2833(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 >>= 24;
#if 0
	printf("Event: %08x (len = %d)\n", event, len);
#endif	
	if (event < 10) {
		resp = '0' + event;
	} else if (event < 11) {
		resp = '*';
	} else if (event < 12) {
		resp = '#';
	} else if (event < 16) {
		resp = 'A' + (event - 12);
	}
	if (rtp->resp && (rtp->resp != resp)) {
		f = send_dtmf(rtp);
	}
	rtp->resp = resp;
	rtp->dtmfcount = dtmftimeout;
	return f;
}

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 0
	printf("RFC3389: %d bytes, format is %d\n", len, rtp->lastrxformat);
#endif	
	ast_log(LOG_NOTICE, "RFC3389 support incomplete.  Turn off on client if possible\n");
	if (!rtp->lastrxformat)
		return 	NULL;
	switch(rtp->lastrxformat) {
	case AST_FORMAT_ULAW:
		rtp->f.frametype = AST_FRAME_VOICE;
		rtp->f.subclass = AST_FORMAT_ULAW;
		rtp->f.datalen = 160;
		rtp->f.samples = 160;
		memset(rtp->f.data, 0x7f, rtp->f.datalen);
		f = &rtp->f;
		break;
	case AST_FORMAT_ALAW:
		rtp->f.frametype = AST_FRAME_VOICE;
		rtp->f.subclass = AST_FORMAT_ALAW;
		rtp->f.datalen = 160;
		rtp->f.samples = 160;
		memset(rtp->f.data, 0x7e, rtp->f.datalen); /* XXX Is this right? XXX */
		f = &rtp->f;
		break;
	case AST_FORMAT_SLINEAR:
		rtp->f.frametype = AST_FRAME_VOICE;
		rtp->f.subclass = AST_FORMAT_SLINEAR;
		rtp->f.datalen = 320;
		rtp->f.samples = 160;
		memset(rtp->f.data, 0x00, rtp->f.datalen);
		f = &rtp->f;
		break;
	default:
		ast_log(LOG_NOTICE, "Don't know how to handle RFC3389 for receive codec %d\n", rtp->lastrxformat);
	}
	return f;
}

static struct ast_frame *process_type121(struct ast_rtp *rtp, unsigned char *data, int len)
{
	char resp = 0;
	struct ast_frame *f = NULL;
	unsigned char b0,b1,b2,b3,b4,b5,b6,b7;
	
	b0=*(data+0);b1=*(data+1);b2=*(data+2);b3=*(data+3);
	b4=*(data+4);b5=*(data+5);b6=*(data+6);b7=*(data+7);
//	printf("%u %u %u %u %u %u %u %u\n",b0,b1,b2,b3,b4,b5,b6,b7);
	if (b2==32) {
//		printf("Start %d\n",b3);
		if (b4==0) {
//			printf("Detection point for DTMF %d\n",b3);
			if (b3<10) {
				resp='0'+b3;
			} else if (b3<11) {
				resp='*';
			} else if (b3<12) {
				resp='#';
			} else if (b3<16) {
				resp='A'+(b3-12);
			}
			rtp->resp=resp;
			f = send_dtmf(rtp);
		}
	}
	if (b2==3) {
//		printf("Stop(3) %d\n",b3);
	}
	if (b2==0) {
//		printf("Stop(0) %d\n",b3);
	}
	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_rtp_read(struct ast_rtp *rtp)
{
	int res;
	struct sockaddr_in sin;
	int len;
	unsigned int seqno;
	int payloadtype;
	int hdrlen = 12;
	unsigned int timestamp;
	unsigned int *rtpheader;
	static struct ast_frame *f, null_frame = { AST_FRAME_NULL, };
	
	len = sizeof(sin);
	
	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) {
		ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
		if (errno == EBADF)
			CRASH;
		return &null_frame;
	}
	if (res < hdrlen) {
		ast_log(LOG_WARNING, "RTP Read too short\n");
		return &null_frame;
	}
	/* Get fields */
	seqno = ntohl(rtpheader[0]);
	payloadtype = (seqno & 0x7f0000) >> 16;
	seqno &= 0xffff;
	timestamp = ntohl(rtpheader[1]);
#if 0
	printf("Got RTP packet from %s:%d (type %d, seq %d, ts %d, len = %d)\n", inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
#endif	
	rtp->f.frametype = AST_FRAME_VOICE;
	rtp->f.subclass = rtp2ast(payloadtype);
	if (rtp->f.subclass < 0) {
		f = NULL;
		if (payloadtype == 101) {
			/* It's special -- rfc2833 process it */
			f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
		} else if (payloadtype == 121) {
			/* CISCO proprietary DTMF bridge */
			f = process_type121(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
		} else if (payloadtype == 100) {
			/* CISCO's notso proprietary DTMF bridge */
			f = process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
		} else if (payloadtype == 13) {
			f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
		} else {
			ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
		}
		if (f)
			return f;
		else
			return &null_frame;
	} else
		rtp->lastrxformat = rtp->f.subclass;

