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Bug #6109: Fix unprotected list in RTP, implement AST_LIST macros, update doxygen docs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@7730 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
@@ -26,12 +26,13 @@
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#ifndef _ASTERISK_RTP_H
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#define _ASTERISK_RTP_H
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#include <netinet/in.h>
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#include "asterisk/frame.h"
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#include "asterisk/io.h"
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#include "asterisk/sched.h"
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#include "asterisk/channel.h"
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#include <netinet/in.h>
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#include "asterisk/linkedlists.h"
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#if defined(__cplusplus) || defined(c_plusplus)
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extern "C" {
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@@ -48,15 +49,15 @@ extern "C" {
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#define AST_RTP_MAX AST_RTP_CISCO_DTMF
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struct ast_rtp_protocol {
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/* Get RTP struct, or NULL if unwilling to transfer */
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/*! Get RTP struct, or NULL if unwilling to transfer */
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struct ast_rtp *(* const get_rtp_info)(struct ast_channel *chan);
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/* Get RTP struct, or NULL if unwilling to transfer */
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/*! Get RTP struct, or NULL if unwilling to transfer */
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struct ast_rtp *(* const get_vrtp_info)(struct ast_channel *chan);
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/* Set RTP peer */
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/*! Set RTP peer */
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int (* const set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer, int codecs, int nat_active);
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int (* const get_codec)(struct ast_channel *chan);
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const char * const type;
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struct ast_rtp_protocol *next;
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AST_LIST_ENTRY(ast_rtp_protocol) list;
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};
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/*!
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@@ -124,25 +125,25 @@ int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
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int ast_rtp_settos(struct ast_rtp *rtp, int tos);
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/* Setting RTP payload types from lines in a SDP description: */
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/*! \brief Setting RTP payload types from lines in a SDP description: */
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void ast_rtp_pt_clear(struct ast_rtp* rtp);
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/* Set payload types to defaults */
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/*! \brief Set payload types to defaults */
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void ast_rtp_pt_default(struct ast_rtp* rtp);
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void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt);
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void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
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char* mimeType, char* mimeSubtype);
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/* Mapping between RTP payload format codes and Asterisk codes: */
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/*! \brief Mapping between RTP payload format codes and Asterisk codes: */
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struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt);
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int ast_rtp_lookup_code(struct ast_rtp* rtp, int isAstFormat, int code);
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void ast_rtp_get_current_formats(struct ast_rtp* rtp,
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int* astFormats, int* nonAstFormats);
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/* Mapping an Asterisk code into a MIME subtype (string): */
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/*! \brief Mapping an Asterisk code into a MIME subtype (string): */
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char* ast_rtp_lookup_mime_subtype(int isAstFormat, int code);
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/* Build a string of MIME subtype names from a capability list */
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/*! \brief Build a string of MIME subtype names from a capability list */
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char *ast_rtp_lookup_mime_multiple(char *buf, int size, const int capability, const int isAstFormat);
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void ast_rtp_setnat(struct ast_rtp *rtp, int nat);
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221
rtp.c
221
rtp.c
@@ -61,38 +61,38 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#define RTP_MTU 1200
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#define DEFAULT_DTMF_TIMEOUT 3000 /* samples */
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#define DEFAULT_DTMF_TIMEOUT 3000 /*!< samples */
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static int dtmftimeout = DEFAULT_DTMF_TIMEOUT;
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static int rtpstart = 0;
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static int rtpend = 0;
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static int rtpdebug = 0; /* Are we debugging? */
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static struct sockaddr_in rtpdebugaddr; /* Debug packets to/from this host */
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static int rtpstart = 0; /*!< First port for RTP sessions (set in rtp.conf) */
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static int rtpend = 0; /*!< Last port for RTP sessions (set in rtp.conf) */
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static int rtpdebug = 0; /*!< Are we debugging? */
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static struct sockaddr_in rtpdebugaddr; /*!< Debug packets to/from this host */
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#ifdef SO_NO_CHECK
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static int nochecksums = 0;
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#endif
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/* The value of each payload format mapping: */
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/*! \brief The value of each payload format mapping: */
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struct rtpPayloadType {
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int isAstFormat; /* whether the following code is an AST_FORMAT */
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int isAstFormat; /*!< whether the following code is an AST_FORMAT */
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int code;
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};
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#define MAX_RTP_PT 256
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#define MAX_RTP_PT 256
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#define FLAG_3389_WARNING (1 << 0)
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#define FLAG_NAT_ACTIVE (3 << 1)
