/
rtp.c
1919 lines (1740 loc) · 54.4 KB
/
rtp.c
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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2005, 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.
*
* RTP is deffined in RFC 3550.
*/
#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"
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"
#include "asterisk/utils.h"
#define MAX_TIMESTAMP_SKEW 640
#define RTP_MTU 1200
#define DEFAULT_DTMF_TIMEOUT 3000 /* samples */
static int dtmftimeout = DEFAULT_DTMF_TIMEOUT;
static int rtpstart = 0;
static int rtpend = 0;
static int rtpdebug = 0; /* Are we debugging? */
static struct sockaddr_in rtpdebugaddr; /* Debug packets to/from this host */
#ifdef SO_NO_CHECK
static int nochecksums = 0;
#endif
/* The value of each payload format mapping: */
struct rtpPayloadType {
int isAstFormat; /* whether the following code is an AST_FORMAT */
int code;
};
#define MAX_RTP_PT 256
#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)
struct ast_rtp {
int s;
char resp;
struct ast_frame f;
unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
/*! Synchronization source, RFC 3550, page 10. */
unsigned int ssrc;
unsigned int rxssrc;
unsigned int lastts;
unsigned int lastdigitts;
unsigned int lastrxts;
unsigned int lastividtimestamp;
unsigned int lastovidtimestamp;
unsigned int lasteventseqn;
unsigned int lasteventendseqn;
int lasttxformat;
int lastrxformat;
int dtmfcount;
unsigned int dtmfduration;
int nat;
unsigned int flags;
/*! Socket representation of the local endpoint. */
struct sockaddr_in us;
/*! Socket representation of the remote endpoint. */
struct sockaddr_in them;
struct timeval rxcore;
struct timeval txcore;
struct timeval dtmfmute;
struct ast_smoother *smoother;
int *ioid;
/*! Sequence number, RFC 3550, page 13. */
unsigned short seqno;
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];
/*! a cache for the result of rtp_lookup_code(): */
int rtp_lookup_code_cache_isAstFormat;
int rtp_lookup_code_cache_code;
int rtp_lookup_code_cache_result;
int rtp_offered_from_local;
struct ast_rtcp *rtcp;
};
/*!
* \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 {
/*! Socket */
int s;
/*! Socket representation of the local endpoint. */
struct sockaddr_in us;
/*! Socket representation of the remote endpoint. */
struct sockaddr_in them;
};
static struct ast_rtp_protocol *protos = NULL;
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;
}
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;
}
static struct ast_frame *send_dtmf(struct ast_rtp *rtp)
{
static struct ast_frame null_frame = { AST_FRAME_NULL, };
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));
rtp->resp = 0;
rtp->dtmfduration = 0;
return &null_frame;
}
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;
}
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 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 0
printf("Cisco Digit: %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);
} 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)
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))) {
char iabuf[INET_ADDRSTRLEN];
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;
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)
{
static struct ast_frame null_frame = { AST_FRAME_NULL, };
socklen_t len;
int hdrlen = 8;
int res;
struct sockaddr_in sin;
unsigned int rtcpdata[1024];
char iabuf[INET_ADDRSTRLEN];
if (!rtp || !rtp->rtcp)
return &null_frame;
len = sizeof(sin);
res = recvfrom(rtp->rtcp->s, rtcpdata, sizeof(rtcpdata),
0, (struct sockaddr *)&sin, &len);
if (res < 0) {
if (errno == EBADF)
CRASH;
if (errno != EAGAIN) {
ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up now.\n", strerror(errno));
return NULL;
}
return &null_frame;
}
if (res < hdrlen) {
ast_log(LOG_WARNING, "RTP Read too short\n");
return &null_frame;
}
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));
}
}
if (option_debug)
ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);
return &null_frame;
}
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;
}
*tv = ast_tvadd(rtp->rxcore, ts);
}
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 hdrlen = 12;
int padding;
int mark;
int ext;
int cc;
int x;
char iabuf[INET_ADDRSTRLEN];
unsigned int ssrc;
unsigned int timestamp;
unsigned int *rtpheader;
struct ast_frame *f;
static struct ast_frame null_frame = { AST_FRAME_NULL, };
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 == EBADF)
CRASH;
if (errno != EAGAIN) {
ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up now.\n", strerror(errno));
return NULL;
}
return &null_frame;
}
if (res < hdrlen) {
ast_log(LOG_WARNING, "RTP Read too short\n");
return &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 &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)) {
memcpy(&rtp->them, &sin, sizeof(rtp->them));
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(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port));
}
}
/* Get fields */
seqno = ntohl(rtpheader[0]);
/* Check RTP version */
version = (seqno & 0xC0000000) >> 30;
if (version != 2)
return &null_frame;
payloadtype = (seqno & 0x7f0000) >> 16;
padding = seqno & (1 << 29);
mark = seqno & (1 << 23);
ext = seqno & (1 << 28);
cc = (seqno & 0xF000000) >> 24;
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 (cc) {
/* CSRC fields present */
hdrlen += cc*4;
}
if (ext) {
/* RTP Extension present */
hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
hdrlen += 4;
}
if (res < hdrlen) {
ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
return &null_frame;
}
if(rtp_debug_test_addr(&sin))
ast_verbose("Got RTP packet from %s:%u (type %d, seq %u, ts %u, 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) {
/* 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
f = NULL;
if (f)
return f;
else
return &null_frame;
} 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
f = NULL;
if (f)
return f;
else
return &null_frame;
} else if (rtpPT.code == AST_RTP_CN) {
/* Comfort Noise */
f = process_rfc3389(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
if (f)
return f;
else
return &null_frame;
} else {
ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
return &null_frame;
}
}
rtp->f.subclass = rtpPT.code;
if (rtp->f.subclass < AST_FORMAT_MAX_AUDIO)
rtp->f.frametype = AST_FRAME_VOICE;
else
rtp->f.frametype = AST_FRAME_VIDEO;
rtp->lastrxformat = rtp->f.subclass;
if (!rtp->lastrxts)
rtp->lastrxts = timestamp;
if (rtp->rxseqno) {
for (x=rtp->rxseqno + 1; x < seqno; x++) {
/* Queue empty frames */
rtp->f.mallocd = 0;
rtp->f.datalen = 0;
rtp->f.data = NULL;
rtp->f.offset = 0;
rtp->f.samples = 0;
rtp->f.src = "RTPMissedFrame";
}
}
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);
} 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"},
{{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"},
};
/* 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},
[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)
{
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;
}
/* Make a note of a RTP paymoad type that was seen in a SDP "m=" line. */
/* By default, use the well-known value for this type (although it may */
/* still be set to a different value by a subsequent "a=rtpmap:" line): */
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt) {
if (pt < 0 || pt > MAX_RTP_PT)
return; /* bogus payload type */
if (static_RTP_PT[pt].code != 0) {
rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
}
}
/* Make a note of a RTP payload type (with MIME type) that was seen in */
/* a SDP "a=rtpmap:" line. */
void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
char* mimeType, char* mimeSubtype) {
int i;
if (pt < 0 || pt > MAX_RTP_PT)
return; /* bogus payload type */
for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
if (strcasecmp(mimeSubtype, mimeTypes[i].subtype) == 0 &&
strcasecmp(mimeType, mimeTypes[i].type) == 0) {