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

	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) {
		printf("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;
	switch(rtp->f.subclass) {
	case AST_FORMAT_ULAW:
	case AST_FORMAT_ALAW:
		rtp->f.samples = rtp->f.datalen;
		break;
	case AST_FORMAT_SLINEAR:
		rtp->f.samples = rtp->f.datalen / 2;
		break;
	case AST_FORMAT_GSM:
		rtp->f.samples = 160 * (rtp->f.datalen / 33);
		break;
	case AST_FORMAT_ADPCM:
		rtp->f.samples = rtp->f.datalen * 2;
		break;
	case AST_FORMAT_G729A:
		rtp->f.samples = rtp->f.datalen * 8;
		break;
	case AST_FORMAT_G723_1:
		rtp->f.samples = g723_samples(rtp->f.data, rtp->f.datalen);
		break;
	default:
		ast_log(LOG_NOTICE, "Unable to calculate samples for format %d\n", rtp->f.subclass);
		break;
	}
	rtp->f.src = "RTP";
	return &rtp->f;
}

static struct {
	int rtp;
	int ast;
	char *label;
} cmap[] = {
	{ 0, AST_FORMAT_ULAW, "PCMU" },
	{ 3, AST_FORMAT_GSM, "GSM" },
	{ 4, AST_FORMAT_G723_1, "G723" },
	{ 5, AST_FORMAT_ADPCM, "ADPCM" },
	{ 8, AST_FORMAT_ALAW, "PCMA" },
	{ 18, AST_FORMAT_G729A, "G729" },
};

int rtp2ast(int id)
{
	int x;
	for (x=0;x<sizeof(cmap) / sizeof(cmap[0]); x++) {
		if (cmap[x].rtp == id)
			return cmap[x].ast;
	}
	return -1;
}

int ast2rtp(int id)
{
	int x;
	for (x=0;x<sizeof(cmap) / sizeof(cmap[0]); x++) {
		if (cmap[x].ast == id)
			return cmap[x].rtp;
	}
	return -1;
}

char *ast2rtpn(int id)
{
	int x;
	for (x=0;x<sizeof(cmap) / sizeof(cmap[0]); x++) {
		if (cmap[x].ast == id)
			return cmap[x].label;
	}
	return "";
}
struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io)
{
	struct ast_rtp *rtp;
	int x;
	int flags;
	rtp = malloc(sizeof(struct ast_rtp));
	if (!rtp)
		return NULL;
	memset(rtp, 0, sizeof(struct ast_rtp));
	rtp->them.sin_family = AF_INET;
	rtp->us.sin_family = AF_INET;
	rtp->s = socket(AF_INET, SOCK_DGRAM, 0);
	rtp->ssrc = rand();
	rtp->seqno = rand() & 0xffff;
	if (rtp->s < 0) {
		free(rtp);
		ast_log(LOG_WARNING, "Unable to allocate socket: %s\n", strerror(errno));
		return NULL;
	}
	flags = fcntl(rtp->s, F_GETFL);
	fcntl(rtp->s, F_SETFL, flags | O_NONBLOCK);
	for (;;) {
		/* Find us a place */
		x = (rand() % (65000-1025)) + 1025;
		/* Must be an even port number by RTP spec */
		x = x & ~1;
		rtp->us.sin_port = htons(x);
		if (!bind(rtp->s, &rtp->us, sizeof(rtp->us)))
			break;
		if (errno != EADDRINUSE) {
			ast_log(LOG_WARNING, "Unexpected bind error: %s\n", strerror(errno));
			close(rtp->s);
			free(rtp);
			return NULL;
		}
	}
	if (io && sched) {
		/* 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);
	}
	return rtp;
}

int ast_rtp_settos(struct ast_rtp *rtp, int tos)
{
	int res;
	if ((res = setsockopt(rtp->s, SOL_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;
}

void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
{
	them->sin_family = AF_INET;
	them->sin_port = rtp->them.sin_port;
	them->sin_addr = rtp->them.sin_addr;
}

