git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@13423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1936 lines
55 KiB
C
1936 lines
55 KiB
C
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2006, Digium, Inc.
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*
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* Mark Spencer <markster@digium.com>
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*
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* See http://www.asterisk.org for more information about
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* the Asterisk project. Please do not directly contact
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* any of the maintainers of this project for assistance;
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* the project provides a web site, mailing lists and IRC
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* channels for your use.
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License Version 2. See the LICENSE file
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* at the top of the source tree.
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*/
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/*!
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* \file
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*
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* \brief Supports RTP and RTCP with Symmetric RTP support for NAT traversal.
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*
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* \author Mark Spencer <markster@digium.com>
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*
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* \note RTP is deffined in RFC 3550.
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*/
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#include <stdio.h>
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#include <stdlib.h>
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#include <string.h>
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#include <sys/time.h>
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#include <signal.h>
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#include <errno.h>
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#include <unistd.h>
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#include <netinet/in.h>
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#include <sys/time.h>
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#include <sys/socket.h>
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#include <arpa/inet.h>
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#include <fcntl.h>
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include "asterisk/rtp.h"
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#include "asterisk/frame.h"
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#include "asterisk/logger.h"
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#include "asterisk/options.h"
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#include "asterisk/channel.h"
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#include "asterisk/acl.h"
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#include "asterisk/channel.h"
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#include "asterisk/config.h"
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#include "asterisk/lock.h"
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#include "asterisk/utils.h"
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#include "asterisk/cli.h"
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#include "asterisk/unaligned.h"
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#include "asterisk/utils.h"
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#define MAX_TIMESTAMP_SKEW 640
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#define RTP_MTU 1200
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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; /*!< 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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/*! \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 code;
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};
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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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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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unsigned int lastividtimestamp;
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unsigned int lastovidtimestamp;
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unsigned int lasteventseqn;
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unsigned int lasteventendseqn;
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int lasttxformat;
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int lastrxformat;
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int dtmfcount;
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unsigned int dtmfduration;
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int nat;
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unsigned int flags;
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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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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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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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};
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/*!
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* \brief Structure defining an RTCP session.
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*
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* The concept "RTCP session" is not defined in RFC 3550, but since
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* this structure is analogous to ast_rtp, which tracks a RTP session,
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* it is logical to think of this as a RTCP session.
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*
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* RTCP packet is defined on page 9 of RFC 3550.
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*
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*/
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struct ast_rtcp {
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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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/*! \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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return rtp->s;
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}
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int ast_rtcp_fd(struct ast_rtp *rtp)
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{
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if (rtp->rtcp)
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return rtp->rtcp->s;
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return -1;
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}
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void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
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{
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rtp->data = data;
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}
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void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback)
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{
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rtp->callback = callback;
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}
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void ast_rtp_setnat(struct ast_rtp *rtp, int nat)
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{
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rtp->nat = nat;
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}
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static struct ast_frame *send_dtmf(struct ast_rtp *rtp)
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{
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char iabuf[INET_ADDRSTRLEN];
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if (ast_tvcmp(ast_tvnow(), rtp->dtmfmute) < 0) {
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if (option_debug)
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ast_log(LOG_DEBUG, "Ignore potential DTMF echo from '%s'\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr));
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rtp->resp = 0;
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rtp->dtmfduration = 0;
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return &ast_null_frame;
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}
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if (option_debug)
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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));
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if (rtp->resp == 'X') {
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rtp->f.frametype = AST_FRAME_CONTROL;
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rtp->f.subclass = AST_CONTROL_FLASH;
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} else {
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rtp->f.frametype = AST_FRAME_DTMF;
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rtp->f.subclass = rtp->resp;
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}
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rtp->f.datalen = 0;
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rtp->f.samples = 0;
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rtp->f.mallocd = 0;
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rtp->f.src = "RTP";
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rtp->resp = 0;
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rtp->dtmfduration = 0;
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return &rtp->f;
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}
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static inline int rtp_debug_test_addr(struct sockaddr_in *addr)
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{
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if (rtpdebug == 0)
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return 0;
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if (rtpdebugaddr.sin_addr.s_addr) {
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if (((ntohs(rtpdebugaddr.sin_port) != 0)
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&& (rtpdebugaddr.sin_port != addr->sin_port))
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|| (rtpdebugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
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return 0;
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}
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return 1;
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}
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static struct ast_frame *process_cisco_dtmf(struct ast_rtp *rtp, unsigned char *data, int len)
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{
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unsigned int event;
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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 &= 0x001F;
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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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resp = '*';
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} else if (event < 12) {
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resp = '#';
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} else if (event < 16) {
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resp = 'A' + (event - 12);
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} else if (event < 17) {
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resp = 'X';
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}
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if (rtp->resp && (rtp->resp != resp)) {
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f = send_dtmf(rtp);
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}
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rtp->resp = resp;
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rtp->dtmfcount = dtmftimeout;
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return f;
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}
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/*!
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* \brief Process RTP DTMF and events according to RFC 2833.
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*
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* RFC 2833 is "RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals".
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*
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* \param rtp
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* \param data
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* \param len
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* \param seqno
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* \returns
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*/
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static struct ast_frame *process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len, unsigned int seqno)
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{
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unsigned int event;
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unsigned int event_end;
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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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event_end <<= 8;
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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 || 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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} else if (event < 11) {
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resp = '*';
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} else if (event < 12) {
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resp = '#';
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} else if (event < 16) {
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resp = 'A' + (event - 12);
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} else if (event < 17) { /* Event 16: Hook flash */
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resp = 'X';
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}
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if (rtp->resp && (rtp->resp != resp)) {
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f = send_dtmf(rtp);
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} else if(event_end & 0x80) {
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if (rtp->resp) {
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if(rtp->lasteventendseqn != seqno) {
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f = send_dtmf(rtp);
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rtp->lasteventendseqn = seqno;
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}
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rtp->resp = 0;
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}
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resp = 0;
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duration = 0;
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} else if (rtp->resp && rtp->dtmfduration && (duration < rtp->dtmfduration)) {
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f = send_dtmf(rtp);
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}
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if (!(event_end & 0x80))
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rtp->resp = resp;
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rtp->dtmfcount = dtmftimeout;
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rtp->dtmfduration = duration;
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return f;
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}
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/*!
