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5436 lines
178 KiB
5436 lines
178 KiB
/*
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* Asterisk -- An open source telephony toolkit.
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*
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* Copyright (C) 1999 - 2008, 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 defined in RFC 3550.
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*
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* \ingroup rtp_engines
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*/
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/*** MODULEINFO
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<use type="external">pjproject</use>
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<support_level>core</support_level>
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***/
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#include "asterisk.h"
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ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
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#include <sys/time.h>
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#include <signal.h>
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#include <fcntl.h>
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#ifdef HAVE_OPENSSL_SRTP
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#include <openssl/ssl.h>
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#include <openssl/err.h>
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#include <openssl/bio.h>
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#endif
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#ifdef HAVE_PJPROJECT
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#include <pjlib.h>
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#include <pjlib-util.h>
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#include <pjnath.h>
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#endif
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#include "asterisk/stun.h"
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#include "asterisk/pbx.h"
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#include "asterisk/frame.h"
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#include "asterisk/format_cache.h"
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#include "asterisk/channel.h"
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#include "asterisk/acl.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/manager.h"
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#include "asterisk/unaligned.h"
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#include "asterisk/module.h"
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#include "asterisk/rtp_engine.h"
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#include "asterisk/smoother.h"
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#include "asterisk/test.h"
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#define MAX_TIMESTAMP_SKEW 640
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#define RTP_SEQ_MOD (1<<16) /*!< A sequence number can't be more than 16 bits */
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#define RTCP_DEFAULT_INTERVALMS 5000 /*!< Default milli-seconds between RTCP reports we send */
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#define RTCP_MIN_INTERVALMS 500 /*!< Min milli-seconds between RTCP reports we send */
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#define RTCP_MAX_INTERVALMS 60000 /*!< Max milli-seconds between RTCP reports we send */
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#define DEFAULT_RTP_START 5000 /*!< Default port number to start allocating RTP ports from */
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#define DEFAULT_RTP_END 31000 /*!< Default maximum port number to end allocating RTP ports at */
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#define MINIMUM_RTP_PORT 1024 /*!< Minimum port number to accept */
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#define MAXIMUM_RTP_PORT 65535 /*!< Maximum port number to accept */
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#define DEFAULT_TURN_PORT 3478
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#define TURN_STATE_WAIT_TIME 2000
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#define RTCP_PT_FUR 192
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#define RTCP_PT_SR AST_RTP_RTCP_SR
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#define RTCP_PT_RR AST_RTP_RTCP_RR
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#define RTCP_PT_SDES 202
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#define RTCP_PT_BYE 203
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#define RTCP_PT_APP 204
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/* VP8: RTCP Feedback */
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#define RTCP_PT_PSFB 206
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#define RTP_MTU 1200
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#define DTMF_SAMPLE_RATE_MS 8 /*!< DTMF samples per millisecond */
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#define DEFAULT_DTMF_TIMEOUT (150 * (8000 / 1000)) /*!< samples */
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#define ZFONE_PROFILE_ID 0x505a
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#define DEFAULT_LEARNING_MIN_SEQUENTIAL 4
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#define SRTP_MASTER_KEY_LEN 16
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#define SRTP_MASTER_SALT_LEN 14
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#define SRTP_MASTER_LEN (SRTP_MASTER_KEY_LEN + SRTP_MASTER_SALT_LEN)
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enum strict_rtp_state {
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STRICT_RTP_OPEN = 0, /*! No RTP packets should be dropped, all sources accepted */
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STRICT_RTP_LEARN, /*! Accept next packet as source */
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STRICT_RTP_CLOSED, /*! Drop all RTP packets not coming from source that was learned */
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};
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#define DEFAULT_STRICT_RTP STRICT_RTP_CLOSED
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#define DEFAULT_ICESUPPORT 1
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extern struct ast_srtp_res *res_srtp;
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extern struct ast_srtp_policy_res *res_srtp_policy;
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static int dtmftimeout = DEFAULT_DTMF_TIMEOUT;
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static int rtpstart = DEFAULT_RTP_START; /*!< First port for RTP sessions (set in rtp.conf) */
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static int rtpend = DEFAULT_RTP_END; /*!< Last port for RTP sessions (set in rtp.conf) */
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static int rtpdebug; /*!< Are we debugging? */
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static int rtcpdebug; /*!< Are we debugging RTCP? */
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static int rtcpstats; /*!< Are we debugging RTCP? */
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static int rtcpinterval = RTCP_DEFAULT_INTERVALMS; /*!< Time between rtcp reports in millisecs */
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static struct ast_sockaddr rtpdebugaddr; /*!< Debug packets to/from this host */
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static struct ast_sockaddr rtcpdebugaddr; /*!< Debug RTCP packets to/from this host */
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static int rtpdebugport; /*< Debug only RTP packets from IP or IP+Port if port is > 0 */
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static int rtcpdebugport; /*< Debug only RTCP packets from IP or IP+Port if port is > 0 */
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#ifdef SO_NO_CHECK
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static int nochecksums;
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#endif
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static int strictrtp = DEFAULT_STRICT_RTP; /*< Only accept RTP frames from a defined source. If we receive an indication of a changing source, enter learning mode. */
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static int learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL; /*< Number of sequential RTP frames needed from a single source during learning mode to accept new source. */
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#ifdef HAVE_PJPROJECT
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static int icesupport = DEFAULT_ICESUPPORT;
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static struct sockaddr_in stunaddr;
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static pj_str_t turnaddr;
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static int turnport = DEFAULT_TURN_PORT;
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static pj_str_t turnusername;
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static pj_str_t turnpassword;
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/*! \brief Pool factory used by pjlib to allocate memory. */
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static pj_caching_pool cachingpool;
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/*! \brief Global memory pool for configuration and timers */
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static pj_pool_t *pool;
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/*! \brief Global timer heap */
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static pj_timer_heap_t *timer_heap;
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/*! \brief Thread executing the timer heap */
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static pj_thread_t *timer_thread;
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/*! \brief Used to tell the timer thread to terminate */
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static int timer_terminate;
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/*! \brief Structure which contains ioqueue thread information */
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struct ast_rtp_ioqueue_thread {
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/*! \brief Pool used by the thread */
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pj_pool_t *pool;
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/*! \brief The thread handling the queue and timer heap */
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pj_thread_t *thread;
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/*! \brief Ioqueue which polls on sockets */
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pj_ioqueue_t *ioqueue;
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/*! \brief Timer heap for scheduled items */
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pj_timer_heap_t *timerheap;
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/*! \brief Termination request */
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int terminate;
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/*! \brief Current number of descriptors being waited on */
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unsigned int count;
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/*! \brief Linked list information */
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AST_LIST_ENTRY(ast_rtp_ioqueue_thread) next;
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};
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/*! \brief List of ioqueue threads */
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static AST_LIST_HEAD_STATIC(ioqueues, ast_rtp_ioqueue_thread);
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#endif
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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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#define FLAG_NEED_MARKER_BIT (1 << 3)
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#define FLAG_DTMF_COMPENSATE (1 << 4)
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#define TRANSPORT_SOCKET_RTP 0
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#define TRANSPORT_SOCKET_RTCP 1
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#define TRANSPORT_TURN_RTP 2
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#define TRANSPORT_TURN_RTCP 3
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/*! \brief RTP learning mode tracking information */
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struct rtp_learning_info {
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int max_seq; /*!< The highest sequence number received */
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int packets; /*!< The number of remaining packets before the source is accepted */
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};
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#ifdef HAVE_OPENSSL_SRTP
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struct dtls_details {
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ast_mutex_t lock; /*!< Lock for timeout timer synchronization */
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SSL *ssl; /*!< SSL session */
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BIO *read_bio; /*!< Memory buffer for reading */
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BIO *write_bio; /*!< Memory buffer for writing */
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enum ast_rtp_dtls_setup dtls_setup; /*!< Current setup state */
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enum ast_rtp_dtls_connection connection; /*!< Whether this is a new or existing connection */
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int timeout_timer; /*!< Scheduler id for timeout timer */
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};
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#endif
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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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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 themssrc; /*!< Their SSRC */
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unsigned int rxssrc;
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unsigned int lastts;
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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 lastitexttimestamp;
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unsigned int lastotexttimestamp;
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unsigned int lasteventseqn;
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int lastrxseqno; /*!< Last received sequence number */
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unsigned short seedrxseqno; /*!< What sequence number did they start with?*/
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unsigned int seedrxts; /*!< What RTP timestamp did they start with? */
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unsigned int rxcount; /*!< How many packets have we received? */
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unsigned int rxoctetcount; /*!< How many octets have we received? should be rxcount *160*/
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unsigned int txcount; /*!< How many packets have we sent? */
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unsigned int txoctetcount; /*!< How many octets have we sent? (txcount*160)*/
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unsigned int cycles; /*!< Shifted count of sequence number cycles */
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double rxjitter; /*!< Interarrival jitter at the moment in seconds */
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double rxtransit; /*!< Relative transit time for previous packet */
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struct ast_format *lasttxformat;
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struct ast_format *lastrxformat;
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int rtptimeout; /*!< RTP timeout time (negative or zero means disabled, negative value means temporarily disabled) */
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int rtpholdtimeout; /*!< RTP timeout when on hold (negative or zero means disabled, negative value means temporarily disabled). */
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int rtpkeepalive; /*!< Send RTP comfort noice packets for keepalive */
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/* DTMF Reception Variables */
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char resp; /*!< The current digit being processed */
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unsigned int last_seqno; /*!< The last known sequence number for any DTMF packet */
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unsigned int last_end_timestamp; /*!< The last known timestamp received from an END packet */
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unsigned int dtmf_duration; /*!< Total duration in samples since the digit start event */
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unsigned int dtmf_timeout; /*!< When this timestamp is reached we consider END frame lost and forcibly abort digit */
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unsigned int dtmfsamples;
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enum ast_rtp_dtmf_mode dtmfmode; /*!< The current DTMF mode of the RTP stream */
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/* DTMF Transmission Variables */
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unsigned int lastdigitts;
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char sending_digit; /*!< boolean - are we sending digits */
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char send_digit; /*!< digit we are sending */
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int send_payload;
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int send_duration;
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unsigned int flags;
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struct timeval rxcore;
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struct timeval txcore;
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double drxcore; /*!< The double representation of the first received packet */
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struct timeval lastrx; /*!< timeval when we last received a packet */
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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 ast_sched_context *sched;
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struct io_context *io;
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void *data;
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struct ast_rtcp *rtcp;
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struct ast_rtp *bridged; /*!< Who we are Packet bridged to */
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enum strict_rtp_state strict_rtp_state; /*!< Current state that strict RTP protection is in */
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struct ast_sockaddr strict_rtp_address; /*!< Remote address information for strict RTP purposes */
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/*
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* Learning mode values based on pjmedia's probation mode. Many of these values are redundant to the above,
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* but these are in place to keep learning mode sequence values sealed from their normal counterparts.
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*/
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struct rtp_learning_info rtp_source_learn; /* Learning mode track for the expected RTP source */
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struct rtp_learning_info alt_source_learn; /* Learning mode tracking for a new RTP source after one has been chosen */
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struct rtp_red *red;