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

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

static unsigned int calc_txstamp(struct ast_rtp *rtp)
{
	struct timeval now;
	unsigned int ms;
	if (!rtp->txcore.tv_sec && !rtp->txcore.tv_usec) {
		gettimeofday(&rtp->txcore, NULL);
	}
	gettimeofday(&now, NULL);
	ms = (now.tv_sec - rtp->txcore.tv_sec) * 1000;
	ms += (now.tv_usec - rtp->txcore.tv_usec) / 1000;
	return ms;
}

int ast_rtp_senddigit(struct ast_rtp *rtp, char digit)
{
	unsigned int *rtpheader;
	int hdrlen = 12;
	int res;
	int ms;
	int pred;
	int x;
	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;
	}
	

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

	ms = calc_txstamp(rtp);
	/* Default prediction */
	pred = ms * 8;
	
	/* Get a pointer to the header */
	rtpheader = (unsigned int *)data;
	rtpheader[0] = htonl((2 << 30) | (1 << 23) | (101 << 16) | (rtp->seqno++));
	rtpheader[1] = htonl(rtp->lastts);
	rtpheader[2] = htonl(rtp->ssrc); 
	rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0));
	for (x=0;x<4;x++) {
		if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
			res = sendto(rtp->s, (void *)rtpheader, hdrlen + 4, 0, &rtp->them, sizeof(rtp->them));
			if (res <0) 
				ast_log(LOG_NOTICE, "RTP Transmission error to %s:%d: %s\n", inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
	#if 0
		printf("Sent %d bytes of RTP data to %s:%d\n", res, inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
	#endif		
		}
		if (x ==0) {
			/* Clear marker bit and increment seqno */
			rtpheader[0] = htonl((2 << 30)  | (101 << 16) | (rtp->seqno++));
			/* Make duration 240 */
			rtpheader[3] |= htonl((240));
			/* Set the End bit for the last 3 */
			rtpheader[3] |= htonl((1 << 23));
		}
	}
	return 0;
}

static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec)
{
	unsigned int *rtpheader;
	int hdrlen = 12;
	int res;
	int ms;
	int pred;

	ms = calc_txstamp(rtp);
	/* Default prediction */
	pred = ms * 8;
	
	switch(f->subclass) {
	case AST_FORMAT_ULAW:
	case AST_FORMAT_ALAW:
		/* If we're within +/- 20ms from when where we
		   predict we should be, use that */
		pred = rtp->lastts + f->datalen;
		break;
	case AST_FORMAT_G729A:
		pred = rtp->lastts + f->datalen * 8;
		break;
	case AST_FORMAT_GSM:
		pred = rtp->lastts + f->datalen * 20 / 33;
		break;
	case AST_FORMAT_G723_1:
		pred = rtp->lastts + g723_samples(f->data, f->datalen);
		break;
	default:
		ast_log(LOG_WARNING, "Not sure about timestamp format for codec format %d\n", f->subclass);
	}

	/* Re-calculate last TS */
	rtp->lastts = ms * 8;
	
	/* If it's close to ou prediction, go for it */
	if (abs(rtp->lastts - pred) < 640)
		rtp->lastts = pred;
#if 0
	else
		printf("Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
#endif			
	/* Get a pointer to the header */
	rtpheader = (unsigned int *)(f->data - hdrlen);
	rtpheader[0] = htonl((2 << 30) | (codec << 16) | (rtp->seqno++));
	rtpheader[1] = htonl(rtp->lastts);
	rtpheader[2] = htonl(rtp->ssrc); 
	if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
		res = sendto(rtp->s, (void *)rtpheader, f->datalen + hdrlen, 0, &rtp->them, sizeof(rtp->them));
		if (res <0) 
			ast_log(LOG_NOTICE, "RTP Transmission error to %s:%d: %s\n", inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
#if 0
		printf("Sent %d bytes of RTP data to %s:%d\n", res, inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
#endif		
	}
	return 0;
}

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

	/* If we have no peer, return immediately */	
	if (!rtp->them.sin_addr.s_addr)
		return 0;
	
	/* Make sure we have enough space for RTP header */
	if (_f->frametype != AST_FRAME_VOICE) {
		ast_log(LOG_WARNING, "RTP can only send voice\n");
		return -1;
	}

	codec = ast2rtp(_f->subclass);
	if (codec < 0) {
		ast_log(LOG_WARNING, "Don't know how to send format %d packets with RTP\n", _f->subclass);
		return -1;
	}