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* \brief Process Comfort Noise RTP.
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*
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* This is incomplete at the moment.
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*
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*/
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static struct ast_frame *process_rfc3389(struct ast_rtp *rtp, unsigned char *data, int len)
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{
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struct ast_frame *f = NULL;
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/* Convert comfort noise into audio with various codecs. Unfortunately this doesn't
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totally help us out becuase we don't have an engine to keep it going and we are not
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guaranteed to have it every 20ms or anything */
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if (rtpdebug)
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ast_log(LOG_DEBUG, "- RTP 3389 Comfort noise event: Level %d (len = %d)\n", rtp->lastrxformat, len);
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if (!(ast_test_flag(rtp, FLAG_3389_WARNING))) {
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char iabuf[INET_ADDRSTRLEN];
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ast_log(LOG_NOTICE, "Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: %s\n",
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ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr));
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ast_set_flag(rtp, FLAG_3389_WARNING);
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}
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/* Must have at least one byte */
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if (!len)
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return NULL;
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if (len < 24) {
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rtp->f.data = rtp->rawdata + AST_FRIENDLY_OFFSET;
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rtp->f.datalen = len - 1;
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rtp->f.offset = AST_FRIENDLY_OFFSET;
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memcpy(rtp->f.data, data + 1, len - 1);
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} else {
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rtp->f.data = NULL;
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rtp->f.offset = 0;
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rtp->f.datalen = 0;
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}
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rtp->f.frametype = AST_FRAME_CNG;
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rtp->f.subclass = data[0] & 0x7f;
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rtp->f.datalen = len - 1;
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rtp->f.samples = 0;
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rtp->f.delivery.tv_usec = rtp->f.delivery.tv_sec = 0;
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f = &rtp->f;
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return f;
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}
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static int rtpread(int *id, int fd, short events, void *cbdata)
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{
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struct ast_rtp *rtp = cbdata;
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struct ast_frame *f;
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f = ast_rtp_read(rtp);
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if (f) {
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if (rtp->callback)
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rtp->callback(rtp, f, rtp->data);
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}
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return 1;
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}
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struct ast_frame *ast_rtcp_read(struct ast_rtp *rtp)
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{
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socklen_t len;
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int hdrlen = 8;
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int res;
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struct sockaddr_in sin;
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unsigned int rtcpdata[1024];
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char iabuf[INET_ADDRSTRLEN];
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if (!rtp || !rtp->rtcp)
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return &ast_null_frame;
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len = sizeof(sin);
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res = recvfrom(rtp->rtcp->s, rtcpdata, sizeof(rtcpdata),
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0, (struct sockaddr *)&sin, &len);
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if (res < 0) {
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if (errno != EAGAIN)
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ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
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if (errno == EBADF)
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CRASH;
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return &ast_null_frame;
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}
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if (res < hdrlen) {
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ast_log(LOG_WARNING, "RTP Read too short\n");
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return &ast_null_frame;