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ast_mutex_t lock; /*!< Lock for synchronization purposes */
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ast_cond_t cond; /*!< Condition for signaling */
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#ifdef HAVE_PJPROJECT
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pj_ice_sess *ice; /*!< ICE session */
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pj_turn_sock *turn_rtp; /*!< RTP TURN relay */
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pj_turn_sock *turn_rtcp; /*!< RTCP TURN relay */
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pj_turn_state_t turn_state; /*!< Current state of the TURN relay session */
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unsigned int passthrough:1; /*!< Bit to indicate that the received packet should be passed through */
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unsigned int rtp_passthrough:1; /*!< Bit to indicate that TURN RTP should be passed through */
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unsigned int rtcp_passthrough:1; /*!< Bit to indicate that TURN RTCP should be passed through */
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unsigned int ice_port; /*!< Port that ICE was started with if it was previously started */
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struct ast_sockaddr rtp_loop; /*!< Loopback address for forwarding RTP from TURN */
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struct ast_sockaddr rtcp_loop; /*!< Loopback address for forwarding RTCP from TURN */
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struct ast_rtp_ioqueue_thread *ioqueue; /*!< The ioqueue thread handling us */
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char remote_ufrag[256]; /*!< The remote ICE username */
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char remote_passwd[256]; /*!< The remote ICE password */
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char local_ufrag[256]; /*!< The local ICE username */
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char local_passwd[256]; /*!< The local ICE password */
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struct ao2_container *ice_local_candidates; /*!< The local ICE candidates */
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struct ao2_container *ice_active_remote_candidates; /*!< The remote ICE candidates */
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struct ao2_container *ice_proposed_remote_candidates; /*!< Incoming remote ICE candidates for new session */
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struct ast_sockaddr ice_original_rtp_addr; /*!< rtp address that ICE started on first session */
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#endif
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#ifdef HAVE_OPENSSL_SRTP
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SSL_CTX *ssl_ctx; /*!< SSL context */
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enum ast_rtp_dtls_verify dtls_verify; /*!< What to verify */
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enum ast_srtp_suite suite; /*!< SRTP crypto suite */
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enum ast_rtp_dtls_hash local_hash; /*!< Local hash used for the fingerprint */
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char local_fingerprint[160]; /*!< Fingerprint of our certificate */
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enum ast_rtp_dtls_hash remote_hash; /*!< Remote hash used for the fingerprint */
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unsigned char remote_fingerprint[EVP_MAX_MD_SIZE]; /*!< Fingerprint of the peer certificate */
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unsigned int rekey; /*!< Interval at which to renegotiate and rekey */
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int rekeyid; /*!< Scheduled item id for rekeying */
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struct dtls_details dtls; /*!< DTLS state information */
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#endif
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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 rtcp_info;
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int s; /*!< Socket */
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struct ast_sockaddr us; /*!< Socket representation of the local endpoint. */
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struct ast_sockaddr them; /*!< Socket representation of the remote endpoint. */
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unsigned int soc; /*!< What they told us */
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unsigned int spc; /*!< What they told us */
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unsigned int themrxlsr; /*!< The middle 32 bits of the NTP timestamp in the last received SR*/
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struct timeval rxlsr; /*!< Time when we got their last SR */
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struct timeval txlsr; /*!< Time when we sent or last SR*/
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unsigned int expected_prior; /*!< no. packets in previous interval */
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unsigned int received_prior; /*!< no. packets received in previous interval */
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int schedid; /*!< Schedid returned from ast_sched_add() to schedule RTCP-transmissions*/
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unsigned int rr_count; /*!< number of RRs we've sent, not including report blocks in SR's */
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unsigned int sr_count; /*!< number of SRs we've sent */
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unsigned int lastsrtxcount; /*!< Transmit packet count when last SR sent */
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double accumulated_transit; /*!< accumulated a-dlsr-lsr */
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double rtt; /*!< Last reported rtt */
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unsigned int reported_jitter; /*!< The contents of their last jitter entry in the RR */
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unsigned int reported_lost; /*!< Reported lost packets in their RR */
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double reported_maxjitter;
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double reported_minjitter;
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double reported_normdev_jitter;
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double reported_stdev_jitter;
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unsigned int reported_jitter_count;
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double reported_maxlost;
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double reported_minlost;
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double reported_normdev_lost;
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double reported_stdev_lost;
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double rxlost;
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double maxrxlost;
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double minrxlost;
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double normdev_rxlost;
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double stdev_rxlost;
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unsigned int rxlost_count;
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double maxrxjitter;
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double minrxjitter;
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double normdev_rxjitter;
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double stdev_rxjitter;
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unsigned int rxjitter_count;
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double maxrtt;
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double minrtt;
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double normdevrtt;
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double stdevrtt;
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unsigned int rtt_count;
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/* VP8: sequence number for the RTCP FIR FCI */
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int firseq;
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#ifdef HAVE_OPENSSL_SRTP
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struct dtls_details dtls; /*!< DTLS state information */
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#endif
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};
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struct rtp_red {
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struct ast_frame t140; /*!< Primary data */
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struct ast_frame t140red; /*!< Redundant t140*/
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unsigned char pt[AST_RED_MAX_GENERATION]; /*!< Payload types for redundancy data */
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unsigned char ts[AST_RED_MAX_GENERATION]; /*!< Time stamps */
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unsigned char len[AST_RED_MAX_GENERATION]; /*!< length of each generation */
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int num_gen; /*!< Number of generations */
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int schedid; /*!< Timer id */
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int ti; /*!< How long to buffer data before send */
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unsigned char t140red_data[64000];
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unsigned char buf_data[64000]; /*!< buffered primary data */
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int hdrlen;
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long int prev_ts;
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};
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AST_LIST_HEAD_NOLOCK(frame_list, ast_frame);
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/* Forward Declarations */
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static int ast_rtp_new(struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data);
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static int ast_rtp_destroy(struct ast_rtp_instance *instance);
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static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit);
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static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit);
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static int ast_rtp_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration);
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static int ast_rtp_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode);
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static enum ast_rtp_dtmf_mode ast_rtp_dtmf_mode_get(struct ast_rtp_instance *instance);
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static void ast_rtp_update_source(struct ast_rtp_instance *instance);
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static void ast_rtp_change_source(struct ast_rtp_instance *instance);
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static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame);
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static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp);
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static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value);
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static int ast_rtp_fd(struct ast_rtp_instance *instance, int rtcp);
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static void ast_rtp_remote_address_set(struct ast_rtp_instance *instance, struct ast_sockaddr *addr);
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|
static int rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations);
|
|
static int rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame);
|
|
static int ast_rtp_local_bridge(struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1);
|
|
static int ast_rtp_get_stat(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat);
|
|
static int ast_rtp_dtmf_compatible(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1);
|
|
static void ast_rtp_stun_request(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username);
|
|
static void ast_rtp_stop(struct ast_rtp_instance *instance);
|
|
static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char* desc);
|
|
static int ast_rtp_sendcng(struct ast_rtp_instance *instance, int level);
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
static int ast_rtp_activate(struct ast_rtp_instance *instance);
|
|
static void dtls_srtp_check_pending(struct ast_rtp_instance *instance, struct ast_rtp *rtp, int rtcp);
|
|
static void dtls_srtp_start_timeout_timer(struct ast_rtp_instance *instance, struct ast_rtp *rtp, int rtcp);
|
|
static void dtls_srtp_stop_timeout_timer(struct ast_rtp_instance *instance, struct ast_rtp *rtp, int rtcp);
|
|
#endif
|
|
|
|
static int __rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int rtcp, int *ice, int use_srtp);
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
/*! \brief Helper function which updates an ast_sockaddr with the candidate used for the component */
|
|
static void update_address_with_ice_candidate(struct ast_rtp *rtp, enum ast_rtp_ice_component_type component,
|
|
struct ast_sockaddr *cand_address)
|
|
{
|
|
char address[PJ_INET6_ADDRSTRLEN];
|
|
|
|
if (!rtp->ice || (component < 1) || !rtp->ice->comp[component - 1].valid_check) {
|
|
return;
|
|
}
|
|
|
|
ast_sockaddr_parse(cand_address, pj_sockaddr_print(&rtp->ice->comp[component - 1].valid_check->rcand->addr, address, sizeof(address), 0), 0);
|
|
ast_sockaddr_set_port(cand_address, pj_sockaddr_get_port(&rtp->ice->comp[component - 1].valid_check->rcand->addr));
|
|
}
|
|
|
|
/*! \brief Destructor for locally created ICE candidates */
|
|
static void ast_rtp_ice_candidate_destroy(void *obj)
|
|
{
|
|
struct ast_rtp_engine_ice_candidate *candidate = obj;
|
|
|
|
if (candidate->foundation) {
|
|
ast_free(candidate->foundation);
|
|
}
|
|
|
|
if (candidate->transport) {
|
|
ast_free(candidate->transport);
|
|
}
|
|
}
|
|
|
|
static void ast_rtp_ice_set_authentication(struct ast_rtp_instance *instance, const char *ufrag, const char *password)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (!ast_strlen_zero(ufrag)) {
|
|
ast_copy_string(rtp->remote_ufrag, ufrag, sizeof(rtp->remote_ufrag));
|
|
}
|
|
|
|
if (!ast_strlen_zero(password)) {
|
|
ast_copy_string(rtp->remote_passwd, password, sizeof(rtp->remote_passwd));
|
|
}
|
|
}
|
|
|
|
static int ice_candidate_cmp(void *obj, void *arg, int flags)
|
|
{
|
|
struct ast_rtp_engine_ice_candidate *candidate1 = obj, *candidate2 = arg;
|
|
|
|
if (strcmp(candidate1->foundation, candidate2->foundation) ||
|
|
candidate1->id != candidate2->id ||
|
|
ast_sockaddr_cmp(&candidate1->address, &candidate2->address) ||
|
|
candidate1->type != candidate1->type) {
|
|
return 0;
|
|
}
|
|
|
|
return CMP_MATCH | CMP_STOP;
|
|
}
|
|
|
|
static void ast_rtp_ice_add_remote_candidate(struct ast_rtp_instance *instance, const struct ast_rtp_engine_ice_candidate *candidate)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_rtp_engine_ice_candidate *remote_candidate;
|
|
|
|
/* ICE sessions only support UDP candidates */
|
|
if (strcasecmp(candidate->transport, "udp")) {
|
|
return;
|
|
}
|
|
|
|
if (!rtp->ice_proposed_remote_candidates &&
|
|
!(rtp->ice_proposed_remote_candidates = ao2_container_alloc(1, NULL, ice_candidate_cmp))) {
|
|
return;
|
|
}
|
|
|
|
/* If this is going to exceed the maximum number of ICE candidates don't even add it */
|
|
if (ao2_container_count(rtp->ice_proposed_remote_candidates) == PJ_ICE_MAX_CAND) {
|
|
return;
|
|
}
|
|
|
|
if (!(remote_candidate = ao2_alloc(sizeof(*remote_candidate), ast_rtp_ice_candidate_destroy))) {
|
|
return;
|
|
}
|
|
|
|
remote_candidate->foundation = ast_strdup(candidate->foundation);
|
|
remote_candidate->id = candidate->id;
|
|
remote_candidate->transport = ast_strdup(candidate->transport);
|
|
remote_candidate->priority = candidate->priority;
|
|
ast_sockaddr_copy(&remote_candidate->address, &candidate->address);
|
|
ast_sockaddr_copy(&remote_candidate->relay_address, &candidate->relay_address);
|
|
remote_candidate->type = candidate->type;
|
|
|
|
ao2_link(rtp->ice_proposed_remote_candidates, remote_candidate);
|
|
ao2_ref(remote_candidate, -1);
|
|
}
|
|
|
|
AST_THREADSTORAGE(pj_thread_storage);
|
|
|
|
/*! \brief Function used to check if the calling thread is registered with pjlib. If it is not it will be registered. */
|
|
static void pj_thread_register_check(void)
|
|
{
|
|
pj_thread_desc *desc;
|
|
pj_thread_t *thread;
|
|
|
|
if (pj_thread_is_registered() == PJ_TRUE) {
|
|
return;
|
|
}
|
|
|
|
desc = ast_threadstorage_get(&pj_thread_storage, sizeof(pj_thread_desc));
|
|
if (!desc) {
|
|
ast_log(LOG_ERROR, "Could not get thread desc from thread-local storage. Expect awful things to occur\n");
|
|
return;
|
|
}
|
|
pj_bzero(*desc, sizeof(*desc));
|
|
|
|
if (pj_thread_register("Asterisk Thread", *desc, &thread) != PJ_SUCCESS) {
|
|
ast_log(LOG_ERROR, "Coudln't register thread with PJLIB.\n");
|
|
}
|
|
return;
|
|
}
|
|
|
|
static int ice_create(struct ast_rtp_instance *instance, struct ast_sockaddr *addr,
|
|
int port, int replace);
|
|
|
|
static void ast_rtp_ice_stop(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (!rtp->ice) {
|
|
return;
|
|
}
|
|
|
|
pj_thread_register_check();
|
|
|
|
pj_ice_sess_destroy(rtp->ice);
|
|
rtp->ice = NULL;
|
|
}
|
|
|
|
static int ice_reset_session(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
pj_ice_sess_role role = rtp->ice->role;
|
|
int res;
|
|
|
|
if (!rtp->ice->is_nominating && !rtp->ice->is_complete) {
|
|
return 0;
|
|
}
|
|
|
|
ast_rtp_ice_stop(instance);
|
|
|
|
res = ice_create(instance, &rtp->ice_original_rtp_addr, rtp->ice_port, 1);
|
|
if (!res) {
|
|
/* Preserve the role that the old ICE session used */
|
|
pj_ice_sess_change_role(rtp->ice, role);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static int ice_candidates_compare(struct ao2_container *left, struct ao2_container *right)
|
|
{
|
|
struct ao2_iterator i;
|
|
struct ast_rtp_engine_ice_candidate *right_candidate;
|
|
|
|
if (ao2_container_count(left) != ao2_container_count(right)) {
|
|
return -1;
|
|
}
|
|
|
|
i = ao2_iterator_init(right, 0);
|
|
while ((right_candidate = ao2_iterator_next(&i))) {
|
|
struct ast_rtp_engine_ice_candidate *left_candidate = ao2_find(left, right_candidate, OBJ_POINTER);
|
|
|
|
if (!left_candidate) {
|
|
ao2_ref(right_candidate, -1);
|
|
ao2_iterator_destroy(&i);
|
|
return -1;
|
|
}
|
|
|
|
ao2_ref(left_candidate, -1);
|
|
ao2_ref(right_candidate, -1);
|
|
}
|
|
ao2_iterator_destroy(&i);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void ast_rtp_ice_start(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
pj_str_t ufrag = pj_str(rtp->remote_ufrag), passwd = pj_str(rtp->remote_passwd);
|
|
pj_ice_sess_cand candidates[PJ_ICE_MAX_CAND];
|
|
struct ao2_iterator i;
|
|
struct ast_rtp_engine_ice_candidate *candidate;
|
|
int cand_cnt = 0, has_rtp = 0, has_rtcp = 0;
|
|
|
|
if (!rtp->ice || !rtp->ice_proposed_remote_candidates) {
|
|
return;
|
|
}
|
|
|
|
/* Check for equivalence in the lists */
|
|
if (rtp->ice_active_remote_candidates &&
|
|
!ice_candidates_compare(rtp->ice_proposed_remote_candidates, rtp->ice_active_remote_candidates)) {
|
|
ao2_cleanup(rtp->ice_proposed_remote_candidates);
|
|
rtp->ice_proposed_remote_candidates = NULL;
|
|
return;
|
|
}
|
|
|
|
/* Out with the old, in with the new */
|
|
ao2_cleanup(rtp->ice_active_remote_candidates);
|
|
rtp->ice_active_remote_candidates = rtp->ice_proposed_remote_candidates;
|
|
rtp->ice_proposed_remote_candidates = NULL;
|
|
|
|
/* Reset the ICE session. Is this going to work? */
|
|
if (ice_reset_session(instance)) {
|
|
ast_log(LOG_NOTICE, "Failed to create replacement ICE session\n");
|
|
return;
|
|
}
|
|
|
|
pj_thread_register_check();
