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fix some formatting and add some comments (issue #5403)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.2-netsec
Russell Bryant 21 years ago
parent bdef480cdd
commit 8e5d45f3ac
2 changed files with 113 additions and 39 deletions
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34
include/asterisk/rtp.h
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@ -16,8 +16,11 @@
* at the top of the source tree.
*/
/*
* Real-time Transport Protocol support
/*!
* \file rtp.h
* \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
*
* RTP is deffined in RFC 3550.
*/
#ifndef _ASTERISK_RTP_H
@ -56,12 +59,39 @@ struct ast_rtp_protocol {
struct ast_rtp_protocol *next;
};
/*!
* \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 [...]"
*
*/
struct ast_rtp;
typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
/*!
* \brief Initializate a RTP session.
*
* \param sched
* \param io
* \param rtcpenable
* \param callbackmode
* \returns A representation (structure) of an RTP session.
*/
struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode);
/*!
* \brief Initializate a RTP session using an in_addr structure.
*
* This fuction gets called by ast_rtp_new().
*
* \param sched
* \param io
* \param rtcpenable
* \param callbackmode
* \param in
* \returns A representation (structure) of an RTP session.
*/
struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr in);
void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);

118
rtp.c
Unescape View File

@ -16,12 +16,11 @@
* at the top of the source tree.
*/
/*
*
* Real-time Protocol Support
* Supports RTP and RTCP with Symmetric RTP support for NAT
* traversal
/*!
* \file rtp.c
* \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
*
* RTP is deffined in RFC 3550.
*/
#include <stdio.h>
@ -59,7 +58,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#define RTP_MTU 1200
static int dtmftimeout = 3000; /* 3000 samples */
static int dtmftimeout = 3000; /* 3000 samples */
static int rtpstart = 0;
static int rtpend = 0;
@ -87,6 +86,7 @@ struct ast_rtp {
char resp;
struct ast_frame f;
unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
/*! Synchronization source, RFC 3550, page 10. */
unsigned int ssrc;
unsigned int lastts;
unsigned int lastdigitts;
@ -101,13 +101,16 @@ struct ast_rtp {
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;
@ -115,16 +118,30 @@ struct ast_rtp {
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(): */
/*! 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 {
int s; /* Socket */
/*! Socket */
int s;
/*! Socket representation of the local endpoint. */
struct sockaddr_in us;
/*! Socket representation of the remote endpoint. */
struct sockaddr_in them;
};
@ -230,9 +247,17 @@ static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *
return f;
}
/* process_rfc2833: Process RTP DTMF and events according to RFC 2833:
"RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals"
*/
/*!
* \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;
@ -282,8 +307,11 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
return f;
}
/*--- process_rfc3389: Process Comfort Noise RTP.
This is incomplete at the moment.
/*!
* \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)
{
@ -483,7 +511,7 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
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);
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) {
@ -503,37 +531,37 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
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 (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
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
} 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)
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)
} 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;
}
} 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)
@ -862,6 +890,11 @@ static int rtp_socket(void)
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;
@ -903,16 +936,21 @@ struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io
rtp->sched = sched;
rtp->rtcp = ast_rtcp_new();
}
/* Find us a place */
/* 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_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;
@ -922,6 +960,7 @@ struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io
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) {
@ -931,10 +970,15 @@ struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io
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) {

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