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}
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if (rtp->nat) {
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/* Send to whoever sent to us */
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if ((rtp->rtcp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
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(rtp->rtcp->them.sin_port != sin.sin_port)) {
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memcpy(&rtp->rtcp->them, &sin, sizeof(rtp->rtcp->them));
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if (option_debug || rtpdebug)
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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));
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}
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}
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if (option_debug)
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ast_log(LOG_DEBUG, "Got RTCP report of %d bytes\n", res);
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return &ast_null_frame;
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}
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static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int timestamp, int mark)
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{
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struct timeval ts = ast_samp2tv( timestamp, 8000);
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if (ast_tvzero(rtp->rxcore) || mark) {
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rtp->rxcore = ast_tvsub(ast_tvnow(), ts);
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/* Round to 20ms for nice, pretty timestamps */
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rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 20000;
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}
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*tv = ast_tvadd(rtp->rxcore, ts);
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}
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struct ast_frame *ast_rtp_read(struct ast_rtp *rtp)
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{
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int res;
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struct sockaddr_in sin;
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socklen_t len;
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unsigned int seqno;
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int version;
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int payloadtype;
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int hdrlen = 12;
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int padding;
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int mark;
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int ext;
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int x;
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char iabuf[INET_ADDRSTRLEN];
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unsigned int timestamp;
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unsigned int *rtpheader;
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static struct ast_frame *f;
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struct rtpPayloadType rtpPT;
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len = sizeof(sin);
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/* Cache where the header will go */
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res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
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0, (struct sockaddr *)&sin, &len);
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rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
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if (res < 0) {
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if (errno != EAGAIN)
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ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
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if (errno == EBADF)
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CRASH;
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return &ast_null_frame;
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}
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if (res < hdrlen) {
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ast_log(LOG_WARNING, "RTP Read too short\n");
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return &ast_null_frame;
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}
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/* Ignore if the other side hasn't been given an address
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yet. */
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if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
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return &ast_null_frame;
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if (rtp->nat) {
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/* Send to whoever sent to us */
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if ((rtp->them.sin_addr.s_addr != sin.sin_addr.s_addr) ||
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(rtp->them.sin_port != sin.sin_port)) {
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memcpy(&rtp->them, &sin, sizeof(rtp->them));
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rtp->rxseqno = 0;
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ast_set_flag(rtp, FLAG_NAT_ACTIVE);
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if (option_debug || rtpdebug)
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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));
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}
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}
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/* Get fields */