|
|
|
|
i = ao2_iterator_init(rtp->ice_active_remote_candidates, 0);
|
|
|
|
while ((candidate = ao2_iterator_next(&i)) && (cand_cnt < PJ_ICE_MAX_CAND)) {
|
|
pj_str_t address;
|
|
|
|
/* there needs to be at least one rtp and rtcp candidate in the list */
|
|
has_rtp |= candidate->id == AST_RTP_ICE_COMPONENT_RTP;
|
|
has_rtcp |= candidate->id == AST_RTP_ICE_COMPONENT_RTCP;
|
|
|
|
pj_strdup2(rtp->ice->pool, &candidates[cand_cnt].foundation, candidate->foundation);
|
|
candidates[cand_cnt].comp_id = candidate->id;
|
|
candidates[cand_cnt].prio = candidate->priority;
|
|
|
|
pj_sockaddr_parse(pj_AF_UNSPEC(), 0, pj_cstr(&address, ast_sockaddr_stringify(&candidate->address)), &candidates[cand_cnt].addr);
|
|
|
|
if (!ast_sockaddr_isnull(&candidate->relay_address)) {
|
|
pj_sockaddr_parse(pj_AF_UNSPEC(), 0, pj_cstr(&address, ast_sockaddr_stringify(&candidate->relay_address)), &candidates[cand_cnt].rel_addr);
|
|
}
|
|
|
|
if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_HOST) {
|
|
candidates[cand_cnt].type = PJ_ICE_CAND_TYPE_HOST;
|
|
} else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX) {
|
|
candidates[cand_cnt].type = PJ_ICE_CAND_TYPE_SRFLX;
|
|
} else if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_RELAYED) {
|
|
candidates[cand_cnt].type = PJ_ICE_CAND_TYPE_RELAYED;
|
|
}
|
|
|
|
if (candidate->id == AST_RTP_ICE_COMPONENT_RTP && rtp->turn_rtp) {
|
|
pj_turn_sock_set_perm(rtp->turn_rtp, 1, &candidates[cand_cnt].addr, 1);
|
|
} else if (candidate->id == AST_RTP_ICE_COMPONENT_RTCP && rtp->turn_rtcp) {
|
|
pj_turn_sock_set_perm(rtp->turn_rtcp, 1, &candidates[cand_cnt].addr, 1);
|
|
}
|
|
|
|
cand_cnt++;
|
|
ao2_ref(candidate, -1);
|
|
}
|
|
|
|
ao2_iterator_destroy(&i);
|
|
|
|
if (has_rtp && has_rtcp &&
|
|
pj_ice_sess_create_check_list(rtp->ice, &ufrag, &passwd, ao2_container_count(
|
|
rtp->ice_active_remote_candidates), &candidates[0]) == PJ_SUCCESS) {
|
|
ast_test_suite_event_notify("ICECHECKLISTCREATE", "Result: SUCCESS");
|
|
pj_ice_sess_start_check(rtp->ice);
|
|
pj_timer_heap_poll(timer_heap, NULL);
|
|
rtp->strict_rtp_state = STRICT_RTP_OPEN;
|
|
return;
|
|
}
|
|
|
|
ast_test_suite_event_notify("ICECHECKLISTCREATE", "Result: FAILURE");
|
|
|
|
/* even though create check list failed don't stop ice as
|
|
it might still work */
|
|
ast_debug(1, "Failed to create ICE session check list\n");
|
|
/* however we do need to reset remote candidates since
|
|
this function may be re-entered */
|
|
ao2_ref(rtp->ice_active_remote_candidates, -1);
|
|
rtp->ice_active_remote_candidates = NULL;
|
|
rtp->ice->rcand_cnt = rtp->ice->clist.count = 0;
|
|
}
|
|
|
|
static const char *ast_rtp_ice_get_ufrag(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
return rtp->local_ufrag;
|
|
}
|
|
|
|
static const char *ast_rtp_ice_get_password(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
return rtp->local_passwd;
|
|
}
|
|
|
|
static struct ao2_container *ast_rtp_ice_get_local_candidates(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (rtp->ice_local_candidates) {
|
|
ao2_ref(rtp->ice_local_candidates, +1);
|
|
}
|
|
|
|
return rtp->ice_local_candidates;
|
|
}
|
|
|
|
static void ast_rtp_ice_lite(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (!rtp->ice) {
|
|
return;
|
|
}
|
|
|
|
pj_thread_register_check();
|
|
|
|
pj_ice_sess_change_role(rtp->ice, PJ_ICE_SESS_ROLE_CONTROLLING);
|
|
}
|
|
|
|
static void ast_rtp_ice_set_role(struct ast_rtp_instance *instance, enum ast_rtp_ice_role role)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (!rtp->ice) {
|
|
return;
|
|
}
|
|
|
|
pj_thread_register_check();
|
|
|
|
pj_ice_sess_change_role(rtp->ice, role == AST_RTP_ICE_ROLE_CONTROLLED ?
|
|
PJ_ICE_SESS_ROLE_CONTROLLED : PJ_ICE_SESS_ROLE_CONTROLLING);
|
|
}
|
|
|
|
static void ast_rtp_ice_add_cand(struct ast_rtp *rtp, unsigned comp_id, unsigned transport_id, pj_ice_cand_type type, pj_uint16_t local_pref,
|
|
const pj_sockaddr_t *addr, const pj_sockaddr_t *base_addr, const pj_sockaddr_t *rel_addr, int addr_len)
|
|
{
|
|
pj_str_t foundation;
|
|
struct ast_rtp_engine_ice_candidate *candidate, *existing;
|
|
char address[PJ_INET6_ADDRSTRLEN];
|
|
|
|
pj_thread_register_check();
|
|
|
|
pj_ice_calc_foundation(rtp->ice->pool, &foundation, type, addr);
|
|
|
|
if (!rtp->ice_local_candidates && !(rtp->ice_local_candidates = ao2_container_alloc(1, NULL, ice_candidate_cmp))) {
|
|
return;
|
|
}
|
|
|
|
if (!(candidate = ao2_alloc(sizeof(*candidate), ast_rtp_ice_candidate_destroy))) {
|
|
return;
|
|
}
|
|
|
|
candidate->foundation = ast_strndup(pj_strbuf(&foundation), pj_strlen(&foundation));
|
|
candidate->id = comp_id;
|
|
candidate->transport = ast_strdup("UDP");
|
|
|
|
ast_sockaddr_parse(&candidate->address, pj_sockaddr_print(addr, address, sizeof(address), 0), 0);
|
|
ast_sockaddr_set_port(&candidate->address, pj_sockaddr_get_port(addr));
|
|
|
|
if (rel_addr) {
|
|
ast_sockaddr_parse(&candidate->relay_address, pj_sockaddr_print(rel_addr, address, sizeof(address), 0), 0);
|
|
ast_sockaddr_set_port(&candidate->relay_address, pj_sockaddr_get_port(rel_addr));
|
|
}
|
|
|
|
if (type == PJ_ICE_CAND_TYPE_HOST) {
|
|
candidate->type = AST_RTP_ICE_CANDIDATE_TYPE_HOST;
|
|
} else if (type == PJ_ICE_CAND_TYPE_SRFLX) {
|
|
candidate->type = AST_RTP_ICE_CANDIDATE_TYPE_SRFLX;
|
|
} else if (type == PJ_ICE_CAND_TYPE_RELAYED) {
|
|
candidate->type = AST_RTP_ICE_CANDIDATE_TYPE_RELAYED;
|
|
}
|
|
|
|
if ((existing = ao2_find(rtp->ice_local_candidates, candidate, OBJ_POINTER))) {
|
|
ao2_ref(existing, -1);
|
|
ao2_ref(candidate, -1);
|
|
return;
|
|
}
|
|
|
|
if (pj_ice_sess_add_cand(rtp->ice, comp_id, transport_id, type, local_pref, &foundation, addr, base_addr, rel_addr, addr_len, NULL) != PJ_SUCCESS) {
|
|
ao2_ref(candidate, -1);
|
|
return;
|
|
}
|
|
|
|
/* By placing the candidate into the ICE session it will have produced the priority, so update the local candidate with it */
|
|
candidate->priority = rtp->ice->lcand[rtp->ice->lcand_cnt - 1].prio;
|
|
|
|
ao2_link(rtp->ice_local_candidates, candidate);
|
|
ao2_ref(candidate, -1);
|
|
}
|
|
|
|
static void ast_rtp_on_turn_rx_rtp_data(pj_turn_sock *turn_sock, void *pkt, unsigned pkt_len, const pj_sockaddr_t *peer_addr, unsigned addr_len)
|
|
{
|
|
struct ast_rtp_instance *instance = pj_turn_sock_get_user_data(turn_sock);
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
pj_status_t status;
|
|
|
|
status = pj_ice_sess_on_rx_pkt(rtp->ice, AST_RTP_ICE_COMPONENT_RTP, TRANSPORT_TURN_RTP, pkt, pkt_len, peer_addr,
|
|
addr_len);
|
|
if (status != PJ_SUCCESS) {
|
|
char buf[100];
|
|
|
|
pj_strerror(status, buf, sizeof(buf));
|
|
ast_log(LOG_WARNING, "PJ ICE Rx error status code: %d '%s'.\n",
|
|
(int)status, buf);
|
|
return;
|
|
}
|
|
if (!rtp->rtp_passthrough) {
|
|
return;
|
|
}
|
|
rtp->rtp_passthrough = 0;
|
|
|
|
ast_sendto(rtp->s, pkt, pkt_len, 0, &rtp->rtp_loop);
|
|
}
|
|
|
|
static void ast_rtp_on_turn_rtp_state(pj_turn_sock *turn_sock, pj_turn_state_t old_state, pj_turn_state_t new_state)
|
|
{
|
|
struct ast_rtp_instance *instance = pj_turn_sock_get_user_data(turn_sock);
|
|
struct ast_rtp *rtp;
|
|
|
|
/* If this is a leftover from an already notified RTP instance just ignore the state change */
|
|
if (!instance) {
|
|
return;
|
|
}
|
|
|
|
rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
/* We store the new state so the other thread can actually handle it */
|
|
ast_mutex_lock(&rtp->lock);
|
|
rtp->turn_state = new_state;
|
|
ast_cond_signal(&rtp->cond);
|
|
|
|
if (new_state == PJ_TURN_STATE_DESTROYING) {
|
|
pj_turn_sock_set_user_data(rtp->turn_rtp, NULL);
|
|
rtp->turn_rtp = NULL;
|
|
}
|
|
|
|
ast_mutex_unlock(&rtp->lock);
|
|
}
|
|
|
|
/* RTP TURN Socket interface declaration */
|
|
static pj_turn_sock_cb ast_rtp_turn_rtp_sock_cb = {
|
|
.on_rx_data = ast_rtp_on_turn_rx_rtp_data,
|
|
.on_state = ast_rtp_on_turn_rtp_state,
|
|
};
|
|
|
|
static void ast_rtp_on_turn_rx_rtcp_data(pj_turn_sock *turn_sock, void *pkt, unsigned pkt_len, const pj_sockaddr_t *peer_addr, unsigned addr_len)
|
|
{
|
|
struct ast_rtp_instance *instance = pj_turn_sock_get_user_data(turn_sock);
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
pj_status_t status;
|
|
|
|
status = pj_ice_sess_on_rx_pkt(rtp->ice, AST_RTP_ICE_COMPONENT_RTCP, TRANSPORT_TURN_RTCP, pkt, pkt_len, peer_addr,
|
|
addr_len);
|
|
if (status != PJ_SUCCESS) {
|
|
char buf[100];
|
|
|
|
pj_strerror(status, buf, sizeof(buf));
|
|
ast_log(LOG_WARNING, "PJ ICE Rx error status code: %d '%s'.\n",
|
|
(int)status, buf);
|
|
return;
|
|
}
|
|
if (!rtp->rtcp_passthrough) {
|
|
return;
|
|
}
|
|
rtp->rtcp_passthrough = 0;
|
|
|
|
ast_sendto(rtp->rtcp->s, pkt, pkt_len, 0, &rtp->rtcp_loop);
|
|
}
|
|
|
|
static void ast_rtp_on_turn_rtcp_state(pj_turn_sock *turn_sock, pj_turn_state_t old_state, pj_turn_state_t new_state)
|
|
{
|
|
struct ast_rtp_instance *instance = pj_turn_sock_get_user_data(turn_sock);
|
|
struct ast_rtp *rtp = NULL;
|
|
|
|
/* If this is a leftover from an already destroyed RTP instance just ignore the state change */
|
|
if (!instance) {
|
|
return;
|
|
}
|
|
|
|
rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
/* We store the new state so the other thread can actually handle it */
|
|
ast_mutex_lock(&rtp->lock);
|
|
rtp->turn_state = new_state;
|
|
ast_cond_signal(&rtp->cond);
|
|
|
|
if (new_state == PJ_TURN_STATE_DESTROYING) {
|
|
pj_turn_sock_set_user_data(rtp->turn_rtcp, NULL);
|
|
rtp->turn_rtcp = NULL;
|
|
}
|
|
|
|
ast_mutex_unlock(&rtp->lock);
|
|
}
|
|
|
|
/* RTCP TURN Socket interface declaration */
|
|
static pj_turn_sock_cb ast_rtp_turn_rtcp_sock_cb = {
|
|
.on_rx_data = ast_rtp_on_turn_rx_rtcp_data,
|
|
.on_state = ast_rtp_on_turn_rtcp_state,
|
|
};
|
|
|
|
/*! \brief Worker thread for ioqueue and timerheap */
|
|
static int ioqueue_worker_thread(void *data)
|
|
{
|
|
struct ast_rtp_ioqueue_thread *ioqueue = data;
|
|
|
|
while (!ioqueue->terminate) {
|
|
const pj_time_val delay = {0, 10};
|
|
|
|
pj_ioqueue_poll(ioqueue->ioqueue, &delay);
|
|
|
|
pj_timer_heap_poll(ioqueue->timerheap, NULL);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*! \brief Destroyer for ioqueue thread */
|
|
static void rtp_ioqueue_thread_destroy(struct ast_rtp_ioqueue_thread *ioqueue)
|
|
{
|
|
if (ioqueue->thread) {
|
|
ioqueue->terminate = 1;
|
|
pj_thread_join(ioqueue->thread);
|
|
pj_thread_destroy(ioqueue->thread);
|
|
}
|
|
|
|
pj_pool_release(ioqueue->pool);
|
|
ast_free(ioqueue);
|
|
}
|
|
|
|
/*! \brief Removal function for ioqueue thread, determines if it should be terminated and destroyed */
|
|
static void rtp_ioqueue_thread_remove(struct ast_rtp_ioqueue_thread *ioqueue)
|
|
{
|
|
int destroy = 0;
|
|
|
|
/* If nothing is using this ioqueue thread destroy it */
|
|
AST_LIST_LOCK(&ioqueues);
|
|
if ((ioqueue->count - 2) == 0) {
|
|
destroy = 1;
|
|
AST_LIST_REMOVE(&ioqueues, ioqueue, next);
|
|
}
|
|
AST_LIST_UNLOCK(&ioqueues);
|
|
|
|
if (!destroy) {
|
|
return;
|
|
}
|
|
|
|
rtp_ioqueue_thread_destroy(ioqueue);
|
|
}
|
|
|
|
/*! \brief Finder and allocator for an ioqueue thread */
|
|
static struct ast_rtp_ioqueue_thread *rtp_ioqueue_thread_get_or_create(void)
|
|
{
|
|
struct ast_rtp_ioqueue_thread *ioqueue;
|
|
pj_lock_t *lock;
|
|
|
|
AST_LIST_LOCK(&ioqueues);
|
|
|
|
/* See if an ioqueue thread exists that can handle more */
|
|
AST_LIST_TRAVERSE(&ioqueues, ioqueue, next) {
|
|
if ((ioqueue->count + 2) < PJ_IOQUEUE_MAX_HANDLES) {
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* If we found one bump it up and return it */
|
|
if (ioqueue) {
|
|
ioqueue->count += 2;
|
|
goto end;
|
|
}
|
|
|
|
ioqueue = ast_calloc(1, sizeof(*ioqueue));
|
|
if (!ioqueue) {
|
|
goto end;
|
|
}
|
|
|
|
ioqueue->pool = pj_pool_create(&cachingpool.factory, "rtp", 512, 512, NULL);
|
|
|
|
/* We use a timer on the ioqueue thread for TURN so that two threads aren't operating
|
|
* on a session at the same time
|
|
*/
|
|
if (pj_timer_heap_create(ioqueue->pool, 4, &ioqueue->timerheap) != PJ_SUCCESS) {
|
|
goto fatal;
|
|
}
|
|
|
|
if (pj_lock_create_recursive_mutex(ioqueue->pool, "rtp%p", &lock) != PJ_SUCCESS) {
|
|
goto fatal;
|
|
}
|
|
|
|
pj_timer_heap_set_lock(ioqueue->timerheap, lock, PJ_TRUE);
|
|
|
|
if (pj_ioqueue_create(ioqueue->pool, PJ_IOQUEUE_MAX_HANDLES, &ioqueue->ioqueue) != PJ_SUCCESS) {
|
|
goto fatal;
|
|
}
|
|
|
|
if (pj_thread_create(ioqueue->pool, "ice", &ioqueue_worker_thread, ioqueue, 0, 0, &ioqueue->thread) != PJ_SUCCESS) {
|
|
goto fatal;
|
|
}
|
|
|
|
AST_LIST_INSERT_HEAD(&ioqueues, ioqueue, next);
|
|
|
|
/* Since this is being returned to an active session the count always starts at 2 */
|
|
ioqueue->count = 2;
|
|
|
|
goto end;
|
|
|
|
fatal:
|
|
rtp_ioqueue_thread_destroy(ioqueue);
|
|
ioqueue = NULL;
|
|
|
|
end:
|
|
AST_LIST_UNLOCK(&ioqueues);
|
|
return ioqueue;
|
|
}
|
|
|
|
static void ast_rtp_ice_turn_request(struct ast_rtp_instance *instance, enum ast_rtp_ice_component_type component,
|
|
enum ast_transport transport, const char *server, unsigned int port, const char *username, const char *password)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
pj_turn_sock **turn_sock;
|
|
const pj_turn_sock_cb *turn_cb;
|
|
pj_turn_tp_type conn_type;
|
|
int conn_transport;
|
|
pj_stun_auth_cred cred = { 0, };
|
|
pj_str_t turn_addr;
|
|
struct ast_sockaddr addr = { { 0, } };
|
|
pj_stun_config stun_config;
|
|
struct timeval wait = ast_tvadd(ast_tvnow(), ast_samp2tv(TURN_STATE_WAIT_TIME, 1000));
|
|
struct timespec ts = { .tv_sec = wait.tv_sec, .tv_nsec = wait.tv_usec * 1000, };
|
|
pj_turn_session_info info;
|
|
struct ast_sockaddr local, loop;
|
|
|
|
ast_rtp_instance_get_local_address(instance, &local);
|
|
if (ast_sockaddr_is_ipv4(&local)) {
|
|
ast_sockaddr_parse(&loop, "127.0.0.1", PARSE_PORT_FORBID);
|
|
} else {
|
|
ast_sockaddr_parse(&loop, "::1", PARSE_PORT_FORBID);
|
|
}
|
|
|
|
/* Determine what component we are requesting a TURN session for */
|
|
if (component == AST_RTP_ICE_COMPONENT_RTP) {
|
|
turn_sock = &rtp->turn_rtp;
|
|
turn_cb = &ast_rtp_turn_rtp_sock_cb;
|
|
conn_transport = TRANSPORT_TURN_RTP;
|
|
ast_sockaddr_set_port(&loop, ast_sockaddr_port(&local));
|
|
} else if (component == AST_RTP_ICE_COMPONENT_RTCP) {
|
|
turn_sock = &rtp->turn_rtcp;
|
|
turn_cb = &ast_rtp_turn_rtcp_sock_cb;
|
|
conn_transport = TRANSPORT_TURN_RTCP;
|
|
ast_sockaddr_set_port(&loop, ast_sockaddr_port(&rtp->rtcp->us));
|
|
} else {
|
|
return;
|
|
}
|
|
|
|
if (transport == AST_TRANSPORT_UDP) {
|
|
conn_type = PJ_TURN_TP_UDP;
|
|
} else if (transport == AST_TRANSPORT_TCP) {
|
|
conn_type = PJ_TURN_TP_TCP;
|
|
} else {
|
|
ast_assert(0);
|
|
return;
|
|
}
|
|
|
|
ast_sockaddr_parse(&addr, server, PARSE_PORT_FORBID);
|
|
|
|
ast_mutex_lock(&rtp->lock);
|
|
if (*turn_sock) {
|
|
pj_turn_sock_destroy(*turn_sock);
|
|
rtp->turn_state = PJ_TURN_STATE_NULL;
|
|
while (rtp->turn_state != PJ_TURN_STATE_DESTROYING) {
|
|
ast_cond_timedwait(&rtp->cond, &rtp->lock, &ts);
|
|
}
|
|
}
|
|
ast_mutex_unlock(&rtp->lock);
|
|
|
|
if (component == AST_RTP_ICE_COMPONENT_RTP && !rtp->ioqueue) {
|
|
rtp->ioqueue = rtp_ioqueue_thread_get_or_create();
|
|
if (!rtp->ioqueue) {
|
|
return;
|
|
}
|
|
}
|
|
|
|
pj_stun_config_init(&stun_config, &cachingpool.factory, 0, rtp->ioqueue->ioqueue, rtp->ioqueue->timerheap);
|
|
|
|
if (pj_turn_sock_create(&stun_config, ast_sockaddr_is_ipv4(&addr) ? pj_AF_INET() : pj_AF_INET6(), conn_type,
|
|
turn_cb, NULL, instance, turn_sock) != PJ_SUCCESS) {
|
|
ast_log(LOG_WARNING, "Could not create a TURN client socket\n");
|
|
return;
|
|
}
|
|
|
|
cred.type = PJ_STUN_AUTH_CRED_STATIC;
|
|
pj_strset2(&cred.data.static_cred.username, (char*)username);
|
|
cred.data.static_cred.data_type = PJ_STUN_PASSWD_PLAIN;
|
|
pj_strset2(&cred.data.static_cred.data, (char*)password);
|
|
|
|
/* Because the TURN socket is asynchronous but we are synchronous we need to wait until it is done */
|
|
ast_mutex_lock(&rtp->lock);
|
|
pj_turn_sock_alloc(*turn_sock, pj_cstr(&turn_addr, server), port, NULL, &cred, NULL);
|
|
while (rtp->turn_state < PJ_TURN_STATE_READY) {
|
|
ast_cond_timedwait(&rtp->cond, &rtp->lock, &ts);
|
|
}
|
|
ast_mutex_unlock(&rtp->lock);
|
|
|
|
/* If a TURN session was allocated add it as a candidate */
|
|
if (rtp->turn_state != PJ_TURN_STATE_READY) {
|
|
return;
|
|
}
|
|
|
|
pj_turn_sock_get_info(*turn_sock, &info);
|
|
|
|
ast_rtp_ice_add_cand(rtp, component, conn_transport, PJ_ICE_CAND_TYPE_RELAYED, 65535, &info.relay_addr,
|
|
&info.relay_addr, &info.mapped_addr, pj_sockaddr_get_len(&info.relay_addr));
|
|
|
|
if (component == AST_RTP_ICE_COMPONENT_RTP) {
|
|
ast_sockaddr_copy(&rtp->rtp_loop, &loop);
|
|
} else if (component == AST_RTP_ICE_COMPONENT_RTCP) {
|
|
ast_sockaddr_copy(&rtp->rtcp_loop, &loop);
|
|
}
|
|
}
|
|
|
|
static char *generate_random_string(char *buf, size_t size)
|
|
{
|
|
long val[4];
|
|
int x;
|
|
|
|
for (x=0; x<4; x++) {
|
|
val[x] = ast_random();
|
|
}
|
|
snprintf(buf, size, "%08lx%08lx%08lx%08lx", (long unsigned)val[0], (long unsigned)val[1], (long unsigned)val[2], (long unsigned)val[3]);
|
|
|
|
return buf;
|
|
}
|
|
|
|
/* ICE RTP Engine interface declaration */
|
|
static struct ast_rtp_engine_ice ast_rtp_ice = {
|
|
.set_authentication = ast_rtp_ice_set_authentication,
|
|
.add_remote_candidate = ast_rtp_ice_add_remote_candidate,
|
|
.start = ast_rtp_ice_start,
|
|
.stop = ast_rtp_ice_stop,
|
|
.get_ufrag = ast_rtp_ice_get_ufrag,
|
|
.get_password = ast_rtp_ice_get_password,
|
|
.get_local_candidates = ast_rtp_ice_get_local_candidates,
|
|
.ice_lite = ast_rtp_ice_lite,
|
|
.set_role = ast_rtp_ice_set_role,
|
|
.turn_request = ast_rtp_ice_turn_request,
|
|
};
|
|
#endif
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
static int dtls_verify_callback(int preverify_ok, X509_STORE_CTX *ctx)
|
|
{
|
|
/* We don't want to actually verify the certificate so just accept what they have provided */
|
|
return 1;
|
|
}
|
|
|
|
static int dtls_details_initialize(struct dtls_details *dtls, SSL_CTX *ssl_ctx,
|
|
enum ast_rtp_dtls_setup setup)
|
|
{
|
|
dtls->dtls_setup = setup;
|
|
|
|
if (!(dtls->ssl = SSL_new(ssl_ctx))) {
|
|
ast_log(LOG_ERROR, "Failed to allocate memory for SSL\n");
|
|
goto error;
|
|
}
|
|
|
|
if (!(dtls->read_bio = BIO_new(BIO_s_mem()))) {
|
|
ast_log(LOG_ERROR, "Failed to allocate memory for inbound SSL traffic\n");
|
|
goto error;
|
|
}
|
|
BIO_set_mem_eof_return(dtls->read_bio, -1);
|
|
|
|
if (!(dtls->write_bio = BIO_new(BIO_s_mem()))) {
|
|
ast_log(LOG_ERROR, "Failed to allocate memory for outbound SSL traffic\n");
|
|
goto error;
|
|
}
|
|
BIO_set_mem_eof_return(dtls->write_bio, -1);
|
|
|
|
SSL_set_bio(dtls->ssl, dtls->read_bio, dtls->write_bio);
|
|
|
|
if (dtls->dtls_setup == AST_RTP_DTLS_SETUP_PASSIVE) {
|
|
SSL_set_accept_state(dtls->ssl);
|
|
} else {
|
|
SSL_set_connect_state(dtls->ssl);
|
|
}
|
|
dtls->connection = AST_RTP_DTLS_CONNECTION_NEW;
|
|
|
|
ast_mutex_init(&dtls->lock);
|
|
|
|
return 0;
|
|
|
|
error:
|
|
if (dtls->read_bio) {
|
|
BIO_free(dtls->read_bio);
|
|
dtls->read_bio = NULL;
|
|
}
|
|
|
|
if (dtls->write_bio) {
|
|
BIO_free(dtls->write_bio);
|
|
dtls->write_bio = NULL;
|
|
}
|
|
|
|
if (dtls->ssl) {
|
|
SSL_free(dtls->ssl);
|
|
dtls->ssl = NULL;
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
static int dtls_setup_rtcp(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (!rtp->ssl_ctx || !rtp->rtcp) {
|
|
return 0;
|
|
}
|
|
|
|
return dtls_details_initialize(&rtp->rtcp->dtls, rtp->ssl_ctx, rtp->dtls.dtls_setup);
|
|
}
|
|
|
|
static int ast_rtp_dtls_set_configuration(struct ast_rtp_instance *instance, const struct ast_rtp_dtls_cfg *dtls_cfg)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
int res;
|
|
|
|
if (!dtls_cfg->enabled) {
|
|
return 0;
|
|
}
|
|
|
|
if (!ast_rtp_engine_srtp_is_registered()) {
|
|
ast_log(LOG_ERROR, "SRTP support module is not loaded or available. Try loading res_srtp.so.\n");
|
|
return -1;
|
|
}
|
|
|
|
if (rtp->ssl_ctx) {
|
|
return 0;
|
|
}
|
|
|
|
if (!(rtp->ssl_ctx = SSL_CTX_new(DTLSv1_method()))) {
|
|
return -1;
|
|
}
|
|
|
|
SSL_CTX_set_read_ahead(rtp->ssl_ctx, 1);
|
|
|
|
rtp->dtls_verify = dtls_cfg->verify;
|
|
|
|
SSL_CTX_set_verify(rtp->ssl_ctx, (rtp->dtls_verify & AST_RTP_DTLS_VERIFY_FINGERPRINT) || (rtp->dtls_verify & AST_RTP_DTLS_VERIFY_CERTIFICATE) ?
|
|
SSL_VERIFY_PEER | SSL_VERIFY_FAIL_IF_NO_PEER_CERT : SSL_VERIFY_NONE, !(rtp->dtls_verify & AST_RTP_DTLS_VERIFY_CERTIFICATE) ?
