[svn-commits] trunk - r7730 in /trunk: include/asterisk/rtp.h rtp.c
svn-commits at lists.digium.com
svn-commits at lists.digium.com
Tue Jan 3 03:30:21 CST 2006
Author: oej
Date: Tue Jan 3 03:30:19 2006
New Revision: 7730
URL: http://svn.digium.com/view/asterisk?rev=7730&view=rev
Log:
Bug #6109: Fix unprotected list in RTP, implement AST_LIST macros, update doxygen docs
Modified:
trunk/include/asterisk/rtp.h
trunk/rtp.c
Modified: trunk/include/asterisk/rtp.h
URL: http://svn.digium.com/view/asterisk/trunk/include/asterisk/rtp.h?rev=7730&r1=7729&r2=7730&view=diff
==============================================================================
--- trunk/include/asterisk/rtp.h (original)
+++ trunk/include/asterisk/rtp.h Tue Jan 3 03:30:19 2006
@@ -26,12 +26,13 @@
#ifndef _ASTERISK_RTP_H
#define _ASTERISK_RTP_H
+#include <netinet/in.h>
+
#include "asterisk/frame.h"
#include "asterisk/io.h"
#include "asterisk/sched.h"
#include "asterisk/channel.h"
-
-#include <netinet/in.h>
+#include "asterisk/linkedlists.h"
#if defined(__cplusplus) || defined(c_plusplus)
extern "C" {
@@ -48,15 +49,15 @@
#define AST_RTP_MAX AST_RTP_CISCO_DTMF
struct ast_rtp_protocol {
- /* Get RTP struct, or NULL if unwilling to transfer */
+ /*! Get RTP struct, or NULL if unwilling to transfer */
struct ast_rtp *(* const get_rtp_info)(struct ast_channel *chan);
- /* Get RTP struct, or NULL if unwilling to transfer */
+ /*! Get RTP struct, or NULL if unwilling to transfer */
struct ast_rtp *(* const get_vrtp_info)(struct ast_channel *chan);
- /* Set RTP peer */
+ /*! Set RTP peer */
int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, int codecs, int nat_active);
int (* const get_codec)(struct ast_channel *chan);
const char * const type;
- struct ast_rtp_protocol *next;
+ AST_LIST_ENTRY(ast_rtp_protocol) list;
};
/*!
@@ -124,25 +125,25 @@
int ast_rtp_settos(struct ast_rtp *rtp, int tos);
-/* Setting RTP payload types from lines in a SDP description: */
+/*! \brief Setting RTP payload types from lines in a SDP description: */
void ast_rtp_pt_clear(struct ast_rtp* rtp);
-/* Set payload types to defaults */
+/*! \brief Set payload types to defaults */
void ast_rtp_pt_default(struct ast_rtp* rtp);
void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
char* mimeType, char* mimeSubtype);
-/* Mapping between RTP payload format codes and Asterisk codes: */
+/*! \brief Mapping between RTP payload format codes and Asterisk codes: */
struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
void ast_rtp_get_current_formats(struct ast_rtp* rtp,
int* astFormats, int* nonAstFormats);
-/* Mapping an Asterisk code into a MIME subtype (string): */
+/*! \brief Mapping an Asterisk code into a MIME subtype (string): */
char* ast_rtp_lookup_mime_subtype(int isAstFormat, int code);
-/* Build a string of MIME subtype names from a capability list */
+/*! \brief Build a string of MIME subtype names from a capability list */
char *ast_rtp_lookup_mime_multiple(char *buf, int size, const int capability, const int isAstFormat);
void ast_rtp_setnat(struct ast_rtp *rtp, int nat);
Modified: trunk/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/rtp.c?rev=7730&r1=7729&r2=7730&view=diff
==============================================================================
--- trunk/rtp.c (original)
+++ trunk/rtp.c Tue Jan 3 03:30:19 2006
@@ -61,38 +61,38 @@
#define RTP_MTU 1200
-#define DEFAULT_DTMF_TIMEOUT 3000 /* samples */
+#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 */
+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 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: */
+/*! \brief The value of each payload format mapping: */
struct rtpPayloadType {
- int isAstFormat; /* whether the following code is an AST_FORMAT */
+ int isAstFormat; /*!< whether the following code is an AST_FORMAT */
int code;
};
-#define MAX_RTP_PT 256
+#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)
+/*! \brief RTP session description */
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 ssrc; /*!< Synchronization source, RFC 3550, page 10. */
unsigned int lastts;
unsigned int lastdigitts;
unsigned int lastrxts;
@@ -106,25 +106,21 @@
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 sockaddr_in us; /*!< Socket representation of the local endpoint. */
+ struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
struct timeval rxcore;
struct timeval txcore;
struct timeval dtmfmute;
struct ast_smoother *smoother;
int *ioid;
- /*! Sequence number, RFC 3550, page 13. */
- unsigned short seqno;
+ unsigned short seqno; /*!< Sequence number, RFC 3550, page 13. */
unsigned short rxseqno;
struct sched_context *sched;
struct io_context *io;
void *data;
ast_rtp_callback callback;
struct rtpPayloadType current_RTP_PT[MAX_RTP_PT];
- /*! a cache for the result of rtp_lookup_code(): */
- int rtp_lookup_code_cache_isAstFormat;
+ int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */
int rtp_lookup_code_cache_code;
int rtp_lookup_code_cache_result;
struct ast_rtcp *rtcp;
@@ -141,15 +137,13 @@
*
*/
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;
+ int s; /*!< Socket */
+ struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
+ struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