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seqno = ntohl(rtpheader[0]);
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/* Check RTP version */
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version = (seqno & 0xC0000000) >> 30;
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if (version != 2)
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return &ast_null_frame;
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payloadtype = (seqno & 0x7f0000) >> 16;
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padding = seqno & (1 << 29);
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mark = seqno & (1 << 23);
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ext = seqno & (1 << 28);
|
|
seqno &= 0xffff;
|
|
timestamp = ntohl(rtpheader[1]);
|
|
|
|
if (padding) {
|
|
/* Remove padding bytes */
|
|
res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
|
|
}
|
|
|
|
if (ext) {
|
|
/* RTP Extension present */
|
|
hdrlen += 4;
|
|
hdrlen += (ntohl(rtpheader[3]) & 0xffff) << 2;
|
|
}
|
|
|
|
if (res < hdrlen) {
|
|
ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d)\n", res, hdrlen);
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
if(rtp_debug_test_addr(&sin))
|
|
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);
|
|
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 &ast_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 &ast_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 &ast_null_frame;
|
|
} else {
|
|
ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
|
|
return &ast_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"},
|
|
{{1, AST_FORMAT_H264}, "video", "H264"},
|
|
};
|
|
|
|
/* 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},
|
|
[99] = {1, AST_FORMAT_H264},
|
|
[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;
|
|
}
|
|
|
|
static void ast_rtp_pt_copy(struct ast_rtp *dest, struct ast_rtp *src)
|
|
{
|
|
int i;
|
|
/* Copy payload types from source to destination */
|
|
for (i=0; i < MAX_RTP_PT; ++i) {
|
|
dest->current_RTP_PT[i].isAstFormat =
|
|
src->current_RTP_PT[i].isAstFormat;
|
|
dest->current_RTP_PT[i].code =
|
|
src->current_RTP_PT[i].code;
|
|
}
|
|
dest->rtp_lookup_code_cache_isAstFormat = 0;
|
|
dest->rtp_lookup_code_cache_code = 0;
|
|
dest->rtp_lookup_code_cache_result = 0;
|
|
}
|
|
|
|
/*! \brief Get channel driver interface structure */
|
|
static struct ast_rtp_protocol *get_proto(struct ast_channel *chan)
|
|
{
|
|
struct ast_rtp_protocol *cur = NULL;
|
|
|
|
AST_LIST_LOCK(&protos);
|
|
AST_LIST_TRAVERSE(&protos, cur, list) {
|
|
if (cur->type == chan->tech->type)
|
|
break;
|
|
}
|
|
AST_LIST_UNLOCK(&protos);
|
|
|
|
return cur;
|
|
}
|
|
|
|
int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src)
|
|
{
|
|
struct ast_rtp *destp, *srcp; /* Audio RTP Channels */
|
|
struct ast_rtp *vdestp, *vsrcp; /* Video RTP channels */
|
|
struct ast_rtp_protocol *destpr, *srcpr;
|
|
/* Lock channels */
|
|
ast_mutex_lock(&dest->lock);
|
|
while(ast_mutex_trylock(&src->lock)) {
|
|
ast_mutex_unlock(&dest->lock);
|
|
usleep(1);
|
|
ast_mutex_lock(&dest->lock);
|
|
}
|
|
|
|
/* Find channel driver interfaces */
|
|
destpr = get_proto(dest);
|
|
srcpr = get_proto(src);
|
|
if (!destpr) {
|
|
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) {
|
|
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;
|
|
}
|
|
|
|
/* Get audio and video interface (if native bridge is possible) */
|
|
destp = destpr->get_rtp_info(dest);
|
|
if (destpr->get_vrtp_info)
|
|
vdestp = destpr->get_vrtp_info(dest);
|
|
else
|
|
vdestp = NULL;
|
|
srcp = srcpr->get_rtp_info(src);
|
|
if (srcpr->get_vrtp_info)
|
|
vsrcp = srcpr->get_vrtp_info(src);
|
|
else
|
|
vsrcp = NULL;
|
|
|
|
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
|
|
if (!destp || !srcp) {
|
|
/* Somebody doesn't want to play... */
|
|
ast_mutex_unlock(&dest->lock);
|
|
ast_mutex_unlock(&src->lock);
|
|
return 0;
|
|
}
|
|
ast_rtp_pt_copy(destp, srcp);
|
|
if (vdestp && vsrcp)
|
|
ast_rtp_pt_copy(vdestp, vsrcp);
|
|
ast_mutex_unlock(&dest->lock);
|
|
ast_mutex_unlock(&src->lock);
|
|
if (option_debug)
|
|
ast_log(LOG_DEBUG, "Seeded SDP of '%s' with that of '%s'\n", dest->name, src->name);
|
|
return 1;
|
|
}
|
|
|
|
/*! \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)
|
|
return; /* bogus payload type */
|
|
|
|
if (static_RTP_PT[pt].code != 0) {
|
|
rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
|
|
}
|
|
}
|
|
|
|
/*! \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;
|
|
|
|
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;
|
|
}
|
|
}
|
|
}
|
|
|
|
/*! \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;
|
|
|
|
*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;
|
|
}
|
|
}
|
|
}
|
|
|
|
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 */
|
|
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;
|
|
}
|
|
|
|
/*! \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;
|
|
|
|
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;
|
|
|
|
if (!(rtcp = ast_calloc(1, sizeof(*rtcp))))
|
|
return NULL;
|
|
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;
|
|
|
|
if (!(rtp = ast_calloc(1, sizeof(*rtp))))
|
|
return NULL;
|
|
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_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). */
|
|
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) {
|
|
close(rtp->rtcp->s);
|
|
free(rtp->rtcp);
|
|
}
|
|
free(rtp);
|
|
return NULL;
|
|
}
|
|
}
|
|
if (io && sched && callbackmode) {
|
|
/* 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);
|
|
}
|
|
ast_rtp_pt_default(rtp);
|
|
return rtp;
|
|
}
|
|
|
|
struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode)
|
|
{
|
|
struct in_addr ia;
|
|
|
|
memset(&ia, 0, sizeof(ia));
|
|
return ast_rtp_new_with_bindaddr(sched, io, rtcpenable, callbackmode, ia);
|
|
}
|
|
|
|
int ast_rtp_settos(struct ast_rtp *rtp, int tos)
|
|
{
|
|
int res;
|
|
|
|