|
|
dtls_verify_callback : NULL);
|
|
|
|
if (dtls_cfg->suite == AST_AES_CM_128_HMAC_SHA1_80) {
|
|
SSL_CTX_set_tlsext_use_srtp(rtp->ssl_ctx, "SRTP_AES128_CM_SHA1_80");
|
|
} else if (dtls_cfg->suite == AST_AES_CM_128_HMAC_SHA1_32) {
|
|
SSL_CTX_set_tlsext_use_srtp(rtp->ssl_ctx, "SRTP_AES128_CM_SHA1_32");
|
|
} else {
|
|
ast_log(LOG_ERROR, "Unsupported suite specified for DTLS-SRTP on RTP instance '%p'\n", instance);
|
|
return -1;
|
|
}
|
|
|
|
rtp->local_hash = dtls_cfg->hash;
|
|
|
|
if (!ast_strlen_zero(dtls_cfg->certfile)) {
|
|
char *private = ast_strlen_zero(dtls_cfg->pvtfile) ? dtls_cfg->certfile : dtls_cfg->pvtfile;
|
|
BIO *certbio;
|
|
X509 *cert;
|
|
const EVP_MD *type;
|
|
unsigned int size, i;
|
|
unsigned char fingerprint[EVP_MAX_MD_SIZE];
|
|
char *local_fingerprint = rtp->local_fingerprint;
|
|
|
|
if (!SSL_CTX_use_certificate_file(rtp->ssl_ctx, dtls_cfg->certfile, SSL_FILETYPE_PEM)) {
|
|
ast_log(LOG_ERROR, "Specified certificate file '%s' for RTP instance '%p' could not be used\n",
|
|
dtls_cfg->certfile, instance);
|
|
return -1;
|
|
}
|
|
|
|
if (!SSL_CTX_use_PrivateKey_file(rtp->ssl_ctx, private, SSL_FILETYPE_PEM) ||
|
|
!SSL_CTX_check_private_key(rtp->ssl_ctx)) {
|
|
ast_log(LOG_ERROR, "Specified private key file '%s' for RTP instance '%p' could not be used\n",
|
|
private, instance);
|
|
return -1;
|
|
}
|
|
|
|
if (!(certbio = BIO_new(BIO_s_file()))) {
|
|
ast_log(LOG_ERROR, "Failed to allocate memory for certificate fingerprinting on RTP instance '%p'\n",
|
|
instance);
|
|
return -1;
|
|
}
|
|
|
|
if (rtp->local_hash == AST_RTP_DTLS_HASH_SHA1) {
|
|
type = EVP_sha1();
|
|
} else if (rtp->local_hash == AST_RTP_DTLS_HASH_SHA256) {
|
|
type = EVP_sha256();
|
|
} else {
|
|
ast_log(LOG_ERROR, "Unsupported fingerprint hash type on RTP instance '%p'\n",
|
|
instance);
|
|
return -1;
|
|
}
|
|
|
|
if (!BIO_read_filename(certbio, dtls_cfg->certfile) ||
|
|
!(cert = PEM_read_bio_X509(certbio, NULL, 0, NULL)) ||
|
|
!X509_digest(cert, type, fingerprint, &size) ||
|
|
!size) {
|
|
ast_log(LOG_ERROR, "Could not produce fingerprint from certificate '%s' for RTP instance '%p'\n",
|
|
dtls_cfg->certfile, instance);
|
|
BIO_free_all(certbio);
|
|
return -1;
|
|
}
|
|
|
|
for (i = 0; i < size; i++) {
|
|
sprintf(local_fingerprint, "%02hhX:", fingerprint[i]);
|
|
local_fingerprint += 3;
|
|
}
|
|
|
|
*(local_fingerprint-1) = 0;
|
|
|
|
BIO_free_all(certbio);
|
|
}
|
|
|
|
if (!ast_strlen_zero(dtls_cfg->cipher)) {
|
|
if (!SSL_CTX_set_cipher_list(rtp->ssl_ctx, dtls_cfg->cipher)) {
|
|
ast_log(LOG_ERROR, "Invalid cipher specified in cipher list '%s' for RTP instance '%p'\n",
|
|
dtls_cfg->cipher, instance);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
if (!ast_strlen_zero(dtls_cfg->cafile) || !ast_strlen_zero(dtls_cfg->capath)) {
|
|
if (!SSL_CTX_load_verify_locations(rtp->ssl_ctx, S_OR(dtls_cfg->cafile, NULL), S_OR(dtls_cfg->capath, NULL))) {
|
|
ast_log(LOG_ERROR, "Invalid certificate authority file '%s' or path '%s' specified for RTP instance '%p'\n",
|
|
S_OR(dtls_cfg->cafile, ""), S_OR(dtls_cfg->capath, ""), instance);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
rtp->rekey = dtls_cfg->rekey;
|
|
rtp->suite = dtls_cfg->suite;
|
|
|
|
res = dtls_details_initialize(&rtp->dtls, rtp->ssl_ctx, dtls_cfg->default_setup);
|
|
if (!res) {
|
|
dtls_setup_rtcp(instance);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static int ast_rtp_dtls_active(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
return !rtp->ssl_ctx ? 0 : 1;
|
|
}
|
|
|
|
static void ast_rtp_dtls_stop(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
dtls_srtp_stop_timeout_timer(instance, rtp, 0);
|
|
|
|
if (rtp->ssl_ctx) {
|
|
SSL_CTX_free(rtp->ssl_ctx);
|
|
rtp->ssl_ctx = NULL;
|
|
}
|
|
|
|
if (rtp->dtls.ssl) {
|
|
SSL_free(rtp->dtls.ssl);
|
|
rtp->dtls.ssl = NULL;
|
|
ast_mutex_destroy(&rtp->dtls.lock);
|
|
}
|
|
|
|
if (rtp->rtcp) {
|
|
dtls_srtp_stop_timeout_timer(instance, rtp, 1);
|
|
|
|
if (rtp->rtcp->dtls.ssl) {
|
|
SSL_free(rtp->rtcp->dtls.ssl);
|
|
rtp->rtcp->dtls.ssl = NULL;
|
|
ast_mutex_destroy(&rtp->rtcp->dtls.lock);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void ast_rtp_dtls_reset(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (SSL_is_init_finished(rtp->dtls.ssl)) {
|
|
SSL_shutdown(rtp->dtls.ssl);
|
|
rtp->dtls.connection = AST_RTP_DTLS_CONNECTION_NEW;
|
|
}
|
|
|
|
if (rtp->rtcp && SSL_is_init_finished(rtp->rtcp->dtls.ssl)) {
|
|
SSL_shutdown(rtp->rtcp->dtls.ssl);
|
|
rtp->rtcp->dtls.connection = AST_RTP_DTLS_CONNECTION_NEW;
|
|
}
|
|
}
|
|
|
|
static enum ast_rtp_dtls_connection ast_rtp_dtls_get_connection(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
return rtp->dtls.connection;
|
|
}
|
|
|
|
static enum ast_rtp_dtls_setup ast_rtp_dtls_get_setup(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
return rtp->dtls.dtls_setup;
|
|
}
|
|
|
|
static void dtls_set_setup(enum ast_rtp_dtls_setup *dtls_setup, enum ast_rtp_dtls_setup setup, SSL *ssl)
|
|
{
|
|
enum ast_rtp_dtls_setup old = *dtls_setup;
|
|
|
|
switch (setup) {
|
|
case AST_RTP_DTLS_SETUP_ACTIVE:
|
|
*dtls_setup = AST_RTP_DTLS_SETUP_PASSIVE;
|
|
break;
|
|
case AST_RTP_DTLS_SETUP_PASSIVE:
|
|
*dtls_setup = AST_RTP_DTLS_SETUP_ACTIVE;
|
|
break;
|
|
case AST_RTP_DTLS_SETUP_ACTPASS:
|
|
/* We can't respond to an actpass setup with actpass ourselves... so respond with active, as we can initiate connections */
|
|
if (*dtls_setup == AST_RTP_DTLS_SETUP_ACTPASS) {
|
|
*dtls_setup = AST_RTP_DTLS_SETUP_ACTIVE;
|
|
}
|
|
break;
|
|
case AST_RTP_DTLS_SETUP_HOLDCONN:
|
|
*dtls_setup = AST_RTP_DTLS_SETUP_HOLDCONN;
|
|
break;
|
|
default:
|
|
/* This should never occur... if it does exit early as we don't know what state things are in */
|
|
return;
|
|
}
|
|
|
|
/* If the setup state did not change we go on as if nothing happened */
|
|
if (old == *dtls_setup) {
|
|
return;
|
|
}
|
|
|
|
/* If they don't want us to establish a connection wait until later */
|
|
if (*dtls_setup == AST_RTP_DTLS_SETUP_HOLDCONN) {
|
|
return;
|
|
}
|
|
|
|
if (*dtls_setup == AST_RTP_DTLS_SETUP_ACTIVE) {
|
|
SSL_set_connect_state(ssl);
|
|
} else if (*dtls_setup == AST_RTP_DTLS_SETUP_PASSIVE) {
|
|
SSL_set_accept_state(ssl);
|
|
} else {
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void ast_rtp_dtls_set_setup(struct ast_rtp_instance *instance, enum ast_rtp_dtls_setup setup)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (rtp->dtls.ssl) {
|
|
dtls_set_setup(&rtp->dtls.dtls_setup, setup, rtp->dtls.ssl);
|
|
}
|
|
|
|
if (rtp->rtcp && rtp->rtcp->dtls.ssl) {
|
|
dtls_set_setup(&rtp->rtcp->dtls.dtls_setup, setup, rtp->rtcp->dtls.ssl);
|
|
}
|
|
}
|
|
|
|
static void ast_rtp_dtls_set_fingerprint(struct ast_rtp_instance *instance, enum ast_rtp_dtls_hash hash, const char *fingerprint)
|
|
{
|
|
char *tmp = ast_strdupa(fingerprint), *value;
|
|
int pos = 0;
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (hash != AST_RTP_DTLS_HASH_SHA1 && hash != AST_RTP_DTLS_HASH_SHA256) {
|
|
return;
|
|
}
|
|
|
|
rtp->remote_hash = hash;
|
|
|
|
while ((value = strsep(&tmp, ":")) && (pos != (EVP_MAX_MD_SIZE - 1))) {
|
|
sscanf(value, "%02hhx", &rtp->remote_fingerprint[pos++]);
|
|
}
|
|
}
|
|
|
|
static enum ast_rtp_dtls_hash ast_rtp_dtls_get_fingerprint_hash(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
return rtp->local_hash;
|
|
}
|
|
|
|
static const char *ast_rtp_dtls_get_fingerprint(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
return rtp->local_fingerprint;
|
|
}
|
|
|
|
/* DTLS RTP Engine interface declaration */
|
|
static struct ast_rtp_engine_dtls ast_rtp_dtls = {
|
|
.set_configuration = ast_rtp_dtls_set_configuration,
|
|
.active = ast_rtp_dtls_active,
|
|
.stop = ast_rtp_dtls_stop,
|
|
.reset = ast_rtp_dtls_reset,
|
|
.get_connection = ast_rtp_dtls_get_connection,
|
|
.get_setup = ast_rtp_dtls_get_setup,
|
|
.set_setup = ast_rtp_dtls_set_setup,
|
|
.set_fingerprint = ast_rtp_dtls_set_fingerprint,
|
|
.get_fingerprint_hash = ast_rtp_dtls_get_fingerprint_hash,
|
|
.get_fingerprint = ast_rtp_dtls_get_fingerprint,
|
|
};
|
|
|
|
#endif
|
|
|
|
/* RTP Engine Declaration */
|
|
static struct ast_rtp_engine asterisk_rtp_engine = {
|
|
.name = "asterisk",
|
|
.new = ast_rtp_new,
|
|
.destroy = ast_rtp_destroy,
|
|
.dtmf_begin = ast_rtp_dtmf_begin,
|
|
.dtmf_end = ast_rtp_dtmf_end,
|
|
.dtmf_end_with_duration = ast_rtp_dtmf_end_with_duration,
|
|
.dtmf_mode_set = ast_rtp_dtmf_mode_set,
|
|
.dtmf_mode_get = ast_rtp_dtmf_mode_get,
|
|
.update_source = ast_rtp_update_source,
|
|
.change_source = ast_rtp_change_source,
|
|
.write = ast_rtp_write,
|
|
.read = ast_rtp_read,
|
|
.prop_set = ast_rtp_prop_set,
|
|
.fd = ast_rtp_fd,
|
|
.remote_address_set = ast_rtp_remote_address_set,
|
|
.red_init = rtp_red_init,
|
|
.red_buffer = rtp_red_buffer,
|
|
.local_bridge = ast_rtp_local_bridge,
|
|
.get_stat = ast_rtp_get_stat,
|
|
.dtmf_compatible = ast_rtp_dtmf_compatible,
|
|
.stun_request = ast_rtp_stun_request,
|
|
.stop = ast_rtp_stop,
|
|
.qos = ast_rtp_qos_set,
|
|
.sendcng = ast_rtp_sendcng,
|
|
#ifdef HAVE_PJPROJECT
|
|
.ice = &ast_rtp_ice,
|
|
#endif
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
.dtls = &ast_rtp_dtls,
|
|
.activate = ast_rtp_activate,
|
|
#endif
|
|
};
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
static void dtls_perform_handshake(struct ast_rtp_instance *instance, struct dtls_details *dtls, int rtcp)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
/* If we are not acting as a client connecting to the remote side then
|
|
* don't start the handshake as it will accomplish nothing and would conflict
|
|
* with the handshake we receive from the remote side.
|
|
*/
|
|
if (!dtls->ssl || (dtls->dtls_setup != AST_RTP_DTLS_SETUP_ACTIVE)) {
|
|
return;
|
|
}
|
|
|
|
SSL_do_handshake(dtls->ssl);
|
|
|
|
/* Since the handshake is started in a thread outside of the channel thread it's possible
|
|
* for the response to be handled in the channel thread before we start the timeout timer.
|
|
* To ensure this doesn't actually happen we hold the DTLS lock. The channel thread will
|
|
* block until we're done at which point the timeout timer will be immediately stopped.
|
|
*/
|
|
ast_mutex_lock(&dtls->lock);
|
|
dtls_srtp_check_pending(instance, rtp, rtcp);
|
|
dtls_srtp_start_timeout_timer(instance, rtp, rtcp);
|
|
ast_mutex_unlock(&dtls->lock);
|
|
}
|
|
#endif
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
static void rtp_learning_seq_init(struct rtp_learning_info *info, uint16_t seq);
|
|
|
|
static void ast_rtp_on_ice_complete(pj_ice_sess *ice, pj_status_t status)
|
|
{
|
|
struct ast_rtp_instance *instance = ice->user_data;
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (status == PJ_SUCCESS) {
|
|
struct ast_sockaddr remote_address;
|
|
|
|
/* Symmetric RTP must be disabled for the remote address to not get overwritten */
|
|
ast_rtp_instance_set_prop(instance, AST_RTP_PROPERTY_NAT, 0);
|
|
|
|
update_address_with_ice_candidate(rtp, AST_RTP_ICE_COMPONENT_RTP, &remote_address);
|
|
ast_rtp_instance_set_remote_address(instance, &remote_address);
|
|
|
|
if (rtp->rtcp) {
|
|
update_address_with_ice_candidate(rtp, AST_RTP_ICE_COMPONENT_RTCP, &rtp->rtcp->them);
|
|
}
|
|
}
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
dtls_perform_handshake(instance, &rtp->dtls, 0);
|
|
|
|
if (rtp->rtcp) {
|
|
dtls_perform_handshake(instance, &rtp->rtcp->dtls, 1);
|
|
}
|
|
#endif
|
|
|
|
if (!strictrtp) {
|
|
return;
|
|
}
|
|
|
|
rtp->strict_rtp_state = STRICT_RTP_LEARN;
|
|
rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t)rtp->seqno);
|
|
}
|
|
|
|
static void ast_rtp_on_ice_rx_data(pj_ice_sess *ice, unsigned comp_id, unsigned transport_id, void *pkt, pj_size_t size, const pj_sockaddr_t *src_addr, unsigned src_addr_len)
|
|
{
|
|
struct ast_rtp_instance *instance = ice->user_data;
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
/* Instead of handling the packet here (which really doesn't work with our architecture) we set a bit to indicate that it should be handled after pj_ice_sess_on_rx_pkt
|
|
* returns */
|
|
if (transport_id == TRANSPORT_SOCKET_RTP || transport_id == TRANSPORT_SOCKET_RTCP) {
|
|
rtp->passthrough = 1;
|
|
} else if (transport_id == TRANSPORT_TURN_RTP) {
|
|
rtp->rtp_passthrough = 1;
|
|
} else if (transport_id == TRANSPORT_TURN_RTCP) {
|
|
rtp->rtcp_passthrough = 1;
|
|
}
|
|
}
|
|
|
|
static pj_status_t ast_rtp_on_ice_tx_pkt(pj_ice_sess *ice, unsigned comp_id, unsigned transport_id, const void *pkt, pj_size_t size, const pj_sockaddr_t *dst_addr, unsigned dst_addr_len)
|
|
{
|
|
struct ast_rtp_instance *instance = ice->user_data;
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
pj_status_t status = PJ_EINVALIDOP;
|
|
pj_ssize_t _size = (pj_ssize_t)size;
|
|
|
|
if (transport_id == TRANSPORT_SOCKET_RTP) {
|
|
/* Traffic is destined to go right out the RTP socket we already have */
|
|
status = pj_sock_sendto(rtp->s, pkt, &_size, 0, dst_addr, dst_addr_len);
|
|
/* sendto on a connectionless socket should send all the data, or none at all */
|
|
ast_assert(_size == size || status != PJ_SUCCESS);
|
|
} else if (transport_id == TRANSPORT_SOCKET_RTCP) {
|
|
/* Traffic is destined to go right out the RTCP socket we already have */
|
|
if (rtp->rtcp) {
|
|
status = pj_sock_sendto(rtp->rtcp->s, pkt, &_size, 0, dst_addr, dst_addr_len);
|
|
/* sendto on a connectionless socket should send all the data, or none at all */
|
|
ast_assert(_size == size || status != PJ_SUCCESS);
|
|
} else {
|
|
status = PJ_SUCCESS;
|
|
}
|
|
} else if (transport_id == TRANSPORT_TURN_RTP) {
|
|
/* Traffic is going through the RTP TURN relay */
|
|
if (rtp->turn_rtp) {
|
|
status = pj_turn_sock_sendto(rtp->turn_rtp, pkt, size, dst_addr, dst_addr_len);
|
|
}
|
|
} else if (transport_id == TRANSPORT_TURN_RTCP) {
|
|
/* Traffic is going through the RTCP TURN relay */
|
|
if (rtp->turn_rtcp) {
|
|
status = pj_turn_sock_sendto(rtp->turn_rtcp, pkt, size, dst_addr, dst_addr_len);
|
|
}
|
|
}
|
|
|
|
return status;
|
|
}
|
|
|
|
/* ICE Session interface declaration */
|
|
static pj_ice_sess_cb ast_rtp_ice_sess_cb = {
|
|
.on_ice_complete = ast_rtp_on_ice_complete,
|
|
.on_rx_data = ast_rtp_on_ice_rx_data,
|
|
.on_tx_pkt = ast_rtp_on_ice_tx_pkt,
|
|
};
|
|
|
|
/*! \brief Worker thread for timerheap */
|
|
static int timer_worker_thread(void *data)
|
|
{
|
|
pj_ioqueue_t *ioqueue;
|
|
|
|
if (pj_ioqueue_create(pool, 1, &ioqueue) != PJ_SUCCESS) {
|
|
return -1;
|
|
}
|
|
|
|
while (!timer_terminate) {
|
|
const pj_time_val delay = {0, 10};
|
|
|
|
pj_timer_heap_poll(timer_heap, NULL);
|
|
pj_ioqueue_poll(ioqueue, &delay);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
#endif
|
|
|
|
static inline int rtp_debug_test_addr(struct ast_sockaddr *addr)
|
|
{
|
|
if (!rtpdebug) {
|
|
return 0;
|
|
}
|
|
if (!ast_sockaddr_isnull(&rtpdebugaddr)) {
|
|
if (rtpdebugport) {
|
|
return (ast_sockaddr_cmp(&rtpdebugaddr, addr) == 0); /* look for RTP packets from IP+Port */
|
|
} else {
|
|
return (ast_sockaddr_cmp_addr(&rtpdebugaddr, addr) == 0); /* only look for RTP packets from IP */