rtp->current_RTP_PT[pt] = mimeTypes[i].payloadType;
return;
}
}
}
/* 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;
}
}
}
void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local) {
if (rtp)
rtp->rtp_offered_from_local = local;
else
ast_log(LOG_WARNING, "rtp structure is null\n");
}
struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
{
struct rtpPayloadType result;
result.isAstFormat = result.code = 0;
if (pt < 0 || pt > MAX_RTP_PT)
return result; /* bogus payload type */
/* Start with the negotiated codecs */
if (!rtp->rtp_offered_from_local)
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;
}
/* Looks up an RTP code out of our *static* outbound list */
int ast_rtp_lookup_code(struct ast_rtp* rtp, const int isAstFormat, const int code) {
int pt;
if (isAstFormat == rtp->rtp_lookup_code_cache_isAstFormat &&
code == rtp->rtp_lookup_code_cache_code) {
/* Use our cached mapping, to avoid the overhead of the loop below */
return rtp->rtp_lookup_code_cache_result;
}
/* Check the dynamic list first */
for (pt = 0; pt < MAX_RTP_PT; ++pt) {
if (rtp->current_RTP_PT[pt].code == code && rtp->current_RTP_PT[pt].isAstFormat == isAstFormat) {
rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
rtp->rtp_lookup_code_cache_code = code;
rtp->rtp_lookup_code_cache_result = pt;
return pt;
}
}
/* Then the static list */
for (pt = 0; pt < MAX_RTP_PT; ++pt) {
if (static_RTP_PT[pt].code == code && static_RTP_PT[pt].isAstFormat == isAstFormat) {
rtp->rtp_lookup_code_cache_isAstFormat = isAstFormat;
rtp->rtp_lookup_code_cache_code = code;
rtp->rtp_lookup_code_cache_result = pt;
return pt;
}
}
return -1;
}
char* ast_rtp_lookup_mime_subtype(const int isAstFormat, const int code) {
int i;
for (i = 0; i < sizeof mimeTypes/sizeof mimeTypes[0]; ++i) {
if (mimeTypes[i].payloadType.code == code && mimeTypes[i].payloadType.isAstFormat == isAstFormat) {
return mimeTypes[i].subtype;
}
}
return "";
}
char *ast_rtp_lookup_mime_multiple(char *buf, int size, const int capability, const int isAstFormat)
{
int format;
unsigned len;
char *end = buf;
char *start = buf;
if (!buf || !size)
return NULL;
snprintf(end, size, "0x%x (", capability);
len = strlen(end);
end += len;
size -= len;
start = end;
for (format = 1; format < AST_RTP_MAX; format <<= 1) {
if (capability & format) {
const char *name = ast_rtp_lookup_mime_subtype(isAstFormat, format);
snprintf(end, size, "%s|", name);
len = strlen(end);
end += len;
size -= len;
}
}
if (start == end)
snprintf(start, size, "nothing)");
else if (size > 1)
*(end -1) = ')';
return buf;
}
static int rtp_socket(void)
{
int s;
long flags;
s = socket(AF_INET, SOCK_DGRAM, 0);
if (s > -1) {
flags = fcntl(s, F_GETFL);
fcntl(s, F_SETFL, flags | O_NONBLOCK);
#ifdef SO_NO_CHECK
if (nochecksums)
setsockopt(s, SOL_SOCKET, SO_NO_CHECK, &nochecksums, sizeof(nochecksums));
#endif
}
return s;
}
/*!
* \brief Initialize a new RTCP session.
*
* \returns The newly initialized RTCP session.
*/
static struct ast_rtcp *ast_rtcp_new(void)
{
struct ast_rtcp *rtcp;
rtcp = malloc(sizeof(struct ast_rtcp));
if (!rtcp)
return NULL;
memset(rtcp, 0, sizeof(struct ast_rtcp));
rtcp->s = rtp_socket();
rtcp->us.sin_family = AF_INET;
if (rtcp->s < 0) {
free(rtcp);
ast_log(LOG_WARNING, "Unable to allocate socket: %s\n", strerror(errno));
return NULL;
}
return rtcp;
}
struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr addr)
{
struct ast_rtp *rtp;
int x;
int first;
int startplace;
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 = rtp_socket();
rtp->ssrc = rand();
rtp->seqno = rand() & 0xffff;
if (rtp->s < 0) {
free(rtp);
ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));
return NULL;
}
if (sched && rtcpenable) {
rtp->sched = sched;
rtp->rtcp = ast_rtcp_new();
}
/* Select a random port number in the range of possible RTP */
x = (rand() % (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_addr = addr;
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). */