if ((res = setsockopt(rtp->s, IPPROTO_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;
|
|
if (rtp->rtcp) {
|
|
rtp->rtcp->them.sin_port = htons(ntohs(them->sin_port) + 1);
|
|
rtp->rtcp->them.sin_addr = them->sin_addr;
|
|
}
|
|
rtp->rxseqno = 0;
|
|
}
|
|
|
|
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_stop(struct ast_rtp *rtp)
|
|
{
|
|
memset(&rtp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
|
|
memset(&rtp->them.sin_port, 0, sizeof(rtp->them.sin_port));
|
|
if (rtp->rtcp) {
|
|
memset(&rtp->rtcp->them.sin_addr, 0, sizeof(rtp->them.sin_addr));
|
|
memset(&rtp->rtcp->them.sin_port, 0, sizeof(rtp->them.sin_port));
|
|
}
|
|
}
|
|
|
|
void ast_rtp_reset(struct ast_rtp *rtp)
|
|
{
|
|
memset(&rtp->rxcore, 0, sizeof(rtp->rxcore));
|
|
memset(&rtp->txcore, 0, sizeof(rtp->txcore));
|
|
memset(&rtp->dtmfmute, 0, sizeof(rtp->dtmfmute));
|
|
rtp->lastts = 0;
|
|
rtp->lastdigitts = 0;
|
|
rtp->lastrxts = 0;
|
|
rtp->lastividtimestamp = 0;
|
|
rtp->lastovidtimestamp = 0;
|
|
rtp->lasteventseqn = 0;
|
|
rtp->lasteventendseqn = 0;
|
|
rtp->lasttxformat = 0;
|
|
rtp->lastrxformat = 0;
|
|
rtp->dtmfcount = 0;
|
|
rtp->dtmfduration = 0;
|
|
rtp->seqno = 0;
|
|
rtp->rxseqno = 0;
|
|
}
|
|
|
|
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);
|
|
if (rtp->rtcp) {
|
|
close(rtp->rtcp->s);
|
|
free(rtp->rtcp);
|
|
}
|
|
free(rtp);
|
|
}
|
|
|
|
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval *delivery)
|
|
{
|
|
struct timeval t;
|
|
long ms;
|
|
if (ast_tvzero(rtp->txcore)) {
|
|
rtp->txcore = ast_tvnow();
|
|
/* Round to 20ms for nice, pretty timestamps */
|
|
rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
|
|
}
|
|
/* Use previous txcore if available */
|
|
t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
|
|
ms = ast_tvdiff_ms(t, rtp->txcore);
|
|
if (ms < 0)
|
|
ms = 0;
|
|
/* Use what we just got for next time */
|
|
rtp->txcore = t;
|
|
return (unsigned int) ms;
|
|
}
|
|
|
|
int ast_rtp_senddigit(struct ast_rtp *rtp, char digit)
|
|
{
|
|
unsigned int *rtpheader;
|
|
int hdrlen = 12;
|
|
int res;
|
|
int x;
|
|
int payload;
|
|
char data[256];
|
|
char iabuf[INET_ADDRSTRLEN];
|
|
|
|
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;
|
|
}
|
|
payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_DTMF);
|
|
|
|
/* If we have no peer, return immediately */
|
|
if (!rtp->them.sin_addr.s_addr)
|
|
return 0;
|
|
|
|
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
|
|
|
|
/* Get a pointer to the header */
|
|
rtpheader = (unsigned int *)data;
|
|
rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
|
|
rtpheader[1] = htonl(rtp->lastdigitts);
|
|
rtpheader[2] = htonl(rtp->ssrc);
|
|
rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0));
|
|
for (x = 0; x < 6; x++) {
|
|
if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
|
|
res = sendto(rtp->s, (void *) rtpheader, hdrlen + 4, 0, (struct sockaddr *) &rtp->them, sizeof(rtp->them));
|
|
if (res < 0)
|
|
ast_log(LOG_ERROR, "RTP Transmission error to %s:%d: %s\n",
|
|
ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr),
|
|
ntohs(rtp->them.sin_port), strerror(errno));
|
|
if (rtp_debug_test_addr(&rtp->them))
|
|
ast_verbose("Sent RTP packet to %s:%d (type %d, seq %u, ts %u, len %u)\n",
|
|
ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr),
|
|
ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
|
|
}
|
|
/* Sequence number of last two end packets does not get incremented */
|
|
if (x < 3)
|
|
rtp->seqno++;
|
|
/* Clear marker bit and set seqno */
|
|
rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
|
|
/* For the last three packets, set the duration and the end bit */
|
|
if (x == 2) {
|
|
#if 0
|
|
/* No, this is wrong... Do not increment lastdigitts, that's not according
|
|
to the RFC, as best we can determine */
|
|
rtp->lastdigitts++; /* or else the SPA3000 will click instead of beeping... */
|
|
rtpheader[1] = htonl(rtp->lastdigitts);
|
|
#endif
|
|
/* Make duration 800 (100ms) */
|
|
rtpheader[3] |= htonl((800));
|
|
/* Set the End bit */
|
|
rtpheader[3] |= htonl((1 << 23));
|
|
}
|
|
}
|
|
/* Increment the digit timestamp by 120ms, to ensure that digits
|
|
sent sequentially with no intervening non-digit packets do not
|
|
get sent with the same timestamp, and that sequential digits
|
|
have some 'dead air' in between them
|
|
*/
|
|
rtp->lastdigitts += 960;
|
|
/* Increment the sequence number to reflect the last packet
|
|
that was sent
|
|
*/
|
|
rtp->seqno++;
|
|
return 0;
|
|
}
|
|
|
|
int ast_rtp_sendcng(struct ast_rtp *rtp, int level)
|
|
{
|
|
unsigned int *rtpheader;
|
|
int hdrlen = 12;
|
|
int res;
|
|
int payload;
|
|
char data[256];
|
|
char iabuf[INET_ADDRSTRLEN];
|
|
level = 127 - (level & 0x7f);
|
|
payload = ast_rtp_lookup_code(rtp, 0, AST_RTP_CN);
|
|
|
|
/* If we have no peer, return immediately */
|
|
if (!rtp->them.sin_addr.s_addr)
|
|
return 0;
|
|
|
|
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
|
|
|
|
/* Get a pointer to the header */
|
|
rtpheader = (unsigned int *)data;
|
|
rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno++));
|
|
rtpheader[1] = htonl(rtp->lastts);
|
|
rtpheader[2] = htonl(rtp->ssrc);
|
|
data[12] = level;
|
|
if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
|
|
res = sendto(rtp->s, (void *)rtpheader, hdrlen + 1, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
|
|
if (res <0)
|
|
ast_log(LOG_ERROR, "RTP Comfort Noise Transmission error to %s:%d: %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
|
|
if(rtp_debug_test_addr(&rtp->them))
|
|
ast_verbose("Sent Comfort Noise RTP packet to %s:%d (type %d, seq %d, ts %d, len %d)\n"
|
|
, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), payload, rtp->seqno, rtp->lastts,res - hdrlen);
|
|
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec)
|
|
{
|
|
unsigned char *rtpheader;
|
|
char iabuf[INET_ADDRSTRLEN];
|
|
int hdrlen = 12;
|
|
int res;
|
|
unsigned int ms;
|
|
int pred;
|
|
int mark = 0;
|
|
|
|
ms = calc_txstamp(rtp, &f->delivery);
|
|
/* Default prediction */