|
|
}
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
static inline int rtcp_debug_test_addr(struct ast_sockaddr *addr)
|
|
{
|
|
if (!rtcpdebug) {
|
|
return 0;
|
|
}
|
|
if (!ast_sockaddr_isnull(&rtcpdebugaddr)) {
|
|
if (rtcpdebugport) {
|
|
return (ast_sockaddr_cmp(&rtcpdebugaddr, addr) == 0); /* look for RTCP packets from IP+Port */
|
|
} else {
|
|
return (ast_sockaddr_cmp_addr(&rtcpdebugaddr, addr) == 0); /* only look for RTCP packets from IP */
|
|
}
|
|
}
|
|
|
|
return 1;
|
|
}
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
static int dtls_srtp_handle_timeout(struct ast_rtp_instance *instance, int rtcp)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct dtls_details *dtls = !rtcp ? &rtp->dtls : &rtp->rtcp->dtls;
|
|
struct timeval dtls_timeout;
|
|
|
|
DTLSv1_handle_timeout(dtls->ssl);
|
|
dtls_srtp_check_pending(instance, rtp, rtcp);
|
|
|
|
/* If a timeout can't be retrieved then this recurring scheduled item must stop */
|
|
if (!DTLSv1_get_timeout(dtls->ssl, &dtls_timeout)) {
|
|
dtls->timeout_timer = -1;
|
|
return 0;
|
|
}
|
|
|
|
return dtls_timeout.tv_sec * 1000 + dtls_timeout.tv_usec / 1000;
|
|
}
|
|
|
|
static int dtls_srtp_handle_rtp_timeout(const void *data)
|
|
{
|
|
struct ast_rtp_instance *instance = (struct ast_rtp_instance *)data;
|
|
int reschedule;
|
|
|
|
reschedule = dtls_srtp_handle_timeout(instance, 0);
|
|
|
|
if (!reschedule) {
|
|
ao2_ref(instance, -1);
|
|
}
|
|
|
|
return reschedule;
|
|
}
|
|
|
|
static int dtls_srtp_handle_rtcp_timeout(const void *data)
|
|
{
|
|
struct ast_rtp_instance *instance = (struct ast_rtp_instance *)data;
|
|
int reschedule;
|
|
|
|
reschedule = dtls_srtp_handle_timeout(instance, 1);
|
|
|
|
if (!reschedule) {
|
|
ao2_ref(instance, -1);
|
|
}
|
|
|
|
return reschedule;
|
|
}
|
|
|
|
static void dtls_srtp_start_timeout_timer(struct ast_rtp_instance *instance, struct ast_rtp *rtp, int rtcp)
|
|
{
|
|
struct dtls_details *dtls = !rtcp ? &rtp->dtls : &rtp->rtcp->dtls;
|
|
struct timeval dtls_timeout;
|
|
|
|
if (DTLSv1_get_timeout(dtls->ssl, &dtls_timeout)) {
|
|
int timeout = dtls_timeout.tv_sec * 1000 + dtls_timeout.tv_usec / 1000;
|
|
|
|
ast_assert(dtls->timeout_timer == -1);
|
|
|
|
ao2_ref(instance, +1);
|
|
if ((dtls->timeout_timer = ast_sched_add(rtp->sched, timeout,
|
|
!rtcp ? dtls_srtp_handle_rtp_timeout : dtls_srtp_handle_rtcp_timeout, instance)) < 0) {
|
|
ao2_ref(instance, -1);
|
|
ast_log(LOG_WARNING, "Scheduling '%s' DTLS retransmission for RTP instance [%p] failed.\n",
|
|
!rtcp ? "RTP" : "RTCP", instance);
|
|
}
|
|
}
|
|
}
|
|
|
|
static void dtls_srtp_stop_timeout_timer(struct ast_rtp_instance *instance, struct ast_rtp *rtp, int rtcp)
|
|
{
|
|
struct dtls_details *dtls = !rtcp ? &rtp->dtls : &rtp->rtcp->dtls;
|
|
|
|
AST_SCHED_DEL_UNREF(rtp->sched, dtls->timeout_timer, ao2_ref(instance, -1));
|
|
}
|
|
|
|
static void dtls_srtp_check_pending(struct ast_rtp_instance *instance, struct ast_rtp *rtp, int rtcp)
|
|
{
|
|
struct dtls_details *dtls = !rtcp ? &rtp->dtls : &rtp->rtcp->dtls;
|
|
size_t pending;
|
|
|
|
if (!dtls->ssl || !dtls->write_bio) {
|
|
return;
|
|
}
|
|
|
|
pending = BIO_ctrl_pending(dtls->write_bio);
|
|
|
|
if (pending > 0) {
|
|
char outgoing[pending];
|
|
size_t out;
|
|
struct ast_sockaddr remote_address = { {0, } };
|
|
int ice;
|
|
|
|
if (!rtcp) {
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
} else {
|
|
ast_sockaddr_copy(&remote_address, &rtp->rtcp->them);
|
|
}
|
|
|
|
/* If we do not yet know an address to send this to defer it until we do */
|
|
if (ast_sockaddr_isnull(&remote_address)) {
|
|
return;
|
|
}
|
|
|
|
out = BIO_read(dtls->write_bio, outgoing, sizeof(outgoing));
|
|
__rtp_sendto(instance, outgoing, out, 0, &remote_address, rtcp, &ice, 0);
|
|
}
|
|
}
|
|
|
|
static int dtls_srtp_renegotiate(const void *data)
|
|
{
|
|
struct ast_rtp_instance *instance = (struct ast_rtp_instance *)data;
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
SSL_renegotiate(rtp->dtls.ssl);
|
|
SSL_do_handshake(rtp->dtls.ssl);
|
|
dtls_srtp_check_pending(instance, rtp, 0);
|
|
|
|
if (rtp->rtcp && rtp->rtcp->dtls.ssl) {
|
|
SSL_renegotiate(rtp->rtcp->dtls.ssl);
|
|
SSL_do_handshake(rtp->rtcp->dtls.ssl);
|
|
dtls_srtp_check_pending(instance, rtp, 1);
|
|
}
|
|
|
|
rtp->rekeyid = -1;
|
|
ao2_ref(instance, -1);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int dtls_srtp_setup(struct ast_rtp *rtp, struct ast_srtp *srtp, struct ast_rtp_instance *instance)
|
|
{
|
|
unsigned char material[SRTP_MASTER_LEN * 2];
|
|
unsigned char *local_key, *local_salt, *remote_key, *remote_salt;
|
|
struct ast_srtp_policy *local_policy, *remote_policy = NULL;
|
|
struct ast_rtp_instance_stats stats = { 0, };
|
|
int res = -1;
|
|
|
|
/* If a fingerprint is present in the SDP make sure that the peer certificate matches it */
|
|
if (rtp->dtls_verify & AST_RTP_DTLS_VERIFY_FINGERPRINT) {
|
|
X509 *certificate;
|
|
|
|
if (!(certificate = SSL_get_peer_certificate(rtp->dtls.ssl))) {
|
|
ast_log(LOG_WARNING, "No certificate was provided by the peer on RTP instance '%p'\n", instance);
|
|
return -1;
|
|
}
|
|
|
|
/* If a fingerprint is present in the SDP make sure that the peer certificate matches it */
|
|
if (rtp->remote_fingerprint[0]) {
|
|
const EVP_MD *type;
|
|
unsigned char fingerprint[EVP_MAX_MD_SIZE];
|
|
unsigned int size;
|
|
|
|
if (rtp->remote_hash == AST_RTP_DTLS_HASH_SHA1) {
|
|
type = EVP_sha1();
|
|
} else if (rtp->remote_hash == AST_RTP_DTLS_HASH_SHA256) {
|
|
type = EVP_sha256();
|
|
} else {
|
|
ast_log(LOG_WARNING, "Unsupported fingerprint hash type on RTP instance '%p'\n", instance);
|
|
return -1;
|
|
}
|
|
|
|
if (!X509_digest(certificate, type, fingerprint, &size) ||
|
|
!size ||
|
|
memcmp(fingerprint, rtp->remote_fingerprint, size)) {
|
|
X509_free(certificate);
|
|
ast_log(LOG_WARNING, "Fingerprint provided by remote party does not match that of peer certificate on RTP instance '%p'\n",
|
|
instance);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
X509_free(certificate);
|
|
}
|
|
|
|
/* Ensure that certificate verification was successful */
|
|
if ((rtp->dtls_verify & AST_RTP_DTLS_VERIFY_CERTIFICATE) && SSL_get_verify_result(rtp->dtls.ssl) != X509_V_OK) {
|
|
ast_log(LOG_WARNING, "Peer certificate on RTP instance '%p' failed verification test\n",
|
|
instance);
|
|
return -1;
|
|
}
|
|
|
|
/* Produce key information and set up SRTP */
|
|
if (!SSL_export_keying_material(rtp->dtls.ssl, material, SRTP_MASTER_LEN * 2, "EXTRACTOR-dtls_srtp", 19, NULL, 0, 0)) {
|
|
ast_log(LOG_WARNING, "Unable to extract SRTP keying material from DTLS-SRTP negotiation on RTP instance '%p'\n",
|
|
instance);
|
|
return -1;
|
|
}
|
|
|
|
/* Whether we are acting as a server or client determines where the keys/salts are */
|
|
if (rtp->dtls.dtls_setup == AST_RTP_DTLS_SETUP_ACTIVE) {
|
|
local_key = material;
|
|
remote_key = local_key + SRTP_MASTER_KEY_LEN;
|
|
local_salt = remote_key + SRTP_MASTER_KEY_LEN;
|
|
remote_salt = local_salt + SRTP_MASTER_SALT_LEN;
|
|
} else {
|
|
remote_key = material;
|
|
local_key = remote_key + SRTP_MASTER_KEY_LEN;
|
|
remote_salt = local_key + SRTP_MASTER_KEY_LEN;
|
|
local_salt = remote_salt + SRTP_MASTER_SALT_LEN;
|
|
}
|
|
|
|
if (!(local_policy = res_srtp_policy->alloc())) {
|
|
return -1;
|
|
}
|
|
|
|
if (res_srtp_policy->set_master_key(local_policy, local_key, SRTP_MASTER_KEY_LEN, local_salt, SRTP_MASTER_SALT_LEN) < 0) {
|
|
ast_log(LOG_WARNING, "Could not set key/salt information on local policy of '%p' when setting up DTLS-SRTP\n", rtp);
|
|
goto error;
|
|
}
|
|
|
|
if (res_srtp_policy->set_suite(local_policy, rtp->suite)) {
|
|
ast_log(LOG_WARNING, "Could not set suite to '%u' on local policy of '%p' when setting up DTLS-SRTP\n", rtp->suite, rtp);
|
|
goto error;
|
|
}
|
|
|
|
if (ast_rtp_instance_get_stats(instance, &stats, AST_RTP_INSTANCE_STAT_LOCAL_SSRC)) {
|
|
goto error;
|
|
}
|
|
|
|
res_srtp_policy->set_ssrc(local_policy, stats.local_ssrc, 0);
|
|
|
|
if (!(remote_policy = res_srtp_policy->alloc())) {
|
|
goto error;
|
|
}
|
|
|
|
if (res_srtp_policy->set_master_key(remote_policy, remote_key, SRTP_MASTER_KEY_LEN, remote_salt, SRTP_MASTER_SALT_LEN) < 0) {
|
|
ast_log(LOG_WARNING, "Could not set key/salt information on remote policy of '%p' when setting up DTLS-SRTP\n", rtp);
|
|
goto error;
|
|
}
|
|
|
|
if (res_srtp_policy->set_suite(remote_policy, rtp->suite)) {
|
|
ast_log(LOG_WARNING, "Could not set suite to '%u' on remote policy of '%p' when setting up DTLS-SRTP\n", rtp->suite, rtp);
|
|
goto error;
|
|
}
|
|
|
|
res_srtp_policy->set_ssrc(remote_policy, 0, 1);
|
|
|
|
if (ast_rtp_instance_add_srtp_policy(instance, remote_policy, local_policy)) {
|
|
ast_log(LOG_WARNING, "Could not set policies when setting up DTLS-SRTP on '%p'\n", rtp);
|
|
goto error;
|
|
}
|
|
|
|
if (rtp->rekey) {
|
|
ao2_ref(instance, +1);
|
|
if ((rtp->rekeyid = ast_sched_add(rtp->sched, rtp->rekey * 1000, dtls_srtp_renegotiate, instance)) < 0) {
|
|
ao2_ref(instance, -1);
|
|
goto error;
|
|
}
|
|
}
|
|
|
|
res = 0;
|
|
|
|
error:
|
|
/* policy->destroy() called even on success to release local reference to these resources */
|
|
res_srtp_policy->destroy(local_policy);
|
|
|
|
if (remote_policy) {
|
|
res_srtp_policy->destroy(remote_policy);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
#endif
|
|
|
|
static int __rtp_recvfrom(struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int rtcp)
|
|
{
|
|
int len;
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance);
|
|
char *in = buf;
|
|
#ifdef HAVE_PJPROJECT
|
|
struct ast_sockaddr *loop = rtcp ? &rtp->rtcp_loop : &rtp->rtp_loop;
|
|
#endif
|
|
|
|
if ((len = ast_recvfrom(rtcp ? rtp->rtcp->s : rtp->s, buf, size, flags, sa)) < 0) {
|
|
return len;
|
|
}
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
/* If this is an SSL packet pass it to OpenSSL for processing. RFC section for first byte value:
|
|
* https://tools.ietf.org/html/rfc5764#section-5.1.2 */
|
|
if ((*in >= 20) && (*in <= 63)) {
|
|
struct dtls_details *dtls = !rtcp ? &rtp->dtls : &rtp->rtcp->dtls;
|
|
int res = 0;
|
|
|
|
/* If no SSL session actually exists terminate things */
|
|
if (!dtls->ssl) {
|
|
ast_log(LOG_ERROR, "Received SSL traffic on RTP instance '%p' without an SSL session\n",
|
|
instance);
|
|
return -1;
|
|
}
|
|
|
|
/* This mutex is locked so that this thread blocks until the dtls_perform_handshake function
|
|
* completes.
|
|
*/
|
|
ast_mutex_lock(&dtls->lock);
|
|
ast_mutex_unlock(&dtls->lock);
|
|
|
|
/* Before we feed data into OpenSSL ensure that the timeout timer is either stopped or completed */
|
|
dtls_srtp_stop_timeout_timer(instance, rtp, rtcp);
|
|
|
|
/* If we don't yet know if we are active or passive and we receive a packet... we are obviously passive */
|
|
if (dtls->dtls_setup == AST_RTP_DTLS_SETUP_ACTPASS) {
|
|
dtls->dtls_setup = AST_RTP_DTLS_SETUP_PASSIVE;
|
|
SSL_set_accept_state(dtls->ssl);
|
|
}
|
|
|
|
dtls_srtp_check_pending(instance, rtp, rtcp);
|
|
|
|
BIO_write(dtls->read_bio, buf, len);
|
|
|
|
len = SSL_read(dtls->ssl, buf, len);
|
|
|
|
if ((len < 0) && (SSL_get_error(dtls->ssl, len) == SSL_ERROR_SSL)) {
|
|
unsigned long error = ERR_get_error();
|
|
ast_log(LOG_ERROR, "DTLS failure occurred on RTP instance '%p' due to reason '%s', terminating\n",
|
|
instance, ERR_reason_error_string(error));
|
|
return -1;
|
|
}
|
|
|
|
dtls_srtp_check_pending(instance, rtp, rtcp);
|
|
|
|
if (SSL_is_init_finished(dtls->ssl)) {
|
|
/* Any further connections will be existing since this is now established */
|
|
dtls->connection = AST_RTP_DTLS_CONNECTION_EXISTING;
|
|
if (!rtcp) {
|
|
/* Use the keying material to set up key/salt information */
|
|
res = dtls_srtp_setup(rtp, srtp, instance);
|
|
}
|
|
} else {
|
|
/* Since we've sent additional traffic start the timeout timer for retransmission */
|
|
dtls_srtp_start_timeout_timer(instance, rtp, rtcp);
|
|
}
|
|
|
|
return res;
|
|
}
|
|
#endif
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
if (!ast_sockaddr_isnull(loop) && !ast_sockaddr_cmp(loop, sa)) {
|
|
/* ICE traffic will have been handled in the TURN callback, so skip it but update the address
|
|
* so it reflects the actual source and not the loopback
|
|
*/
|
|
if (rtcp) {
|
|
ast_sockaddr_copy(sa, &rtp->rtcp->them);
|
|
} else {
|
|
ast_rtp_instance_get_remote_address(instance, sa);
|
|
}
|
|
} else if (rtp->ice) {
|
|
pj_str_t combined = pj_str(ast_sockaddr_stringify(sa));
|
|
pj_sockaddr address;
|
|
pj_status_t status;
|
|
|
|
pj_thread_register_check();
|
|
|
|
pj_sockaddr_parse(pj_AF_UNSPEC(), 0, &combined, &address);
|
|
|
|
status = pj_ice_sess_on_rx_pkt(rtp->ice, rtcp ? AST_RTP_ICE_COMPONENT_RTCP : AST_RTP_ICE_COMPONENT_RTP,
|
|
rtcp ? TRANSPORT_SOCKET_RTCP : TRANSPORT_SOCKET_RTP, buf, len, &address,
|
|
pj_sockaddr_get_len(&address));
|
|
if (status != PJ_SUCCESS) {
|
|
char buf[100];
|
|
|
|
pj_strerror(status, buf, sizeof(buf));
|
|
ast_log(LOG_WARNING, "PJ ICE Rx error status code: %d '%s'.\n",
|
|
(int)status, buf);
|
|
return -1;
|
|
}
|
|
if (!rtp->passthrough) {
|
|
return 0;
|
|
}
|
|
rtp->passthrough = 0;
|
|
}
|
|
#endif
|
|
|
|
if ((*in & 0xC0) && res_srtp && srtp && res_srtp->unprotect(srtp, buf, &len, rtcp) < 0) {
|
|
return -1;
|
|
}
|
|
|
|
return len;
|
|
}
|
|
|
|
static int rtcp_recvfrom(struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa)
|
|
{
|
|
return __rtp_recvfrom(instance, buf, size, flags, sa, 1);
|
|
}
|
|
|
|
static int rtp_recvfrom(struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa)
|
|
{
|
|
return __rtp_recvfrom(instance, buf, size, flags, sa, 0);
|
|
}
|
|
|
|
static int __rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int rtcp, int *ice, int use_srtp)
|
|
{
|
|
int len = size;
|
|
void *temp = buf;
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance);
|
|
int res;
|
|
|
|
*ice = 0;
|
|
|
|
if (use_srtp && res_srtp && srtp && res_srtp->protect(srtp, &temp, &len, rtcp) < 0) {
|
|
return -1;
|
|
}
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
if (rtp->ice) {
|
|
pj_thread_register_check();
|
|
|
|
if (pj_ice_sess_send_data(rtp->ice, rtcp ? AST_RTP_ICE_COMPONENT_RTCP : AST_RTP_ICE_COMPONENT_RTP, temp, len) == PJ_SUCCESS) {
|
|
*ice = 1;
|
|
return len;
|
|
}
|
|
}
|
|
#endif
|
|
|
|
res = ast_sendto(rtcp ? rtp->rtcp->s : rtp->s, temp, len, flags, sa);
|
|
if (res > 0) {
|
|
ast_rtp_instance_set_last_tx(instance, time(NULL));
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static int rtcp_sendto(struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int *ice)
|
|
{
|
|
return __rtp_sendto(instance, buf, size, flags, sa, 1, ice, 1);
|
|
}
|
|
|
|
static int rtp_sendto(struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int *ice)
|
|
{
|
|
return __rtp_sendto(instance, buf, size, flags, sa, 0, ice, 1);
|
|
}
|
|
|
|
static int rtp_get_rate(struct ast_format *format)
|
|
{
|
|
/* For those wondering: due to a fluke in RFC publication, G.722 is advertised
|
|
* as having a sample rate of 8kHz, while implementations must know that its
|
|
* real rate is 16kHz. Seriously.