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#define FLAG_NAT_INACTIVE (0 << 1)
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#define FLAG_NAT_INACTIVE_NOWARN (1 << 1)
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/*! \brief RTP session description */
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struct ast_rtp {
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int s;
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char resp;
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struct ast_frame f;
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unsigned char rawdata[8192 + AST_FRIENDLY_OFFSET];
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/*! Synchronization source, RFC 3550, page 10. */
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unsigned int ssrc;
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unsigned int ssrc; /*!< Synchronization source, RFC 3550, page 10. */
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unsigned int lastts;
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unsigned int lastdigitts;
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unsigned int lastrxts;
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@@ -106,25 +106,21 @@ struct ast_rtp {
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unsigned int dtmfduration;
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int nat;
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unsigned int flags;
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/*! Socket representation of the local endpoint. */
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struct sockaddr_in us;
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/*! Socket representation of the remote endpoint. */
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struct sockaddr_in them;
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struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
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struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
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struct timeval rxcore;
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struct timeval txcore;
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struct timeval dtmfmute;
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struct ast_smoother *smoother;
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int *ioid;
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/*! Sequence number, RFC 3550, page 13. */
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unsigned short seqno;
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unsigned short seqno; /*!< Sequence number, RFC 3550, page 13. */
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unsigned short rxseqno;
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struct sched_context *sched;
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struct io_context *io;
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void *data;
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ast_rtp_callback callback;
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struct rtpPayloadType current_RTP_PT[MAX_RTP_PT];
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/*! a cache for the result of rtp_lookup_code(): */
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int rtp_lookup_code_cache_isAstFormat;
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int rtp_lookup_code_cache_isAstFormat; /*!< a cache for the result of rtp_lookup_code(): */
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int rtp_lookup_code_cache_code;
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int rtp_lookup_code_cache_result;
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struct ast_rtcp *rtcp;
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@@ -141,15 +137,13 @@ struct ast_rtp {
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*
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*/
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struct ast_rtcp {
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/*! Socket */
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int s;
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/*! Socket representation of the local endpoint. */
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struct sockaddr_in us;
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/*! Socket representation of the remote endpoint. */
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struct sockaddr_in them;
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int s; /*!< Socket */
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struct sockaddr_in us; /*!< Socket representation of the local endpoint. */
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struct sockaddr_in them; /*!< Socket representation of the remote endpoint. */
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};
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static struct ast_rtp_protocol *protos = NULL;
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/*! \brief List of current sessions */
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static AST_LIST_HEAD_STATIC(protos, ast_rtp_protocol);
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int ast_rtp_fd(struct ast_rtp *rtp)
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{
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@@ -229,9 +223,8 @@ static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *
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struct ast_frame *f = NULL;
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event = ntohl(*((unsigned int *)(data)));
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event &= 0x001F;
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#if 0
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printf("Cisco Digit: %08x (len = %d)\n", event, len);
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#endif
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if (option_debug > 2 || rtpdebug)
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ast_log(LOG_DEBUG, "Cisco DTMF Digit: %08x (len = %d)\n", event, len);
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if (event < 10) {
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resp = '0' + event;
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} else if (event < 11) {
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@@ -269,6 +262,7 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
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unsigned int duration;
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char resp = 0;
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struct ast_frame *f = NULL;
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event = ntohl(*((unsigned int *)(data)));
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event >>= 24;
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event_end = ntohl(*((unsigned int *)(data)));
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@@ -276,7 +270,7 @@ static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *dat
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event_end >>= 24;
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duration = ntohl(*((unsigned int *)(data)));
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duration &= 0xFFFF;
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if (rtpdebug)
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if (rtpdebug || option_debug > 2)
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ast_log(LOG_DEBUG, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
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if (event < 10) {
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resp = '0' + event;
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@@ -638,61 +632,61 @@ struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
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/* The following array defines the MIME Media type (and subtype) for each