|
|
if (f->subclass < AST_FORMAT_MAX_AUDIO) {
|
|
pred = rtp->lastts + f->samples;
|
|
|
|
/* Re-calculate last TS */
|
|
rtp->lastts = rtp->lastts + ms * 8;
|
|
if (ast_tvzero(f->delivery)) {
|
|
/* If this isn't an absolute delivery time, Check if it is close to our prediction,
|
|
and if so, go with our prediction */
|
|
if (abs(rtp->lastts - pred) < MAX_TIMESTAMP_SKEW)
|
|
rtp->lastts = pred;
|
|
else {
|
|
if (option_debug > 2)
|
|
ast_log(LOG_DEBUG, "Difference is %d, ms is %d\n", abs(rtp->lastts - pred), ms);
|
|
mark = 1;
|
|
}
|
|
}
|
|
} else {
|
|
mark = f->subclass & 0x1;
|
|
pred = rtp->lastovidtimestamp + f->samples;
|
|
/* Re-calculate last TS */
|
|
rtp->lastts = rtp->lastts + ms * 90;
|
|
/* If it's close to our prediction, go for it */
|
|
if (ast_tvzero(f->delivery)) {
|
|
if (abs(rtp->lastts - pred) < 7200) {
|
|
rtp->lastts = pred;
|
|
rtp->lastovidtimestamp += f->samples;
|
|
} else {
|
|
if (option_debug > 2)
|
|
ast_log(LOG_DEBUG, "Difference is %d, ms is %d (%d), pred/ts/samples %d/%d/%d\n", abs(rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, f->samples);
|
|
rtp->lastovidtimestamp = rtp->lastts;
|
|
}
|
|
}
|
|
}
|
|
/* If the timestamp for non-digit packets has moved beyond the timestamp
|
|
for digits, update the digit timestamp.
|
|
*/
|
|
if (rtp->lastts > rtp->lastdigitts)
|
|
rtp->lastdigitts = rtp->lastts;
|
|
|
|
/* Get a pointer to the header */
|
|
rtpheader = (unsigned char *)(f->data - hdrlen);
|
|
|
|
put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (rtp->seqno) | (mark << 23)));
|
|
put_unaligned_uint32(rtpheader + 4, htonl(rtp->lastts));
|
|
put_unaligned_uint32(rtpheader + 8, htonl(rtp->ssrc));
|
|
|
|
if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
|
|
res = sendto(rtp->s, (void *)rtpheader, f->datalen + hdrlen, 0, (struct sockaddr *)&rtp->them, sizeof(rtp->them));
|
|
if (res <0) {
|
|
if (!rtp->nat || (rtp->nat && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
|
|
ast_log(LOG_DEBUG, "RTP Transmission error of packet %d to %s:%d: %s\n", rtp->seqno, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
|
|
} else if ((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) {
|
|
/* Only give this error message once if we are not RTP debugging */
|
|
if (option_debug || rtpdebug)
|
|
ast_log(LOG_DEBUG, "RTP NAT: Can't write RTP to private address %s:%d, waiting for other end to send audio...\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port));
|
|
ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
|
|
}
|
|
}
|
|
|
|
if(rtp_debug_test_addr(&rtp->them))
|
|
ast_verbose("Sent RTP packet to %s:%d (type %d, seq %u, ts %u, len %u)\n"
|
|
, ast_inet_ntoa(iabuf, sizeof(iabuf), rtp->them.sin_addr), ntohs(rtp->them.sin_port), codec, rtp->seqno, rtp->lastts,res - hdrlen);
|
|
}
|
|
|
|
rtp->seqno++;
|
|
|
|
return 0;
|
|
}
|
|
|
|
int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
|
|
{
|
|
struct ast_frame *f;
|
|
int codec;
|
|
int hdrlen = 12;
|
|
int subclass;
|
|
|
|
|
|
/* If we have no peer, return immediately */
|
|
if (!rtp->them.sin_addr.s_addr)
|
|
return 0;
|
|
|
|
/* If there is no data length, return immediately */
|
|
if (!_f->datalen)
|
|
return 0;
|
|
|
|
/* Make sure we have enough space for RTP header */
|
|
if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
|
|
ast_log(LOG_WARNING, "RTP can only send voice\n");
|
|
return -1;
|
|
}
|
|
|
|
subclass = _f->subclass;
|
|
if (_f->frametype == AST_FRAME_VIDEO)
|
|
subclass &= ~0x1;
|
|
|
|
codec = ast_rtp_lookup_code(rtp, 1, subclass);
|
|
if (codec < 0) {
|
|
ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n", ast_getformatname(_f->subclass));
|
|
return -1;
|
|
}
|
|
|
|
if (rtp->lasttxformat != subclass) {
|
|
/* New format, reset the smoother */
|
|
if (option_debug)
|
|
ast_log(LOG_DEBUG, "Ooh, format changed from %s to %s\n", ast_getformatname(rtp->lasttxformat), ast_getformatname(subclass));
|
|
rtp->lasttxformat = subclass;
|
|
if (rtp->smoother)
|
|
ast_smoother_free(rtp->smoother);
|
|
rtp->smoother = NULL;
|
|
}
|
|
|
|
|
|
switch(subclass) {
|
|
case AST_FORMAT_SLINEAR:
|
|
if (!rtp->smoother) {
|
|
rtp->smoother = ast_smoother_new(320);
|
|
}
|
|
if (!rtp->smoother) {
|
|
ast_log(LOG_WARNING, "Unable to create smoother :(\n");
|
|
return -1;
|
|
}
|
|
ast_smoother_feed_be(rtp->smoother, _f);
|
|
|
|
while((f = ast_smoother_read(rtp->smoother)))
|
|
ast_rtp_raw_write(rtp, f, codec);
|
|
break;
|
|
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_ADPCM:
|
|
case AST_FORMAT_G726:
|
|
if (!rtp->smoother) {
|
|
rtp->smoother = ast_smoother_new(80);
|
|
}
|
|
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_smoother_set_flags(rtp->smoother, AST_SMOOTHER_FLAG_G729);
|
|
}
|
|
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;
|
|
case AST_FORMAT_ILBC:
|
|
if (!rtp->smoother) {
|
|
rtp->smoother = ast_smoother_new(50);
|
|
}
|
|
if (!rtp->smoother) {
|
|
ast_log(LOG_WARNING, "Unable to create ILBC 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 %s packets\n", ast_getformatname(subclass));
|
|
/* fall through to... */
|
|
case AST_FORMAT_H261:
|
|
case AST_FORMAT_H263:
|
|
case AST_FORMAT_H263_PLUS:
|
|
case AST_FORMAT_H264:
|
|
case AST_FORMAT_G723_1:
|
|
case AST_FORMAT_LPC10:
|
|
case AST_FORMAT_SPEEX:
|
|
/* Don't buffer outgoing frames; send them one-per-packet: */
|
|
if (_f->offset < hdrlen) {
|
|
f = ast_frdup(_f);
|
|
} else {
|
|
f = _f;
|
|
}
|
|
ast_rtp_raw_write(rtp, f, codec);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Unregister interface to channel driver */
|
|
void ast_rtp_proto_unregister(struct ast_rtp_protocol *proto)
|
|
{
|
|
AST_LIST_LOCK(&protos);
|
|
AST_LIST_REMOVE(&protos, proto, list);
|
|
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;
|
|
|
|
AST_LIST_LOCK(&protos);
|
|
AST_LIST_TRAVERSE(&protos, cur, list) {
|
|
if (!strcmp(cur->type, proto->type)) {
|
|
ast_log(LOG_WARNING, "Tried to register same protocol '%s' twice\n", cur->type);
|
|