|
|
*/
|
|
return (ast_format_cmp(format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) ? 8000 : (int)ast_format_get_sample_rate(format);
|
|
}
|
|
|
|
static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp)
|
|
{
|
|
unsigned int interval;
|
|
/*! \todo XXX Do a more reasonable calculation on this one
|
|
* Look in RFC 3550 Section A.7 for an example*/
|
|
interval = rtcpinterval;
|
|
return interval;
|
|
}
|
|
|
|
/*! \brief Calculate normal deviation */
|
|
static double normdev_compute(double normdev, double sample, unsigned int sample_count)
|
|
{
|
|
normdev = normdev * sample_count + sample;
|
|
sample_count++;
|
|
|
|
return normdev / sample_count;
|
|
}
|
|
|
|
static double stddev_compute(double stddev, double sample, double normdev, double normdev_curent, unsigned int sample_count)
|
|
{
|
|
/*
|
|
for the formula check http://www.cs.umd.edu/~austinjp/constSD.pdf
|
|
return sqrt( (sample_count*pow(stddev,2) + sample_count*pow((sample-normdev)/(sample_count+1),2) + pow(sample-normdev_curent,2)) / (sample_count+1));
|
|
we can compute the sigma^2 and that way we would have to do the sqrt only 1 time at the end and would save another pow 2 compute
|
|
optimized formula
|
|
*/
|
|
#define SQUARE(x) ((x) * (x))
|
|
|
|
stddev = sample_count * stddev;
|
|
sample_count++;
|
|
|
|
return stddev +
|
|
( sample_count * SQUARE( (sample - normdev) / sample_count ) ) +
|
|
( SQUARE(sample - normdev_curent) / sample_count );
|
|
|
|
#undef SQUARE
|
|
}
|
|
|
|
static int create_new_socket(const char *type, int af)
|
|
{
|
|
int sock = socket(af, SOCK_DGRAM, 0);
|
|
|
|
if (sock < 0) {
|
|
if (!type) {
|
|
type = "RTP/RTCP";
|
|
}
|
|
ast_log(LOG_WARNING, "Unable to allocate %s socket: %s\n", type, strerror(errno));
|
|
} else {
|
|
long flags = fcntl(sock, F_GETFL);
|
|
fcntl(sock, F_SETFL, flags | O_NONBLOCK);
|
|
#ifdef SO_NO_CHECK
|
|
if (nochecksums) {
|
|
setsockopt(sock, SOL_SOCKET, SO_NO_CHECK, &nochecksums, sizeof(nochecksums));
|
|
}
|
|
#endif
|
|
}
|
|
|
|
return sock;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Initializes sequence values and probation for learning mode.
|
|
* \note This is an adaptation of pjmedia's pjmedia_rtp_seq_init function.
|
|
*
|
|
* \param info The learning information to track
|
|
* \param seq sequence number read from the rtp header to initialize the information with
|
|
*/
|
|
static void rtp_learning_seq_init(struct rtp_learning_info *info, uint16_t seq)
|
|
{
|
|
info->max_seq = seq - 1;
|
|
info->packets = learning_min_sequential;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Updates sequence information for learning mode and determines if probation/learning mode should remain in effect.
|
|
* \note This function was adapted from pjmedia's pjmedia_rtp_seq_update function.
|
|
*
|
|
* \param info Structure tracking the learning progress of some address
|
|
* \param seq sequence number read from the rtp header
|
|
* \retval 0 if probation mode should exit for this address
|
|
* \retval non-zero if probation mode should continue
|
|
*/
|
|
static int rtp_learning_rtp_seq_update(struct rtp_learning_info *info, uint16_t seq)
|
|
{
|
|
if (seq == info->max_seq + 1) {
|
|
/* packet is in sequence */
|
|
info->packets--;
|
|
} else {
|
|
/* Sequence discontinuity; reset */
|
|
info->packets = learning_min_sequential - 1;
|
|
}
|
|
info->max_seq = seq;
|
|
|
|
return (info->packets == 0);
|
|
}
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
static void rtp_add_candidates_to_ice(struct ast_rtp_instance *instance, struct ast_rtp *rtp, struct ast_sockaddr *addr, int port, int component,
|
|
int transport)
|
|
{
|
|
pj_sockaddr address[16];
|
|
unsigned int count = PJ_ARRAY_SIZE(address), pos = 0;
|
|
|
|
/* Add all the local interface IP addresses */
|
|
if (ast_sockaddr_is_ipv4(addr)) {
|
|
pj_enum_ip_interface(pj_AF_INET(), &count, address);
|
|
} else if (ast_sockaddr_is_any(addr)) {
|
|
pj_enum_ip_interface(pj_AF_UNSPEC(), &count, address);
|
|
} else {
|
|
pj_enum_ip_interface(pj_AF_INET6(), &count, address);
|
|
}
|
|
|
|
for (pos = 0; pos < count; pos++) {
|
|
pj_sockaddr_set_port(&address[pos], port);
|
|
ast_rtp_ice_add_cand(rtp, component, transport, PJ_ICE_CAND_TYPE_HOST, 65535, &address[pos], &address[pos], NULL,
|
|
pj_sockaddr_get_len(&address[pos]));
|
|
}
|
|
|
|
/* If configured to use a STUN server to get our external mapped address do so */
|
|
if (stunaddr.sin_addr.s_addr && ast_sockaddr_is_ipv4(addr) && count) {
|
|
struct sockaddr_in answer;
|
|
|
|
if (!ast_stun_request(component == AST_RTP_ICE_COMPONENT_RTCP ? rtp->rtcp->s : rtp->s, &stunaddr, NULL, &answer)) {
|
|
pj_sockaddr base;
|
|
pj_str_t mapped = pj_str(ast_strdupa(ast_inet_ntoa(answer.sin_addr)));
|
|
|
|
/* Use the first local host candidate as the base */
|
|
pj_sockaddr_cp(&base, &address[0]);
|
|
|
|
pj_sockaddr_init(pj_AF_INET(), &address[0], &mapped, ntohs(answer.sin_port));
|
|
|
|
ast_rtp_ice_add_cand(rtp, component, transport, PJ_ICE_CAND_TYPE_SRFLX, 65535, &address[0], &base,
|
|
&base, pj_sockaddr_get_len(&address[0]));
|
|
}
|
|
}
|
|
|
|
/* If configured to use a TURN relay create a session and allocate */
|
|
if (pj_strlen(&turnaddr)) {
|
|
ast_rtp_ice_turn_request(instance, component, AST_TRANSPORT_TCP, pj_strbuf(&turnaddr), turnport,
|
|
pj_strbuf(&turnusername), pj_strbuf(&turnpassword));
|
|
}
|
|
}
|
|
#endif
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Calculates the elapsed time from issue of the first tx packet in an
|
|
* rtp session and a specified time
|
|
*
|
|
* \param rtp pointer to the rtp struct with the transmitted rtp packet
|
|
* \param delivery time of delivery - if NULL or zero value, will be ast_tvnow()
|
|
*
|
|
* \return time elapsed in milliseconds
|
|
*/
|
|
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();
|
|
rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
|
|
}
|
|
|
|
t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
|
|
if ((ms = ast_tvdiff_ms(t, rtp->txcore)) < 0) {
|
|
ms = 0;
|
|
}
|
|
rtp->txcore = t;
|
|
|
|
return (unsigned int) ms;
|
|
}
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
/*!
|
|
* \internal
|
|
* \brief Creates an ICE session. Can be used to replace a destroyed ICE session.
|
|
*
|
|
* \param instance RTP instance for which the ICE session is being replaced
|
|
* \param addr ast_sockaddr to use for adding RTP candidates to the ICE session
|
|
* \param port port to use for adding RTP candidates to the ICE session
|
|
* \param replace 0 when creating a new session, 1 when replacing a destroyed session
|
|
*
|
|
* \retval 0 on success
|
|
* \retval -1 on failure
|
|
*/
|
|
static int ice_create(struct ast_rtp_instance *instance, struct ast_sockaddr *addr,
|
|
int port, int replace)
|
|
{
|
|
pj_stun_config stun_config;
|
|
pj_str_t ufrag, passwd;
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
ao2_cleanup(rtp->ice_local_candidates);
|
|
rtp->ice_local_candidates = NULL;
|
|
|
|
pj_thread_register_check();
|
|
|
|
pj_stun_config_init(&stun_config, &cachingpool.factory, 0, NULL, timer_heap);
|
|
|
|
ufrag = pj_str(rtp->local_ufrag);
|
|
passwd = pj_str(rtp->local_passwd);
|
|
|
|
/* Create an ICE session for ICE negotiation */
|
|
if (pj_ice_sess_create(&stun_config, NULL, PJ_ICE_SESS_ROLE_UNKNOWN, 2,
|
|
&ast_rtp_ice_sess_cb, &ufrag, &passwd, NULL, &rtp->ice) == PJ_SUCCESS) {
|
|
/* Make this available for the callbacks */
|
|
rtp->ice->user_data = instance;
|
|
|
|
/* Add all of the available candidates to the ICE session */
|
|
rtp_add_candidates_to_ice(instance, rtp, addr, port, AST_RTP_ICE_COMPONENT_RTP,
|
|
TRANSPORT_SOCKET_RTP);
|
|
|
|
/* Only add the RTCP candidates to ICE when replacing the session. New sessions
|
|
* handle this in a separate part of the setup phase */
|
|
if (replace && rtp->rtcp) {
|
|
rtp_add_candidates_to_ice(instance, rtp, &rtp->rtcp->us,
|
|
ast_sockaddr_port(&rtp->rtcp->us), AST_RTP_ICE_COMPONENT_RTCP,
|
|
TRANSPORT_SOCKET_RTCP);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
return -1;
|
|
|
|
}
|
|
#endif
|
|
|
|
static int ast_rtp_new(struct ast_rtp_instance *instance,
|
|
struct ast_sched_context *sched, struct ast_sockaddr *addr,
|
|
void *data)
|
|
{
|
|
struct ast_rtp *rtp = NULL;
|
|
int x, startplace;
|
|
|
|
/* Create a new RTP structure to hold all of our data */
|
|
if (!(rtp = ast_calloc(1, sizeof(*rtp)))) {
|
|
return -1;
|
|
}
|
|
|
|
/* Initialize synchronization aspects */
|
|
ast_mutex_init(&rtp->lock);
|
|
ast_cond_init(&rtp->cond, NULL);
|
|
|
|
/* Set default parameters on the newly created RTP structure */
|
|
rtp->ssrc = ast_random();
|
|
rtp->seqno = ast_random() & 0xffff;
|
|
rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
|
|
if (strictrtp) {
|
|
rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t)rtp->seqno);
|
|
rtp_learning_seq_init(&rtp->alt_source_learn, (uint16_t)rtp->seqno);
|
|
}
|
|
|
|
/* Create a new socket for us to listen on and use */
|
|
if ((rtp->s =
|
|
create_new_socket("RTP",
|
|
ast_sockaddr_is_ipv4(addr) ? AF_INET :
|
|
ast_sockaddr_is_ipv6(addr) ? AF_INET6 : -1)) < 0) {
|
|
ast_debug(1, "Failed to create a new socket for RTP instance '%p'\n", instance);
|
|
ast_free(rtp);
|
|
return -1;
|
|
}
|
|
|
|
/* Now actually find a free RTP port to use */
|
|
x = (rtpend == rtpstart) ? rtpstart : (ast_random() % (rtpend - rtpstart)) + rtpstart;
|
|
x = x & ~1;
|
|
startplace = x;
|
|
|
|
for (;;) {
|
|
ast_sockaddr_set_port(addr, x);
|
|
/* Try to bind, this will tell us whether the port is available or not */
|
|
if (!ast_bind(rtp->s, addr)) {
|
|
ast_debug(1, "Allocated port %d for RTP instance '%p'\n", x, instance);
|
|
ast_rtp_instance_set_local_address(instance, addr);
|
|
break;
|
|
}
|
|
|
|
x += 2;
|
|
if (x > rtpend) {
|
|
x = (rtpstart + 1) & ~1;
|
|
}
|
|
|
|
/* See if we ran out of ports or if the bind actually failed because of something other than the address being in use */
|
|
if (x == startplace || (errno != EADDRINUSE && errno != EACCES)) {
|
|
ast_log(LOG_ERROR, "Oh dear... we couldn't allocate a port for RTP instance '%p'\n", instance);
|
|
close(rtp->s);
|
|
ast_free(rtp);
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
generate_random_string(rtp->local_ufrag, sizeof(rtp->local_ufrag));
|
|
generate_random_string(rtp->local_passwd, sizeof(rtp->local_passwd));
|
|
#endif
|
|
ast_rtp_instance_set_data(instance, rtp);
|
|
#ifdef HAVE_PJPROJECT
|
|
/* Create an ICE session for ICE negotiation */
|
|
if (icesupport) {
|
|
if (ice_create(instance, addr, x, 0)) {
|
|
ast_log(LOG_NOTICE, "Failed to start ICE session\n");
|
|
} else {
|
|
rtp->ice_port = x;
|
|
ast_sockaddr_copy(&rtp->ice_original_rtp_addr, addr);
|
|
}
|
|
}
|
|
#endif
|
|
|
|
/* Record any information we may need */
|
|
rtp->sched = sched;
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
rtp->rekeyid = -1;
|
|
rtp->dtls.timeout_timer = -1;
|
|
#endif
|
|
|
|
rtp->f.subclass.format = ao2_bump(ast_format_none);
|
|
rtp->lastrxformat = ao2_bump(ast_format_none);
|
|
rtp->lasttxformat = ao2_bump(ast_format_none);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int ast_rtp_destroy(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
#ifdef HAVE_PJPROJECT
|
|
struct timeval wait = ast_tvadd(ast_tvnow(), ast_samp2tv(TURN_STATE_WAIT_TIME, 1000));
|
|
struct timespec ts = { .tv_sec = wait.tv_sec, .tv_nsec = wait.tv_usec * 1000, };
|
|
#endif
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
ast_rtp_dtls_stop(instance);
|
|
#endif
|
|
|
|
/* Destroy the smoother that was smoothing out audio if present */
|
|
if (rtp->smoother) {
|
|
ast_smoother_free(rtp->smoother);
|
|
}
|
|
|
|
/* Close our own socket so we no longer get packets */
|
|
if (rtp->s > -1) {
|
|
close(rtp->s);
|
|
}
|
|
|
|
/* Destroy RTCP if it was being used */
|
|
if (rtp->rtcp) {
|
|
/*
|
|
* It is not possible for there to be an active RTCP scheduler
|
|
* entry at this point since it holds a reference to the
|
|
* RTP instance while it's active.