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of our codecs, or RTP-specific data type. */
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static struct {
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struct rtpPayloadType payloadType;
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char* type;
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char* subtype;
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struct rtpPayloadType payloadType;
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char* type;
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char* subtype;
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} mimeTypes[] = {
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{{1, AST_FORMAT_G723_1}, "audio", "G723"},
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{{1, AST_FORMAT_GSM}, "audio", "GSM"},
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{{1, AST_FORMAT_ULAW}, "audio", "PCMU"},
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{{1, AST_FORMAT_ALAW}, "audio", "PCMA"},
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{{1, AST_FORMAT_G726}, "audio", "G726-32"},
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{{1, AST_FORMAT_ADPCM}, "audio", "DVI4"},
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{{1, AST_FORMAT_SLINEAR}, "audio", "L16"},
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{{1, AST_FORMAT_LPC10}, "audio", "LPC"},
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{{1, AST_FORMAT_G729A}, "audio", "G729"},
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{{1, AST_FORMAT_SPEEX}, "audio", "speex"},
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{{1, AST_FORMAT_ILBC}, "audio", "iLBC"},
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{{0, AST_RTP_DTMF}, "audio", "telephone-event"},
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{{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"},
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{{0, AST_RTP_CN}, "audio", "CN"},
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{{1, AST_FORMAT_JPEG}, "video", "JPEG"},
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{{1, AST_FORMAT_PNG}, "video", "PNG"},
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{{1, AST_FORMAT_H261}, "video", "H261"},
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{{1, AST_FORMAT_H263}, "video", "H263"},
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{{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998"},
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{{1, AST_FORMAT_G723_1}, "audio", "G723"},
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{{1, AST_FORMAT_GSM}, "audio", "GSM"},
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{{1, AST_FORMAT_ULAW}, "audio", "PCMU"},
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{{1, AST_FORMAT_ALAW}, "audio", "PCMA"},
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{{1, AST_FORMAT_G726}, "audio", "G726-32"},
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{{1, AST_FORMAT_ADPCM}, "audio", "DVI4"},
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{{1, AST_FORMAT_SLINEAR}, "audio", "L16"},
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{{1, AST_FORMAT_LPC10}, "audio", "LPC"},
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{{1, AST_FORMAT_G729A}, "audio", "G729"},
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{{1, AST_FORMAT_SPEEX}, "audio", "speex"},
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{{1, AST_FORMAT_ILBC}, "audio", "iLBC"},
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{{0, AST_RTP_DTMF}, "audio", "telephone-event"},
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{{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"},
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{{0, AST_RTP_CN}, "audio", "CN"},
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{{1, AST_FORMAT_JPEG}, "video", "JPEG"},
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{{1, AST_FORMAT_PNG}, "video", "PNG"},
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{{1, AST_FORMAT_H261}, "video", "H261"},
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{{1, AST_FORMAT_H263}, "video", "H263"},
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{{1, AST_FORMAT_H263_PLUS}, "video", "h263-1998"},
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};
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/* Static (i.e., well-known) RTP payload types for our "AST_FORMAT..."s:
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also, our own choices for dynamic payload types. This is our master
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table for transmission */
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static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = {
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[0] = {1, AST_FORMAT_ULAW},
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[0] = {1, AST_FORMAT_ULAW},
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#ifdef USE_DEPRECATED_G726
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[2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
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[2] = {1, AST_FORMAT_G726}, /* Technically this is G.721, but if Cisco can do it, so can we... */
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#endif
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[3] = {1, AST_FORMAT_GSM},
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[4] = {1, AST_FORMAT_G723_1},
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[5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
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[6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
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[7] = {1, AST_FORMAT_LPC10},
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[8] = {1, AST_FORMAT_ALAW},
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[10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
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[11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
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[13] = {0, AST_RTP_CN},
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[16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
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[17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
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[18] = {1, AST_FORMAT_G729A},
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[19] = {0, AST_RTP_CN}, /* Also used for CN */
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[26] = {1, AST_FORMAT_JPEG},
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[31] = {1, AST_FORMAT_H261},
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[34] = {1, AST_FORMAT_H263},
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[103] = {1, AST_FORMAT_H263_PLUS},
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[97] = {1, AST_FORMAT_ILBC},
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[101] = {0, AST_RTP_DTMF},
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[110] = {1, AST_FORMAT_SPEEX},
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[111] = {1, AST_FORMAT_G726},
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[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
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[3] = {1, AST_FORMAT_GSM},
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[4] = {1, AST_FORMAT_G723_1},
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[5] = {1, AST_FORMAT_ADPCM}, /* 8 kHz */
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[6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */
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[7] = {1, AST_FORMAT_LPC10},
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[8] = {1, AST_FORMAT_ALAW},
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[10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */
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[11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */
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[13] = {0, AST_RTP_CN},
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[16] = {1, AST_FORMAT_ADPCM}, /* 11.025 kHz */
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[17] = {1, AST_FORMAT_ADPCM}, /* 22.050 kHz */
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[18] = {1, AST_FORMAT_G729A},
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[19] = {0, AST_RTP_CN}, /* Also used for CN */
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[26] = {1, AST_FORMAT_JPEG},
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[31] = {1, AST_FORMAT_H261},
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[34] = {1, AST_FORMAT_H263},
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[103] = {1, AST_FORMAT_H263_PLUS},
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[97] = {1, AST_FORMAT_ILBC},
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[101] = {0, AST_RTP_DTMF},
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[110] = {1, AST_FORMAT_SPEEX},
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[111] = {1, AST_FORMAT_G726},
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[121] = {0, AST_RTP_CISCO_DTMF}, /* Must be type 121 */
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};
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void ast_rtp_pt_clear(struct ast_rtp* rtp)
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@@ -741,18 +735,18 @@ static void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src)
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dest->rtp_lookup_code_cache_result = 0;
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}
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/*--- get_proto: Get channel driver interface structure */
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/*! \brief Get channel driver interface structure */
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static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
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{
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struct ast_rtp_protocol *cur;
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cur = protos;
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while(cur) {
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if (cur->type == chan->type) {
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AST_LIST_LOCK(&protos);
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AST_LIST_TRAVERSE(&protos, cur, list) {
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if (cur->type == chan->type)
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return cur;
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}
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cur = cur->next;
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}
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AST_LIST_UNLOCK(&protos);
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return NULL;
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}
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@@ -773,13 +767,15 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
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destpr = get_proto(dest);
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srcpr = get_proto(src);
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if (!destpr) {
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ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
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if (option_debug)
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ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", dest->name);
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ast_mutex_unlock(&dest->lock);
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ast_mutex_unlock(&src->lock);
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return 0;
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}
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if (!srcpr) {
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ast_log(LOG_WARNING, "Channel '%s' has no RTP, not doing anything\n", src->name);
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if (option_debug)
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ast_log(LOG_DEBUG, "Channel '%s' has no RTP, not doing anything\n", src->name);
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ast_mutex_unlock(&dest->lock);
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ast_mutex_unlock(&src->lock);
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return 0;
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@@ -809,13 +805,15 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
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ast_rtp_pt_copy(vdestp, vsrcp);
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ast_mutex_unlock(&dest->lock);
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ast_mutex_unlock(&src->lock);
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ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
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if (option_debug)
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ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
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return 1;
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}
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/* Make a note of a RTP paymoad type that was seen in a SDP "m=" line. */
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/* By default, use the well-known value for this type (although it may */
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/* still be set to a different value by a subsequent "a=rtpmap:" line): */
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/*! \brief Make a note of a RTP paymoad type that was seen in a SDP "m=" line.
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* By default, use the well-known value for this type (although it may
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* still be set to a different value by a subsequent "a=rtpmap:" line)
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*/
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void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
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{
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if (pt < 0 || pt > MAX_RTP_PT)
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@@ -826,11 +824,10 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
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}
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}
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/* Make a note of a RTP payload type (with MIME type) that was seen in */
|
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/* a SDP "a=rtpmap:" line. */
|
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/*! \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 @@ void ast_rtp_set_rtpmap_type(struct ast_rtp* rtp, int pt,
|
||||
}
|
||||
}
|
||||
|
||||
/* 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 @@ struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt)
|
||||
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 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
|
||||
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;
|
||||
struct ast_rtp_protocol *cur;
|
||||
|
||||
cur = protos;
|
||||
prev = NULL;
|
||||
while(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;
|
||||
AST_LIST_REMOVE_CURRENT(&protos, list);
|
||||
break;
|
||||
}
|
||||
prev = cur;
|
||||
cur = cur->next;
|
||||
}
|
||||
AST_LIST_TRAVERSE_SAFE_END
|
||||
AST_LIST_UNLOCK(&protos);
|
||||
}
|
||||
|
||||
/*--- ast_rtp_proto_register: Register interface to channel driver */
|
||||
/*! \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 @@ void ast_rtp_reload(void)
|
||||
|
||||
}
|
||||
|
||||
/*--- 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);
|
||||
|
||||
Reference in New Issue
Block a user