};
-static struct ast_rtp_protocol *protos = NULL;
+/*! \brief List of current sessions */
+static AST_LIST_HEAD_STATIC(protos, ast_rtp_protocol);
int ast_rtp_fd(struct ast_rtp *rtp)
{
@@ -229,9 +223,8 @@
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 (option_debug > 2 || rtpdebug)
+ ast_log(LOG_DEBUG, "Cisco DTMF Digit: %08x (len = %d)\n", event, len);
if (event < 10) {
resp = '0' + event;
} else if (event < 11) {
@@ -269,6 +262,7 @@
unsigned int duration;
char resp = 0;
struct ast_frame *f = NULL;
+
event = ntohl(*((unsigned int *)(data)));
event >>= 24;
event_end = ntohl(*((unsigned int *)(data)));
@@ -276,7 +270,7 @@
event_end >>= 24;
duration = ntohl(*((unsigned int *)(data)));
duration &= 0xFFFF;
- if (rtpdebug)
+ if (rtpdebug || option_debug > 2)
ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
if (event < 10) {
resp = '0' + event;
@@ -638,61 +632,61 @@
/* 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;
+ 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_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},
+ [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... */
+ [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 */
+ [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)
@@ -741,18 +735,18 @@
dest->rtp_lookup_code_cache_result = 0;
}
-/*--- get_proto: Get channel driver interface structure */
+/*! \brief Get channel driver interface structure */
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) {
+ AST_LIST_LOCK(&protos);
+ AST_LIST_TRAVERSE(&protos, cur, list) {
+ if (cur->type == chan->type)
return cur;
- }
- cur = cur->next;
- }
+ }
+ AST_LIST_UNLOCK(&protos);
+
return NULL;
}
@@ -773,13 +767,15 @@
destpr = get_proto(dest);
srcpr = get_proto(src);
if (!destpr) {
- ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
ast_mutex_unlock(&dest->lock);
ast_mutex_unlock(&src->lock);
return 0;
}
if (!srcpr) {
- ast_log(LOG_WARNING, "Channel '%s' has no RTP, not doing anything\n", src->name);
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
ast_mutex_unlock(&dest->lock);
ast_mutex_unlock(&src->lock);
return 0;
@@ -809,13 +805,15 @@
ast_rtp_pt_copy(vdestp, vsrcp);
ast_mutex_unlock(&dest->lock);
ast_mutex_unlock(&src->lock);
- ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
+ if (option_debug)
+ ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
return 1;
}
-/* 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): */
+/*! \brief 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)
@@ -826,11 +824,10 @@
}
}
-/* Make a note of a RTP payload type (with MIME type) that was seen in */
-/* a SDP "a=rtpmap:" line. */
+/*! \brief 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;
@@ -846,8 +843,8 @@
}
}
-/* 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 */
+/*! \brief Return the union of all of the codecs that were set by rtp_set...() calls
+ * They're returned as two distinct sets: AST_FORMATs, and AST_RTPs */
void ast_rtp_get_current_formats(struct ast_rtp* rtp,
int* astFormats, int* nonAstFormats) {
int pt;
@@ -879,7 +876,7 @@
return result;
}
-/* Looks up an RTP code out of our *static* outbound list */
+/*! \brief Looks up an RTP code out of our *static* outbound list */
int ast_rtp_lookup_code(struct ast_rtp* rtp, const int isAstFormat, const int code) {
int pt;
@@ -1533,44 +1530,42 @@
return 0;
}
-/*--- ast_rtp_proto_unregister: Unregister interface to channel driver */
+/*! \brief Unregister interface to channel driver */
void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
{
- struct ast_rtp_protocol *cur, *prev;
-
- cur = protos;
- prev = NULL;
- while(cur) {
+ struct ast_rtp_protocol *cur;
+
+ AST_LIST_LOCK(&protos);
+ AST_LIST_TRAVERSE_SAFE_BEGIN(&protos, cur, list) {
if (cur == proto) {
- if (prev)
- prev->next = proto->next;
- else
- protos = proto->next;
- return;
- }
- prev = cur;
- cur = cur->next;
- }
-}
-
-/*--- ast_rtp_proto_register: Register interface to channel driver */
+ AST_LIST_REMOVE_CURRENT(&protos, list);
+ break;
+ }
+ }
+ AST_LIST_TRAVERSE_SAFE_END
+ AST_LIST_UNLOCK(&protos);
+}
+
+/*! \brief Register interface to channel driver */
int ast_rtp_proto_register(struct ast_rtp_protocol *proto)
{
struct ast_rtp_protocol *cur;
- cur = protos;
- while(cur) {
+
+ AST_LIST_LOCK(&protos);
+ AST_LIST_TRAVERSE(&protos, cur, list) {
if (cur->type == proto->type) {
ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
+ AST_LIST_UNLOCK(&protos);
return -1;
}
- cur = cur->next;
- }
- proto->next = protos;
- protos = proto;
+ }
+ AST_LIST_INSERT_HEAD(&protos, proto, list);
+ AST_LIST_UNLOCK(&protos);
+
return 0;
}
-/* ast_rtp_bridge: Bridge calls. If possible and allowed, initiate
+/*! \brief Bridge calls. If possible and allowed, initiate
re-invite so the peers exchange media directly outside
of Asterisk. */
enum ast_bridge_result ast_rtp_bridge(struct ast_channel *c0, struct ast_channel *c1, int flags, struct ast_frame **fo, struct ast_channel **rc, int timeoutms)
@@ -1935,7 +1930,7 @@
}
-/*--- ast_rtp_init: Initialize the RTP system in Asterisk */
+/*! \brief Initialize the RTP system in Asterisk */
void ast_rtp_init(void)
{
ast_cli_register(&cli_debug);
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