|
|
*/
|
|
close(rtp->rtcp->s);
|
|
ast_free(rtp->rtcp);
|
|
}
|
|
|
|
/* Destroy RED if it was being used */
|
|
if (rtp->red) {
|
|
AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
|
|
ast_free(rtp->red);
|
|
}
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
pj_thread_register_check();
|
|
|
|
/* Destroy the RTP TURN relay if being used */
|
|
ast_mutex_lock(&rtp->lock);
|
|
if (rtp->turn_rtp) {
|
|
pj_turn_sock_destroy(rtp->turn_rtp);
|
|
rtp->turn_state = PJ_TURN_STATE_NULL;
|
|
while (rtp->turn_state != PJ_TURN_STATE_DESTROYING) {
|
|
ast_cond_timedwait(&rtp->cond, &rtp->lock, &ts);
|
|
}
|
|
}
|
|
|
|
/* Destroy the RTCP TURN relay if being used */
|
|
if (rtp->turn_rtcp) {
|
|
pj_turn_sock_destroy(rtp->turn_rtcp);
|
|
rtp->turn_state = PJ_TURN_STATE_NULL;
|
|
while (rtp->turn_state != PJ_TURN_STATE_DESTROYING) {
|
|
ast_cond_timedwait(&rtp->cond, &rtp->lock, &ts);
|
|
}
|
|
}
|
|
ast_mutex_unlock(&rtp->lock);
|
|
|
|
if (rtp->ioqueue) {
|
|
rtp_ioqueue_thread_remove(rtp->ioqueue);
|
|
}
|
|
|
|
/* Destroy the ICE session if being used */
|
|
if (rtp->ice) {
|
|
pj_ice_sess_destroy(rtp->ice);
|
|
}
|
|
|
|
/* Destroy any candidates */
|
|
if (rtp->ice_local_candidates) {
|
|
ao2_ref(rtp->ice_local_candidates, -1);
|
|
}
|
|
|
|
if (rtp->ice_active_remote_candidates) {
|
|
ao2_ref(rtp->ice_active_remote_candidates, -1);
|
|
}
|
|
#endif
|
|
|
|
ao2_cleanup(rtp->lasttxformat);
|
|
ao2_cleanup(rtp->lastrxformat);
|
|
ao2_cleanup(rtp->f.subclass.format);
|
|
|
|
/* Destroy synchronization items */
|
|
ast_mutex_destroy(&rtp->lock);
|
|
ast_cond_destroy(&rtp->cond);
|
|
|
|
/* Finally destroy ourselves */
|
|
ast_free(rtp);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int ast_rtp_dtmf_mode_set(struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
rtp->dtmfmode = dtmf_mode;
|
|
return 0;
|
|
}
|
|
|
|
static enum ast_rtp_dtmf_mode ast_rtp_dtmf_mode_get(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
return rtp->dtmfmode;
|
|
}
|
|
|
|
static int ast_rtp_dtmf_begin(struct ast_rtp_instance *instance, char digit)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
int hdrlen = 12, res = 0, i = 0, payload = 101;
|
|
char data[256];
|
|
unsigned int *rtpheader = (unsigned int*)data;
|
|
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
|
|
/* If we have no remote address information bail out now */
|
|
if (ast_sockaddr_isnull(&remote_address)) {
|
|
return -1;
|
|
}
|
|
|
|
/* Convert given digit into what we want to transmit */
|
|
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;
|
|
}
|
|
|
|
/* Grab the payload that they expect the RFC2833 packet to be received in */
|
|
payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_DTMF);
|
|
|
|
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
|
|
rtp->send_duration = 160;
|
|
rtp->lastts += calc_txstamp(rtp, NULL) * DTMF_SAMPLE_RATE_MS;
|
|
rtp->lastdigitts = rtp->lastts + rtp->send_duration;
|
|
|
|
/* Create the actual packet that we will be sending */
|
|
rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload << 16) | (rtp->seqno));
|
|
rtpheader[1] = htonl(rtp->lastdigitts);
|
|
rtpheader[2] = htonl(rtp->ssrc);
|
|
|
|
/* Actually send the packet */
|
|
for (i = 0; i < 2; i++) {
|
|
int ice;
|
|
|
|
rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
|
|
res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 4, 0, &remote_address, &ice);
|
|
if (res < 0) {
|
|
ast_log(LOG_ERROR, "RTP Transmission error to %s: %s\n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
strerror(errno));
|
|
}
|
|
if (rtp_debug_test_addr(&remote_address)) {
|
|
ast_verbose("Sent RTP DTMF packet to %s%s (type %-2.2d, seq %-6.6d, ts %-6.6u, len %-6.6d)\n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
ice ? " (via ICE)" : "",
|
|
payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
|
|
}
|
|
rtp->seqno++;
|
|
rtp->send_duration += 160;
|
|
rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
|
|
}
|
|
|
|
/* Record that we are in the process of sending a digit and information needed to continue doing so */
|
|
rtp->sending_digit = 1;
|
|
rtp->send_digit = digit;
|
|
rtp->send_payload = payload;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int ast_rtp_dtmf_continuation(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
int hdrlen = 12, res = 0;
|
|
char data[256];
|
|
unsigned int *rtpheader = (unsigned int*)data;
|
|
int ice;
|
|
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
|
|
/* Make sure we know where the other side is so we can send them the packet */
|
|
if (ast_sockaddr_isnull(&remote_address)) {
|
|
return -1;
|
|
}
|
|
|
|
/* Actually create the packet we will be sending */
|
|
rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
|
|
rtpheader[1] = htonl(rtp->lastdigitts);
|
|
rtpheader[2] = htonl(rtp->ssrc);
|
|
rtpheader[3] = htonl((rtp->send_digit << 24) | (0xa << 16) | (rtp->send_duration));
|
|
|
|
/* Boom, send it on out */
|
|
res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 4, 0, &remote_address, &ice);
|
|
if (res < 0) {
|
|
ast_log(LOG_ERROR, "RTP Transmission error to %s: %s\n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
strerror(errno));
|
|
}
|
|
|
|
if (rtp_debug_test_addr(&remote_address)) {
|
|
ast_verbose("Sent RTP DTMF packet to %s%s (type %-2.2d, seq %-6.6d, ts %-6.6u, len %-6.6d)\n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
ice ? " (via ICE)" : "",
|
|
rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
|
|
}
|
|
|
|
/* And now we increment some values for the next time we swing by */
|
|
rtp->seqno++;
|
|
rtp->send_duration += 160;
|
|
rtp->lastts += calc_txstamp(rtp, NULL) * DTMF_SAMPLE_RATE_MS;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int ast_rtp_dtmf_end_with_duration(struct ast_rtp_instance *instance, char digit, unsigned int duration)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
int hdrlen = 12, res = -1, i = 0;
|
|
char data[256];
|
|
unsigned int *rtpheader = (unsigned int*)data;
|
|
unsigned int measured_samples;
|
|
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
|
|
/* Make sure we know where the remote side is so we can send them the packet we construct */
|
|
if (ast_sockaddr_isnull(&remote_address)) {
|
|
goto cleanup;
|
|
}
|
|
|
|
/* Convert the given digit to the one we are going to send */
|
|
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);
|
|
goto cleanup;
|
|
}
|
|
|
|
rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
|
|
|
|
if (duration > 0 && (measured_samples = duration * rtp_get_rate(rtp->f.subclass.format) / 1000) > rtp->send_duration) {
|
|
ast_debug(2, "Adjusting final end duration from %d to %u\n", rtp->send_duration, measured_samples);
|
|
rtp->send_duration = measured_samples;
|
|
}
|
|
|
|
/* Construct the packet we are going to send */
|
|
rtpheader[1] = htonl(rtp->lastdigitts);
|
|
rtpheader[2] = htonl(rtp->ssrc);
|
|
rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (rtp->send_duration));
|
|
rtpheader[3] |= htonl((1 << 23));
|
|
|
|
/* Send it 3 times, that's the magical number */
|
|
for (i = 0; i < 3; i++) {
|
|
int ice;
|
|
|
|
rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
|
|
|
|
res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 4, 0, &remote_address, &ice);
|
|
|
|
if (res < 0) {
|
|
ast_log(LOG_ERROR, "RTP Transmission error to %s: %s\n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
strerror(errno));
|
|
}
|
|
|
|
if (rtp_debug_test_addr(&remote_address)) {
|
|
ast_verbose("Sent RTP DTMF packet to %s%s (type %-2.2d, seq %-6.6d, ts %-6.6u, len %-6.6d)\n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
ice ? " (via ICE)" : "",
|
|
rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
|
|
}
|
|
|
|
rtp->seqno++;
|
|
}
|
|
res = 0;
|
|
|
|
/* Oh and we can't forget to turn off the stuff that says we are sending DTMF */
|
|
rtp->lastts += calc_txstamp(rtp, NULL) * DTMF_SAMPLE_RATE_MS;
|
|
cleanup:
|
|
rtp->sending_digit = 0;
|
|
rtp->send_digit = 0;
|
|
|
|
return res;
|
|
}
|
|
|
|
static int ast_rtp_dtmf_end(struct ast_rtp_instance *instance, char digit)
|
|
{
|
|
return ast_rtp_dtmf_end_with_duration(instance, digit, 0);
|
|
}
|
|
|
|
static void ast_rtp_update_source(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
/* We simply set this bit so that the next packet sent will have the marker bit turned on */
|
|
ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
|
|
ast_debug(3, "Setting the marker bit due to a source update\n");
|
|
|
|
return;
|
|
}
|
|
|
|
static void ast_rtp_change_source(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance);
|
|
unsigned int ssrc = ast_random();
|
|
|
|
if (!rtp->lastts) {
|
|
ast_debug(3, "Not changing SSRC since we haven't sent any RTP yet\n");
|
|
return;
|
|
}
|
|
|
|
/* We simply set this bit so that the next packet sent will have the marker bit turned on */
|
|
ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
|
|
|
|
ast_debug(3, "Changing ssrc from %u to %u due to a source change\n", rtp->ssrc, ssrc);
|
|
|
|
if (srtp) {
|
|
ast_debug(3, "Changing ssrc for SRTP from %u to %u\n", rtp->ssrc, ssrc);
|
|
res_srtp->change_source(srtp, rtp->ssrc, ssrc);
|
|
}
|
|
|
|
rtp->ssrc = ssrc;
|
|
|
|
return;
|
|
}
|
|
|
|
static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw)
|
|
{
|
|
unsigned int sec, usec, frac;
|
|
sec = tv.tv_sec + 2208988800u; /* Sec between 1900 and 1970 */
|
|
usec = tv.tv_usec;
|
|
frac = (usec << 12) + (usec << 8) - ((usec * 3650) >> 6);
|
|
*msw = sec;
|
|
*lsw = frac;
|
|
}
|
|
|
|
static void ntp2timeval(unsigned int msw, unsigned int lsw, struct timeval *tv)
|
|
{
|
|
tv->tv_sec = msw - 2208988800u;
|
|
tv->tv_usec = ((lsw << 6) / 3650) - (lsw >> 12) - (lsw >> 8);
|
|
}
|
|
|
|
static void calculate_lost_packet_statistics(struct ast_rtp *rtp,
|
|
unsigned int *lost_packets,
|
|
int *fraction_lost)
|
|
{
|
|
unsigned int extended_seq_no;
|
|
unsigned int expected_packets;
|
|
unsigned int expected_interval;
|
|
unsigned int received_interval;
|
|
double rxlost_current;
|
|
int lost_interval;
|
|
|
|
/* Compute statistics */
|
|
extended_seq_no = rtp->cycles + rtp->lastrxseqno;
|
|
expected_packets = extended_seq_no - rtp->seedrxseqno + 1;
|
|
if (rtp->rxcount > expected_packets) {
|
|
expected_packets += rtp->rxcount - expected_packets;
|
|
}
|
|
*lost_packets = expected_packets - rtp->rxcount;
|
|
expected_interval = expected_packets - rtp->rtcp->expected_prior;
|
|
received_interval = rtp->rxcount - rtp->rtcp->received_prior;
|
|
lost_interval = expected_interval - received_interval;
|
|
if (expected_interval == 0 || lost_interval <= 0) {
|
|
*fraction_lost = 0;
|
|
} else {
|
|
*fraction_lost = (lost_interval << 8) / expected_interval;
|
|
}
|
|
|
|
/* Update RTCP statistics */
|
|
rtp->rtcp->received_prior = rtp->rxcount;
|
|
rtp->rtcp->expected_prior = expected_packets;
|
|
if (lost_interval <= 0) {
|
|
rtp->rtcp->rxlost = 0;
|
|
} else {
|
|
rtp->rtcp->rxlost = lost_interval;
|
|
}
|
|
if (rtp->rtcp->rxlost_count == 0) {
|
|
rtp->rtcp->minrxlost = rtp->rtcp->rxlost;
|
|
}
|
|
if (lost_interval < rtp->rtcp->minrxlost) {
|
|
rtp->rtcp->minrxlost = rtp->rtcp->rxlost;
|
|
}
|
|
if (lost_interval > rtp->rtcp->maxrxlost) {
|
|
rtp->rtcp->maxrxlost = rtp->rtcp->rxlost;
|
|
}
|
|
rxlost_current = normdev_compute(rtp->rtcp->normdev_rxlost,
|
|
rtp->rtcp->rxlost,
|
|
rtp->rtcp->rxlost_count);
|
|
rtp->rtcp->stdev_rxlost = stddev_compute(rtp->rtcp->stdev_rxlost,
|
|
rtp->rtcp->rxlost,
|
|
rtp->rtcp->normdev_rxlost,
|
|
rxlost_current,
|
|
rtp->rtcp->rxlost_count);
|
|
rtp->rtcp->normdev_rxlost = rxlost_current;
|
|
rtp->rtcp->rxlost_count++;
|
|
}
|
|
|
|
/*! \brief Send RTCP SR or RR report */
|
|
static int ast_rtcp_write_report(struct ast_rtp_instance *instance, int sr)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
RAII_VAR(struct ast_json *, message_blob, NULL, ast_json_unref);
|
|
int res;
|
|
int len = 0;
|
|
struct timeval now;
|
|
unsigned int now_lsw;
|
|
unsigned int now_msw;
|
|
unsigned int *rtcpheader;
|
|
unsigned int lost_packets;
|
|
int fraction_lost;
|
|
struct timeval dlsr = { 0, };
|
|
char bdata[512];
|
|
int rate = rtp_get_rate(rtp->f.subclass.format);
|
|
int ice;
|
|
int header_offset = 0;
|
|
char *str_remote_address;
|
|
char *str_local_address;
|
|
struct ast_sockaddr remote_address = { { 0, } };
|
|
struct ast_sockaddr local_address = { { 0, } };
|
|
struct ast_sockaddr real_remote_address = { { 0, } };
|
|
struct ast_sockaddr real_local_address = { { 0, } };
|
|
struct ast_rtp_rtcp_report_block *report_block = NULL;
|
|
RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report,
|
|
ast_rtp_rtcp_report_alloc(rtp->themssrc ? 1 : 0),
|
|
ao2_cleanup);
|
|
|
|
if (!rtp || !rtp->rtcp) {
|
|
return 0;
|
|
}
|
|
|
|
if (ast_sockaddr_isnull(&rtp->rtcp->them)) { /* This'll stop rtcp for this rtp session */
|
|
/* RTCP was stopped. */
|
|
return 0;
|
|
}
|
|
|
|
if (!rtcp_report) {
|
|
return 1;
|
|
}
|
|
|
|
/* Compute statistics */
|
|
calculate_lost_packet_statistics(rtp, &lost_packets, &fraction_lost);
|
|
|
|
gettimeofday(&now, NULL);
|
|
rtcp_report->reception_report_count = rtp->themssrc ? 1 : 0;
|
|
rtcp_report->ssrc = rtp->ssrc;
|
|
rtcp_report->type = sr ? RTCP_PT_SR : RTCP_PT_RR;
|
|
if (sr) {
|
|
rtcp_report->sender_information.ntp_timestamp = now;
|
|
rtcp_report->sender_information.rtp_timestamp = rtp->lastts;
|
|
rtcp_report->sender_information.packet_count = rtp->txcount;
|
|
rtcp_report->sender_information.octet_count = rtp->txoctetcount;
|
|
}
|
|
|
|
if (rtp->themssrc) {
|
|
report_block = ast_calloc(1, sizeof(*report_block));
|
|
if (!report_block) {
|
|
return 1;
|
|
}
|
|
|
|
rtcp_report->report_block[0] = report_block;
|
|
report_block->source_ssrc = rtp->themssrc;
|
|
report_block->lost_count.fraction = (fraction_lost & 0xff);
|
|
report_block->lost_count.packets = (lost_packets & 0xffffff);
|
|
report_block->highest_seq_no = (rtp->cycles | (rtp->lastrxseqno & 0xffff));
|
|
report_block->ia_jitter = (unsigned int)(rtp->rxjitter * rate);
|
|
report_block->lsr = rtp->rtcp->themrxlsr;
|
|
/* If we haven't received an SR report, DLSR should be 0 */
|
|
if (!ast_tvzero(rtp->rtcp->rxlsr)) {
|
|
timersub(&now, &rtp->rtcp->rxlsr, &dlsr);
|
|
report_block->dlsr = (((dlsr.tv_sec * 1000) + (dlsr.tv_usec / 1000)) * 65536) / 1000;
|
|
}
|
|
}
|
|
timeval2ntp(rtcp_report->sender_information.ntp_timestamp, &now_msw, &now_lsw);
|
|
rtcpheader = (unsigned int *)bdata;
|
|
rtcpheader[1] = htonl(rtcp_report->ssrc); /* Our SSRC */
|
|
len += 8;
|
|
if (sr) {
|
|
header_offset = 5;
|
|
rtcpheader[2] = htonl(now_msw); /* now, MSW. gettimeofday() + SEC_BETWEEN_1900_AND_1970*/
|
|
rtcpheader[3] = htonl(now_lsw); /* now, LSW */
|
|
rtcpheader[4] = htonl(rtcp_report->sender_information.rtp_timestamp);
|
|
rtcpheader[5] = htonl(rtcp_report->sender_information.packet_count);
|
|
rtcpheader[6] = htonl(rtcp_report->sender_information.octet_count);
|
|
len += 20;
|
|
}
|
|
if (report_block) {
|
|
rtcpheader[2 + header_offset] = htonl(report_block->source_ssrc); /* Their SSRC */
|
|
rtcpheader[3 + header_offset] = htonl((report_block->lost_count.fraction << 24) | report_block->lost_count.packets);
|
|
rtcpheader[4 + header_offset] = htonl(report_block->highest_seq_no);
|
|
rtcpheader[5 + header_offset] = htonl(report_block->ia_jitter);
|
|
rtcpheader[6 + header_offset] = htonl(report_block->lsr);
|
|
rtcpheader[7 + header_offset] = htonl(report_block->dlsr);
|
|
len += 24;
|
|
}
|
|
rtcpheader[0] = htonl((2 << 30) | (rtcp_report->reception_report_count << 24)
|
|
| ((sr ? RTCP_PT_SR : RTCP_PT_RR) << 16) | ((len/4)-1));
|
|
|
|
/* Insert SDES here. Probably should make SDES text equal to mimetypes[code].type (not subtype 'cos */
|
|
/* it can change mid call, and SDES can't) */
|
|
rtcpheader[len/4] = htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | 2);
|
|
rtcpheader[(len/4)+1] = htonl(rtcp_report->ssrc);
|
|
rtcpheader[(len/4)+2] = htonl(0x01 << 24);
|
|
len += 12;
|
|
|
|
ast_sockaddr_copy(&remote_address, &rtp->rtcp->them);
|
|
res = rtcp_sendto(instance, (unsigned int *)rtcpheader, len, 0, &remote_address, &ice);
|
|
if (res < 0) {
|
|
ast_log(LOG_ERROR, "RTCP %s transmission error to %s, rtcp halted %s\n",
|
|
sr ? "SR" : "RR",
|
|
ast_sockaddr_stringify(&rtp->rtcp->them),
|
|
strerror(errno));
|
|
return 0;
|
|
}
|
|
|
|
/* Update RTCP SR/RR statistics */
|
|
if (sr) {
|
|
rtp->rtcp->txlsr = rtcp_report->sender_information.ntp_timestamp;
|
|
rtp->rtcp->sr_count++;
|
|
rtp->rtcp->lastsrtxcount = rtp->txcount;
|
|
} else {
|
|
rtp->rtcp->rr_count++;
|
|
}
|
|
|
|
if (rtcp_debug_test_addr(&rtp->rtcp->them)) {
|
|
ast_verbose("* Sent RTCP %s to %s%s\n", sr ? "SR" : "RR",
|
|
ast_sockaddr_stringify(&remote_address), ice ? " (via ICE)" : "");
|
|
ast_verbose(" Our SSRC: %u\n", rtcp_report->ssrc);
|
|
if (sr) {
|
|
ast_verbose(" Sent(NTP): %u.%010u\n",
|
|
(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_sec,
|
|
(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec * 4096);
|
|
ast_verbose(" Sent(RTP): %u\n", rtcp_report->sender_information.rtp_timestamp);
|
|
ast_verbose(" Sent packets: %u\n", rtcp_report->sender_information.packet_count);
|
|
ast_verbose(" Sent octets: %u\n", rtcp_report->sender_information.octet_count);
|
|
}
|
|
if (report_block) {
|
|
ast_verbose(" Report block:\n");
|
|
ast_verbose(" Their SSRC: %u\n", report_block->source_ssrc);
|
|
ast_verbose(" Fraction lost: %d\n", report_block->lost_count.fraction);
|
|
ast_verbose(" Cumulative loss: %u\n", report_block->lost_count.packets);
|
|
ast_verbose(" Highest seq no: %u\n", report_block->highest_seq_no);
|
|
ast_verbose(" IA jitter: %.4f\n", (double)report_block->ia_jitter / rate);
|
|
ast_verbose(" Their last SR: %u\n", report_block->lsr);
|
|
ast_verbose(" DLSR: %4.4f (sec)\n\n", (double)(report_block->dlsr / 65536.0));
|
|
}
|
|
}
|
|
|
|
ast_rtp_instance_get_local_address(instance, &local_address);
|
|
if (!ast_find_ourip(&real_local_address, &local_address, 0)) {
|
|
str_local_address = ast_strdupa(ast_sockaddr_stringify(&real_local_address));
|
|
} else {
|
|
str_local_address = ast_strdupa(ast_sockaddr_stringify(&local_address));
|
|
}
|
|
|
|
if (!ast_find_ourip(&real_remote_address, &remote_address, 0)) {
|
|
str_remote_address = ast_strdupa(ast_sockaddr_stringify(&real_remote_address));
|
|
} else {
|
|
str_remote_address = ast_strdupa(ast_sockaddr_stringify(&remote_address));
|
|
}
|
|
|
|
message_blob = ast_json_pack("{s: s, s: s}",
|
|
"to", str_remote_address,
|
|
"from", str_local_address);
|
|
ast_rtp_publish_rtcp_message(instance, ast_rtp_rtcp_sent_type(),
|
|
rtcp_report,
|
|
message_blob);
|
|
return res;
|
|
}
|
|
|
|
/*! \brief Write and RTCP packet to the far end
|
|
* \note Decide if we are going to send an SR (with Reception Block) or RR
|
|
* RR is sent if we have not sent any rtp packets in the previous interval */
|
|
static int ast_rtcp_write(const void *data)
|
|
{
|
|
struct ast_rtp_instance *instance = (struct ast_rtp_instance *) data;
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
int res;
|
|
|
|
if (!rtp || !rtp->rtcp || rtp->rtcp->schedid == -1) {
|
|
ao2_ref(instance, -1);
|
|
return 0;
|
|
}
|
|
|
|
if (rtp->txcount > rtp->rtcp->lastsrtxcount) {
|
|
/* Send an SR */
|
|
res = ast_rtcp_write_report(instance, 1);
|
|
} else {
|
|
/* Send an RR */
|
|
res = ast_rtcp_write_report(instance, 0);
|
|
}
|
|
|
|
if (!res) {
|
|
/*
|
|
* Not being rescheduled.