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


	switch(_f->subclass) {
	case AST_FORMAT_ULAW:
	case AST_FORMAT_ALAW:
		if (!rtp->smoother) {
			rtp->smoother = ast_smoother_new(160);
		}
		if (!rtp->smoother) {
			ast_log(LOG_WARNING, "Unable to create smoother :(\n");
			return -1;
		}
		ast_smoother_feed(rtp->smoother, _f);
		
		while((f = ast_smoother_read(rtp->smoother)))
			ast_rtp_raw_write(rtp, f, codec);
		break;
	case AST_FORMAT_G729A:
		if (!rtp->smoother) {
			rtp->smoother = ast_smoother_new(20);
		}
		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;
	default:	
		ast_log(LOG_WARNING, "Not sure about sending format %d packets\n", _f->subclass);
		if (_f->offset < hdrlen) {
			f = ast_frdup(_f);
		} else {
			f = _f;
		}
		ast_rtp_raw_write(rtp, f, codec);
	}
		
	return 0;
}

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

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

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

int ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc)
{
	struct ast_frame *f;
	struct ast_channel *who, *cs[3];
	struct ast_rtp *p0, *p1;
	struct ast_rtp_protocol *pr0, *pr1;
	void *pvt0, *pvt1;
	int to;

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

	/* if need DTMF, cant native bridge */
	if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
		return -2;
	ast_pthread_mutex_lock(&c0->lock);
	ast_pthread_mutex_lock(&c1->lock);
	pr0 = get_proto(c0);
	pr1 = get_proto(c1);
	if (!pr0) {
		ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c0->name);
		ast_pthread_mutex_unlock(&c0->lock);
		ast_pthread_mutex_unlock(&c1->lock);
		return -1;
	}
	if (!pr1) {
		ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
		ast_pthread_mutex_unlock(&c0->lock);
		ast_pthread_mutex_unlock(&c1->lock);
		return -1;
	}
	pvt0 = c0->pvt->pvt;
	pvt1 = c1->pvt->pvt;
	p0 = pr0->get_rtp_info(c0);
	p1 = pr1->get_rtp_info(c1);
	if (!p0 || !p1) {
		/* Somebody doesn't want to play... */
		ast_pthread_mutex_unlock(&c0->lock);
		ast_pthread_mutex_unlock(&c1->lock);
		return -2;
	}
	if (pr0->set_rtp_peer(c0, p1)) 
		ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
	if (pr1->set_rtp_peer(c1, p0)) 
		ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name);
	ast_pthread_mutex_unlock(&c0->lock);
	ast_pthread_mutex_unlock(&c1->lock);
	cs[0] = c0;
	cs[1] = c1;
	cs[2] = NULL;
	for (;;) {
		if ((c0->pvt->pvt != pvt0)  ||
			(c1->pvt->pvt != pvt1) ||
			(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
				ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
				if (c0->pvt->pvt == pvt0) {
					if (pr0->set_rtp_peer(c0, NULL)) 
						ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
				}
				if (c1->pvt->pvt == pvt1) {
					if (pr1->set_rtp_peer(c1, NULL)) 
						ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
				}
				/* Tell it to try again later */
				return -3;
		}
		to = -1;
		who = ast_waitfor_n(cs, 2, &to);
		if (!who) {
			ast_log(LOG_DEBUG, "Ooh, empty read...\n");
			continue;
		}
		f = ast_read(who);
		if (!f || ((f->frametype == AST_FRAME_DTMF) &&
				   (((who == c0) && (flags & AST_BRIDGE_DTMF_CHANNEL_0)) || 
			       ((who == c1) && (flags & AST_BRIDGE_DTMF_CHANNEL_1))))) {
			*fo = f;
			*rc = who;
			ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
			if ((c0->pvt->pvt == pvt0) && (!c0->_softhangup)) {
				if (pr0->set_rtp_peer(c0, NULL)) 
					ast_log(LOG_WARNING, "Channel '%s' failed to revert\n", c0->name);
			}
			if ((c1->pvt->pvt == pvt1) && (!c1->_softhangup)) {
				if (pr1->set_rtp_peer(c1, NULL)) 
					ast_log(LOG_WARNING, "Channel '%s' failed to revert back\n", c1->name);
			}
			/* That's all we needed */
			return 0;
		} else 
			ast_frfree(f);
		/* Swap priority not that it's a big deal at this point */
		cs[2] = cs[0];
		cs[0] = cs[1];
		cs[1] = cs[2];
		
	}
	return -1;
}