AST_LIST_UNLOCK(&protos);
|
|
return -1;
|
|
}
|
|
}
|
|
AST_LIST_INSERT_HEAD(&protos, proto, list);
|
|
AST_LIST_UNLOCK(&protos);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \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)
|
|
{
|
|
struct ast_frame *f;
|
|
struct ast_channel *who, *cs[3];
|
|
struct ast_rtp *p0, *p1; /* Audio RTP Channels */
|
|
struct ast_rtp *vp0, *vp1; /* Video RTP channels */
|
|
struct ast_rtp_protocol *pr0, *pr1;
|
|
struct sockaddr_in ac0, ac1;
|
|
struct sockaddr_in vac0, vac1;
|
|
struct sockaddr_in t0, t1;
|
|
struct sockaddr_in vt0, vt1;
|
|
char iabuf[INET_ADDRSTRLEN];
|
|
|
|
void *pvt0, *pvt1;
|
|
int codec0,codec1, oldcodec0, oldcodec1;
|
|
|
|
memset(&vt0, 0, sizeof(vt0));
|
|
memset(&vt1, 0, sizeof(vt1));
|
|
memset(&vac0, 0, sizeof(vac0));
|
|
memset(&vac1, 0, sizeof(vac1));
|
|
|
|
/* if need DTMF, cant native bridge */
|
|
if (flags & (AST_BRIDGE_DTMF_CHANNEL_0 | AST_BRIDGE_DTMF_CHANNEL_1))
|
|
return AST_BRIDGE_FAILED_NOWARN;
|
|
|
|
/* Lock channels */
|
|
ast_mutex_lock(&c0->lock);
|
|
while(ast_mutex_trylock(&c1->lock)) {
|
|
ast_mutex_unlock(&c0->lock);
|
|
usleep(1);
|
|
ast_mutex_lock(&c0->lock);
|
|
}
|
|
|
|
/* Find channel driver interfaces */
|
|
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_mutex_unlock(&c0->lock);
|
|
ast_mutex_unlock(&c1->lock);
|
|
return AST_BRIDGE_FAILED;
|
|
}
|
|
if (!pr1) {
|
|
ast_log(LOG_WARNING, "Can't find native functions for channel '%s'\n", c1->name);
|
|
ast_mutex_unlock(&c0->lock);
|
|
ast_mutex_unlock(&c1->lock);
|
|
return AST_BRIDGE_FAILED;
|
|
}
|
|
|
|
/* Get channel specific interface structures */
|
|
pvt0 = c0->tech_pvt;
|
|
pvt1 = c1->tech_pvt;
|
|
|
|
/* Get audio and video interface (if native bridge is possible) */
|
|
p0 = pr0->get_rtp_info(c0);
|
|
if (pr0->get_vrtp_info)
|
|
vp0 = pr0->get_vrtp_info(c0);
|
|
else
|
|
vp0 = NULL;
|
|
p1 = pr1->get_rtp_info(c1);
|
|
if (pr1->get_vrtp_info)
|
|
vp1 = pr1->get_vrtp_info(c1);
|
|
else
|
|
vp1 = NULL;
|
|
|
|
/* Check if bridge is still possible (In SIP canreinvite=no stops this, like NAT) */
|
|
if (!p0 || !p1) {
|
|
/* Somebody doesn't want to play... */
|
|
ast_mutex_unlock(&c0->lock);
|
|
ast_mutex_unlock(&c1->lock);
|
|
return AST_BRIDGE_FAILED_NOWARN;
|
|
}
|
|
/* Get codecs from both sides */
|
|
if (pr0->get_codec)
|
|
codec0 = pr0->get_codec(c0);
|
|
else
|
|
codec0 = 0;
|
|
if (pr1->get_codec)
|
|
codec1 = pr1->get_codec(c1);
|
|
else
|
|
codec1 = 0;
|
|
if (pr0->get_codec && pr1->get_codec) {
|
|
/* Hey, we can't do reinvite if both parties speak different codecs */
|
|
if (!(codec0 & codec1)) {
|
|
if (option_debug)
|
|
ast_log(LOG_DEBUG, "Channel codec0 = %d is not codec1 = %d, cannot native bridge in RTP.\n", codec0, codec1);
|
|
ast_mutex_unlock(&c0->lock);
|
|
ast_mutex_unlock(&c1->lock);
|
|
return AST_BRIDGE_FAILED_NOWARN;
|
|
}
|
|
}
|
|
|
|
if (option_verbose > 2)
|
|
ast_verbose(VERBOSE_PREFIX_3 "Native bridging %s and %s\n", c0->name, c1->name);
|
|
|
|
/* Ok, we should be able to redirect the media. Start with one channel */
|
|
if (pr0->set_rtp_peer(c0, p1, vp1, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to talk to '%s'\n", c0->name, c1->name);
|
|
else {
|
|
/* Store RTP peer */
|
|
ast_rtp_get_peer(p1, &ac1);
|
|
if (vp1)
|
|
ast_rtp_get_peer(vp1, &vac1);
|
|
}
|
|
/* Then test the other channel */
|
|
if (pr1->set_rtp_peer(c1, p0, vp0, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to talk back to '%s'\n", c1->name, c0->name);
|
|
else {
|
|
/* Store RTP peer */
|
|
ast_rtp_get_peer(p0, &ac0);
|
|
if (vp0)
|
|
ast_rtp_get_peer(vp0, &vac0);
|
|
}
|
|
ast_mutex_unlock(&c0->lock);
|
|
ast_mutex_unlock(&c1->lock);
|
|
/* External RTP Bridge up, now loop and see if something happes that force us to take the
|
|
media back to Asterisk */
|
|
cs[0] = c0;
|
|
cs[1] = c1;
|
|
cs[2] = NULL;
|
|
oldcodec0 = codec0;
|
|
oldcodec1 = codec1;
|
|
for (;;) {
|
|
/* Check if something changed... */
|
|
if ((c0->tech_pvt != pvt0) ||
|
|
(c1->tech_pvt != pvt1) ||
|
|
(c0->masq || c0->masqr || c1->masq || c1->masqr)) {
|
|
ast_log(LOG_DEBUG, "Oooh, something is weird, backing out\n");
|
|
if (c0->tech_pvt == pvt0) {
|
|
if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
|
|
}
|
|
if (c1->tech_pvt == pvt1) {
|
|
if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
|
|
}
|
|
return AST_BRIDGE_RETRY;
|
|
}
|
|
/* Now check if they have changed address */
|
|
ast_rtp_get_peer(p1, &t1);
|
|
ast_rtp_get_peer(p0, &t0);
|
|
if (pr0->get_codec)
|
|
codec0 = pr0->get_codec(c0);
|
|
if (pr1->get_codec)
|
|
codec1 = pr1->get_codec(c1);
|
|
if (vp1)
|
|
ast_rtp_get_peer(vp1, &vt1);
|
|
if (vp0)
|
|
ast_rtp_get_peer(vp0, &vt0);
|
|
if (inaddrcmp(&t1, &ac1) || (vp1 && inaddrcmp(&vt1, &vac1)) || (codec1 != oldcodec1)) {
|
|
if (option_debug > 1) {
|
|
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
|
|
c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t1.sin_addr), ntohs(t1.sin_port), codec1);
|
|
ast_log(LOG_DEBUG, "Oooh, '%s' changed end vaddress to %s:%d (format %d)\n",
|
|
c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vt1.sin_addr), ntohs(vt1.sin_port), codec1);
|
|
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
|
|
c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), ac1.sin_addr), ntohs(ac1.sin_port), oldcodec1);
|
|
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
|
|
c1->name, ast_inet_ntoa(iabuf, sizeof(iabuf), vac1.sin_addr), ntohs(vac1.sin_port), oldcodec1);
|
|
}
|
|
if (pr0->set_rtp_peer(c0, t1.sin_addr.s_addr ? p1 : NULL, vt1.sin_addr.s_addr ? vp1 : NULL, codec1, ast_test_flag(p1, FLAG_NAT_ACTIVE)))
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c0->name, c1->name);
|
|
memcpy(&ac1, &t1, sizeof(ac1));
|
|
memcpy(&vac1, &vt1, sizeof(vac1));
|
|
oldcodec1 = codec1;
|
|
}
|
|
if (inaddrcmp(&t0, &ac0) || (vp0 && inaddrcmp(&vt0, &vac0))) {
|
|
if (option_debug) {
|
|
ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
|
|
c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), t0.sin_addr), ntohs(t0.sin_port), codec0);
|
|
ast_log(LOG_DEBUG, "Oooh, '%s' was %s:%d/(format %d)\n",
|
|
c0->name, ast_inet_ntoa(iabuf, sizeof(iabuf), ac0.sin_addr), ntohs(ac0.sin_port), oldcodec0);
|
|
}
|
|
if (pr1->set_rtp_peer(c1, t0.sin_addr.s_addr ? p0 : NULL, vt0.sin_addr.s_addr ? vp0 : NULL, codec0, ast_test_flag(p0, FLAG_NAT_ACTIVE)))
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to update to '%s'\n", c1->name, c0->name);
|
|
memcpy(&ac0, &t0, sizeof(ac0));
|
|
memcpy(&vac0, &vt0, sizeof(vac0));
|
|
oldcodec0 = codec0;
|
|
}
|
|
who = ast_waitfor_n(cs, 2, &timeoutms);
|
|
if (!who) {
|
|
if (!timeoutms)
|
|
return AST_BRIDGE_RETRY;
|
|
if (option_debug)
|
|
ast_log(LOG_DEBUG, "Ooh, empty read...\n");
|
|
/* check for hangup / whentohangup */
|
|
if (ast_check_hangup(c0) || ast_check_hangup(c1))
|
|
break;
|
|
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;
|
|
if (option_debug)
|
|
ast_log(LOG_DEBUG, "Oooh, got a %s\n", f ? "digit" : "hangup");
|
|
if ((c0->tech_pvt == pvt0)) {
|
|
if (pr0->set_rtp_peer(c0, NULL, NULL, 0, 0))
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c0->name);
|
|
}
|
|
if ((c1->tech_pvt == pvt1)) {
|
|
if (pr1->set_rtp_peer(c1, NULL, NULL, 0, 0))
|
|
ast_log(LOG_WARNING, "Channel '%s' failed to break RTP bridge\n", c1->name);
|
|
}
|
|
return AST_BRIDGE_COMPLETE;
|
|
} else if ((f->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
|
|
if ((f->subclass == AST_CONTROL_HOLD) || (f->subclass == AST_CONTROL_UNHOLD) ||
|
|
(f->subclass == AST_CONTROL_VIDUPDATE)) {
|
|
ast_indicate(who == c0 ? c1 : c0, f->subclass);
|
|
ast_frfree(f);
|
|
} else {
|
|
*fo = f;
|
|
*rc = who;
|
|
ast_log(LOG_DEBUG, "Got a FRAME_CONTROL (%d) frame on channel %s\n", f->subclass, who->name);
|
|
return AST_BRIDGE_COMPLETE;
|
|
}
|
|
} else {
|
|
if ((f->frametype == AST_FRAME_DTMF) ||
|
|
(f->frametype == AST_FRAME_VOICE) ||
|
|
(f->frametype == AST_FRAME_VIDEO)) {
|
|
/* Forward voice or DTMF frames if they happen upon us */
|
|
if (who == c0) {
|
|
ast_write(c1, f);
|
|
} else if (who == c1) {
|
|
ast_write(c0, f);
|
|
}
|
|
}
|
|
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 AST_BRIDGE_FAILED;
|
|
}
|
|
|
|
static int rtp_do_debug_ip(int fd, int argc, char *argv[])
|
|
{
|
|
struct hostent *hp;
|
|
struct ast_hostent ahp;
|
|
char iabuf[INET_ADDRSTRLEN];
|
|
int port = 0;
|
|
char *p, *arg;
|
|
|
|
if (argc != 4)
|
|
return RESULT_SHOWUSAGE;
|
|
arg = argv[3];
|
|
p = strstr(arg, ":");
|
|
if (p) {
|
|
*p = '\0';
|
|
p++;
|
|
port = atoi(p);
|
|
}
|
|
hp = ast_gethostbyname(arg, &ahp);
|
|
if (hp == NULL)
|
|
return RESULT_SHOWUSAGE;
|
|
rtpdebugaddr.sin_family = AF_INET;
|
|
memcpy(&rtpdebugaddr.sin_addr, hp->h_addr, sizeof(rtpdebugaddr.sin_addr));
|
|
rtpdebugaddr.sin_port = htons(port);
|
|
if (port == 0)
|
|
ast_cli(fd, "RTP Debugging Enabled for IP: %s\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtpdebugaddr.sin_addr));
|
|
else
|
|
ast_cli(fd, "RTP Debugging Enabled for IP: %s:%d\n", ast_inet_ntoa(iabuf, sizeof(iabuf), rtpdebugaddr.sin_addr), port);
|
|
rtpdebug = 1;
|
|
return RESULT_SUCCESS;
|
|
}
|
|
|
|
static int rtp_do_debug(int fd, int argc, char *argv[])
|
|
{
|
|
if(argc != 2) {
|
|
if(argc != 4)
|
|
return RESULT_SHOWUSAGE;
|
|
return rtp_do_debug_ip(fd, argc, argv);
|
|
}
|
|
rtpdebug = 1;
|
|
memset(&rtpdebugaddr,0,sizeof(rtpdebugaddr));
|
|
ast_cli(fd, "RTP Debugging Enabled\n");
|
|
return RESULT_SUCCESS;
|
|
}
|
|
|
|
static int rtp_no_debug(int fd, int argc, char *argv[])
|
|
{
|
|
if(argc !=3)
|
|
return RESULT_SHOWUSAGE;
|
|
rtpdebug = 0;
|
|
ast_cli(fd,"RTP Debugging Disabled\n");
|
|
return RESULT_SUCCESS;
|
|
}
|
|
|
|
static char debug_usage[] =
|
|
"Usage: rtp debug [ip host[:port]]\n"
|
|
" Enable dumping of all RTP packets to and from host.\n";
|
|
|
|
static char no_debug_usage[] =
|
|
"Usage: rtp no debug\n"
|
|
" Disable all RTP debugging\n";
|
|
|
|
static struct ast_cli_entry cli_debug_ip =
|
|
{{ "rtp", "debug", "ip", NULL } , rtp_do_debug, "Enable RTP debugging on IP", debug_usage };
|
|
|
|
static struct ast_cli_entry cli_debug =
|
|
{{ "rtp", "debug", NULL } , rtp_do_debug, "Enable RTP debugging", debug_usage };
|
|
|
|
static struct ast_cli_entry cli_no_debug =
|
|
{{ "rtp", "no", "debug", NULL } , rtp_no_debug, "Disable RTP debugging", no_debug_usage };
|
|
|
|
int ast_rtp_reload(void)
|
|
{
|
|
struct ast_config *cfg;
|
|
char *s;
|
|
|
|
rtpstart = 5000;
|
|
rtpend = 31000;
|
|
dtmftimeout = DEFAULT_DTMF_TIMEOUT;
|
|
cfg = ast_config_load("rtp.conf");
|
|
if (cfg) {
|
|
if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
|
|
rtpstart = atoi(s);
|
|
if (rtpstart < 1024)
|
|
rtpstart = 1024;
|
|
if (rtpstart > 65535)
|
|
rtpstart = 65535;
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
|
|
rtpend = atoi(s);
|
|
if (rtpend < 1024)
|
|
rtpend = 1024;
|
|
if (rtpend > 65535)
|
|
rtpend = 65535;
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
|
|
#ifdef SO_NO_CHECK
|
|
if (ast_false(s))
|
|
nochecksums = 1;
|
|
else
|
|
nochecksums = 0;
|
|
#else
|
|
if (ast_false(s))
|
|
ast_log(LOG_WARNING, "Disabling RTP checksums is not supported on this operating system!\n");
|
|
#endif
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) {
|
|
dtmftimeout = atoi(s);
|
|
if ((dtmftimeout < 0) || (dtmftimeout > 20000)) {
|
|
ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
|
|
dtmftimeout, DEFAULT_DTMF_TIMEOUT);
|
|
dtmftimeout = DEFAULT_DTMF_TIMEOUT;
|
|
};
|
|
}
|
|
ast_config_destroy(cfg);
|
|
}
|
|
if (rtpstart >= rtpend) {
|
|
ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
|
|
rtpstart = 5000;
|
|
rtpend = 31000;
|
|
}
|
|
if (option_verbose > 1)
|
|
ast_verbose(VERBOSE_PREFIX_2 "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Initialize the RTP system in Asterisk */
|
|
void ast_rtp_init(void)
|
|
{
|
|
ast_cli_register(&cli_debug);
|
|
ast_cli_register(&cli_debug_ip);
|
|
ast_cli_register(&cli_no_debug);
|
|
ast_rtp_reload();
|
|
}
|