|
|
*/
|
|
ao2_ref(instance, -1);
|
|
rtp->rtcp->schedid = -1;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static int ast_rtp_raw_write(struct ast_rtp_instance *instance, struct ast_frame *frame, int codec)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
int pred, mark = 0;
|
|
unsigned int ms = calc_txstamp(rtp, &frame->delivery);
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
int rate = rtp_get_rate(frame->subclass.format) / 1000;
|
|
|
|
if (ast_format_cmp(frame->subclass.format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) {
|
|
frame->samples /= 2;
|
|
}
|
|
|
|
if (rtp->sending_digit) {
|
|
return 0;
|
|
}
|
|
|
|
if (frame->frametype == AST_FRAME_VOICE) {
|
|
pred = rtp->lastts + frame->samples;
|
|
|
|
/* Re-calculate last TS */
|
|
rtp->lastts = rtp->lastts + ms * rate;
|
|
if (ast_tvzero(frame->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((int)rtp->lastts - pred) < MAX_TIMESTAMP_SKEW) {
|
|
rtp->lastts = pred;
|
|
} else {
|
|
ast_debug(3, "Difference is %d, ms is %u\n", abs((int)rtp->lastts - pred), ms);
|
|
mark = 1;
|
|
}
|
|
}
|
|
} else if (frame->frametype == AST_FRAME_VIDEO) {
|
|
mark = frame->subclass.frame_ending;
|
|
pred = rtp->lastovidtimestamp + frame->samples;
|
|
/* Re-calculate last TS */
|
|
rtp->lastts = rtp->lastts + ms * 90;
|
|
/* If it's close to our prediction, go for it */
|
|
if (ast_tvzero(frame->delivery)) {
|
|
if (abs((int)rtp->lastts - pred) < 7200) {
|
|
rtp->lastts = pred;
|
|
rtp->lastovidtimestamp += frame->samples;
|
|
} else {
|
|
ast_debug(3, "Difference is %d, ms is %u (%u), pred/ts/samples %u/%d/%d\n", abs((int)rtp->lastts - pred), ms, ms * 90, rtp->lastts, pred, frame->samples);
|
|
rtp->lastovidtimestamp = rtp->lastts;
|
|
}
|
|
}
|
|
} else {
|
|
pred = rtp->lastotexttimestamp + frame->samples;
|
|
/* Re-calculate last TS */
|
|
rtp->lastts = rtp->lastts + ms;
|
|
/* If it's close to our prediction, go for it */
|
|
if (ast_tvzero(frame->delivery)) {
|
|
if (abs((int)rtp->lastts - pred) < 7200) {
|
|
rtp->lastts = pred;
|
|
rtp->lastotexttimestamp += frame->samples;
|
|
} else {
|
|
ast_debug(3, "Difference is %d, ms is %u, pred/ts/samples %u/%d/%d\n", abs((int)rtp->lastts - pred), ms, rtp->lastts, pred, frame->samples);
|
|
rtp->lastotexttimestamp = rtp->lastts;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* If we have been explicitly told to set the marker bit then do so */
|
|
if (ast_test_flag(rtp, FLAG_NEED_MARKER_BIT)) {
|
|
mark = 1;
|
|
ast_clear_flag(rtp, FLAG_NEED_MARKER_BIT);
|
|
}
|
|
|
|
/* If the timestamp for non-digt packets has moved beyond the timestamp for digits, update the digit timestamp */
|
|
if (rtp->lastts > rtp->lastdigitts) {
|
|
rtp->lastdigitts = rtp->lastts;
|
|
}
|
|
|
|
if (ast_test_flag(frame, AST_FRFLAG_HAS_TIMING_INFO)) {
|
|
rtp->lastts = frame->ts * rate;
|
|
}
|
|
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
|
|
/* If we know the remote address construct a packet and send it out */
|
|
if (!ast_sockaddr_isnull(&remote_address)) {
|
|
int hdrlen = 12, res, ice;
|
|
unsigned char *rtpheader = (unsigned char *)(frame->data.ptr - 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 ((res = rtp_sendto(instance, (void *)rtpheader, frame->datalen + hdrlen, 0, &remote_address, &ice)) < 0) {
|
|
if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT) || (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT) && (ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
|
|
ast_debug(1, "RTP Transmission error of packet %d to %s: %s\n",
|
|
rtp->seqno,
|
|
ast_sockaddr_stringify(&remote_address),
|
|
strerror(errno));
|
|
} else if (((ast_test_flag(rtp, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(rtp, FLAG_NAT_INACTIVE_NOWARN)) {
|
|
/* Only give this error message once if we are not RTP debugging */
|
|
if (rtpdebug)
|
|
ast_debug(0, "RTP NAT: Can't write RTP to private address %s, waiting for other end to send audio...\n",
|
|
ast_sockaddr_stringify(&remote_address));
|
|
ast_set_flag(rtp, FLAG_NAT_INACTIVE_NOWARN);
|
|
}
|
|
} else {
|
|
rtp->txcount++;
|
|
rtp->txoctetcount += (res - hdrlen);
|
|
|
|
if (rtp->rtcp && rtp->rtcp->schedid < 1) {
|
|
ast_debug(1, "Starting RTCP transmission on RTP instance '%p'\n", instance);
|
|
ao2_ref(instance, +1);
|
|
rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, instance);
|
|
if (rtp->rtcp->schedid < 0) {
|
|
ao2_ref(instance, -1);
|
|
ast_log(LOG_WARNING, "scheduling RTCP transmission failed.\n");
|
|
}
|
|
}
|
|
}
|
|
|
|
if (rtp_debug_test_addr(&remote_address)) {
|
|
ast_verbose("Sent RTP packet to %s%s (type %-2.2d, seq %-6.6d, ts %-6.6u, len %-6.6d)\n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
ice ? " (via ICE)" : "",
|
|
codec, rtp->seqno, rtp->lastts, res - hdrlen);
|
|
}
|
|
}
|
|
|
|
rtp->seqno++;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static struct ast_frame *red_t140_to_red(struct rtp_red *red) {
|
|
unsigned char *data = red->t140red.data.ptr;
|
|
int len = 0;
|
|
int i;
|
|
|
|
/* replace most aged generation */
|
|
if (red->len[0]) {
|
|
for (i = 1; i < red->num_gen+1; i++)
|
|
len += red->len[i];
|
|
|
|
memmove(&data[red->hdrlen], &data[red->hdrlen+red->len[0]], len);
|
|
}
|
|
|
|
/* Store length of each generation and primary data length*/
|
|
for (i = 0; i < red->num_gen; i++)
|
|
red->len[i] = red->len[i+1];
|
|
red->len[i] = red->t140.datalen;
|
|
|
|
/* write each generation length in red header */
|
|
len = red->hdrlen;
|
|
for (i = 0; i < red->num_gen; i++) {
|
|
len += data[i*4+3] = red->len[i];
|
|
}
|
|
|
|
/* add primary data to buffer */
|
|
memcpy(&data[len], red->t140.data.ptr, red->t140.datalen);
|
|
red->t140red.datalen = len + red->t140.datalen;
|
|
|
|
/* no primary data and no generations to send */
|
|
if (len == red->hdrlen && !red->t140.datalen) {
|
|
return NULL;
|
|
}
|
|
|
|
/* reset t.140 buffer */
|
|
red->t140.datalen = 0;
|
|
|
|
return &red->t140red;
|
|
}
|
|
|
|
static int ast_rtp_write(struct ast_rtp_instance *instance, struct ast_frame *frame)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
struct ast_format *format;
|
|
int codec;
|
|
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
|
|
/* If we don't actually know the remote address don't even bother doing anything */
|
|
if (ast_sockaddr_isnull(&remote_address)) {
|
|
ast_debug(1, "No remote address on RTP instance '%p' so dropping frame\n", instance);
|
|
return 0;
|
|
}
|
|
|
|
/* VP8: is this a request to send a RTCP FIR? */
|
|
if (frame->frametype == AST_FRAME_CONTROL && frame->subclass.integer == AST_CONTROL_VIDUPDATE) {
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
unsigned int *rtcpheader;
|
|
char bdata[1024];
|
|
int len = 20;
|
|
int ice;
|
|
int res;
|
|
|
|
if (!rtp || !rtp->rtcp) {
|
|
return 0;
|
|
}
|
|
|
|
if (ast_sockaddr_isnull(&rtp->rtcp->them)) {
|
|
/*
|
|
* RTCP was stopped.
|
|
*/
|
|
return 0;
|
|
}
|
|
|
|
/* Prepare RTCP FIR (PT=206, FMT=4) */
|
|
rtp->rtcp->firseq++;
|
|
if(rtp->rtcp->firseq == 256) {
|
|
rtp->rtcp->firseq = 0;
|
|
}
|
|
|
|
rtcpheader = (unsigned int *)bdata;
|
|
rtcpheader[0] = htonl((2 << 30) | (4 << 24) | (RTCP_PT_PSFB << 16) | ((len/4)-1));
|
|
rtcpheader[1] = htonl(rtp->ssrc);
|
|
rtcpheader[2] = htonl(rtp->themssrc);
|
|
rtcpheader[3] = htonl(rtp->themssrc); /* FCI: SSRC */
|
|
rtcpheader[4] = htonl(rtp->rtcp->firseq << 24); /* FCI: Sequence number */
|
|
res = rtcp_sendto(instance, (unsigned int *)rtcpheader, len, 0, &rtp->rtcp->them, &ice);
|
|
if (res < 0) {
|
|
ast_log(LOG_ERROR, "RTCP FIR transmission error: %s\n", strerror(errno));
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* If there is no data length we can't very well send the packet */
|
|
if (!frame->datalen) {
|
|
ast_debug(1, "Received frame with no data for RTP instance '%p' so dropping frame\n", instance);
|
|
return 0;
|
|
}
|
|
|
|
/* If the packet is not one our RTP stack supports bail out */
|
|
if (frame->frametype != AST_FRAME_VOICE && frame->frametype != AST_FRAME_VIDEO && frame->frametype != AST_FRAME_TEXT) {
|
|
ast_log(LOG_WARNING, "RTP can only send voice, video, and text\n");
|
|
return -1;
|
|
}
|
|
|
|
if (rtp->red) {
|
|
/* return 0; */
|
|
/* no primary data or generations to send */
|
|
if ((frame = red_t140_to_red(rtp->red)) == NULL)
|
|
return 0;
|
|
}
|
|
|
|
/* Grab the subclass and look up the payload we are going to use */
|
|
codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance),
|
|
1,
|
|
frame->subclass.format,
|
|
0);
|
|
if (codec < 0) {
|
|
ast_log(LOG_WARNING, "Don't know how to send format %s packets with RTP\n",
|
|
ast_format_get_name(frame->subclass.format));
|
|
return -1;
|
|
}
|
|
|
|
/* Note that we do not increase the ref count here as this pointer
|
|
* will not be held by any thing explicitly. The format variable is
|
|
* merely a convenience reference to frame->subclass.format */
|
|
format = frame->subclass.format;
|
|
if (ast_format_cmp(rtp->lasttxformat, format) == AST_FORMAT_CMP_NOT_EQUAL) {
|
|
/* Oh dear, if the format changed we will have to set up a new smoother */
|
|
if (option_debug > 0) {
|
|
ast_debug(1, "Ooh, format changed from %s to %s\n",
|
|
ast_format_get_name(rtp->lasttxformat),
|
|
ast_format_get_name(frame->subclass.format));
|
|
}
|
|
ao2_replace(rtp->lasttxformat, format);
|
|
if (rtp->smoother) {
|
|
ast_smoother_free(rtp->smoother);
|
|
rtp->smoother = NULL;
|
|
}
|
|
}
|
|
|
|
/* If no smoother is present see if we have to set one up */
|
|
if (!rtp->smoother && ast_format_can_be_smoothed(format)) {
|
|
unsigned int framing_ms = ast_rtp_codecs_get_framing(ast_rtp_instance_get_codecs(instance));
|
|
|
|
if (framing_ms) {
|
|
rtp->smoother = ast_smoother_new((framing_ms * ast_format_get_minimum_bytes(format)) / ast_format_get_minimum_ms(format));
|
|
if (!rtp->smoother) {
|
|
ast_log(LOG_WARNING, "Unable to create smoother: format %s ms: %u len: %u\n",
|
|
ast_format_get_name(format), framing_ms, ast_format_get_minimum_bytes(format));
|
|
return -1;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* Feed audio frames into the actual function that will create a frame and send it */
|
|
if (rtp->smoother) {
|
|
struct ast_frame *f;
|
|
|
|
if (ast_smoother_test_flag(rtp->smoother, AST_SMOOTHER_FLAG_BE)) {
|
|
ast_smoother_feed_be(rtp->smoother, frame);
|
|
} else {
|
|
ast_smoother_feed(rtp->smoother, frame);
|
|
}
|
|
|
|
while ((f = ast_smoother_read(rtp->smoother)) && (f->data.ptr)) {
|
|
ast_rtp_raw_write(instance, f, codec);
|
|
}
|
|
} else {
|
|
int hdrlen = 12;
|
|
struct ast_frame *f = NULL;
|
|
|
|
if (frame->offset < hdrlen) {
|
|
f = ast_frdup(frame);
|
|
} else {
|
|
f = frame;
|
|
}
|
|
if (f->data.ptr) {
|
|
ast_rtp_raw_write(instance, f, codec);
|
|
}
|
|
if (f != frame) {
|
|
ast_frfree(f);
|
|
}
|
|
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static void calc_rxstamp(struct timeval *tv, struct ast_rtp *rtp, unsigned int timestamp, int mark)
|
|
{
|
|
struct timeval now;
|
|
struct timeval tmp;
|
|
double transit;
|
|
double current_time;
|
|
double d;
|
|
double dtv;
|
|
double prog;
|
|
int rate = rtp_get_rate(rtp->f.subclass.format);
|
|
|
|
double normdev_rxjitter_current;
|
|
if ((!rtp->rxcore.tv_sec && !rtp->rxcore.tv_usec) || mark) {
|
|
gettimeofday(&rtp->rxcore, NULL);
|
|
rtp->drxcore = (double) rtp->rxcore.tv_sec + (double) rtp->rxcore.tv_usec / 1000000;
|
|
/* map timestamp to a real time */
|
|
rtp->seedrxts = timestamp; /* Their RTP timestamp started with this */
|
|
tmp = ast_samp2tv(timestamp, rate);
|
|
rtp->rxcore = ast_tvsub(rtp->rxcore, tmp);
|
|
/* Round to 0.1ms for nice, pretty timestamps */
|
|
rtp->rxcore.tv_usec -= rtp->rxcore.tv_usec % 100;
|
|
}
|
|
|
|
gettimeofday(&now,NULL);
|
|
/* rxcore is the mapping between the RTP timestamp and _our_ real time from gettimeofday() */
|
|
tmp = ast_samp2tv(timestamp, rate);
|
|
*tv = ast_tvadd(rtp->rxcore, tmp);
|
|
|
|
prog = (double)((timestamp-rtp->seedrxts)/(float)(rate));
|
|
dtv = (double)rtp->drxcore + (double)(prog);
|
|
current_time = (double)now.tv_sec + (double)now.tv_usec/1000000;
|
|
transit = current_time - dtv;
|
|
d = transit - rtp->rxtransit;
|
|
rtp->rxtransit = transit;
|
|
if (d<0) {
|
|
d=-d;
|
|
}
|
|
rtp->rxjitter += (1./16.) * (d - rtp->rxjitter);
|
|
if (rtp->rtcp) {
|
|
if (rtp->rxjitter > rtp->rtcp->maxrxjitter)
|
|
rtp->rtcp->maxrxjitter = rtp->rxjitter;
|
|
if (rtp->rtcp->rxjitter_count == 1)
|
|
rtp->rtcp->minrxjitter = rtp->rxjitter;
|
|
if (rtp->rtcp && rtp->rxjitter < rtp->rtcp->minrxjitter)
|
|
rtp->rtcp->minrxjitter = rtp->rxjitter;
|
|
|
|
normdev_rxjitter_current = normdev_compute(rtp->rtcp->normdev_rxjitter,rtp->rxjitter,rtp->rtcp->rxjitter_count);
|
|
rtp->rtcp->stdev_rxjitter = stddev_compute(rtp->rtcp->stdev_rxjitter,rtp->rxjitter,rtp->rtcp->normdev_rxjitter,normdev_rxjitter_current,rtp->rtcp->rxjitter_count);
|
|
|
|
rtp->rtcp->normdev_rxjitter = normdev_rxjitter_current;
|
|
rtp->rtcp->rxjitter_count++;
|
|
}
|
|
}
|
|
|
|
static struct ast_frame *create_dtmf_frame(struct ast_rtp_instance *instance, enum ast_frame_type type, int compensate)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
|
|
if (((compensate && type == AST_FRAME_DTMF_END) || (type == AST_FRAME_DTMF_BEGIN)) && ast_tvcmp(ast_tvnow(), rtp->dtmfmute) < 0) {
|
|
ast_debug(1, "Ignore potential DTMF echo from '%s'\n",
|
|
ast_sockaddr_stringify(&remote_address));
|
|
rtp->resp = 0;
|
|
rtp->dtmfsamples = 0;
|
|
return &ast_null_frame;
|
|
}
|
|
ast_debug(1, "Creating %s DTMF Frame: %d (%c), at %s\n",
|
|
type == AST_FRAME_DTMF_END ? "END" : "BEGIN",
|
|
rtp->resp, rtp->resp,
|
|
ast_sockaddr_stringify(&remote_address));
|
|
if (rtp->resp == 'X') {
|
|
rtp->f.frametype = AST_FRAME_CONTROL;
|
|
rtp->f.subclass.integer = AST_CONTROL_FLASH;
|
|
} else {
|
|
rtp->f.frametype = type;
|
|
rtp->f.subclass.integer = rtp->resp;
|
|
}
|
|
rtp->f.datalen = 0;
|
|
rtp->f.samples = 0;
|
|
rtp->f.mallocd = 0;
|
|
rtp->f.src = "RTP";
|
|
AST_LIST_NEXT(&rtp->f, frame_list) = NULL;
|
|
|
|
return &rtp->f;
|
|
}
|
|
|
|
static void process_dtmf_rfc2833(struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, struct ast_sockaddr *addr, int payloadtype, int mark, struct frame_list *frames)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
unsigned int event, event_end, samples;
|
|
char resp = 0;
|
|
struct ast_frame *f = NULL;
|
|
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
|
|
/* Figure out event, event end, and samples */
|
|
event = ntohl(*((unsigned int *)(data)));
|
|
event >>= 24;
|
|
event_end = ntohl(*((unsigned int *)(data)));
|
|
event_end <<= 8;
|
|
event_end >>= 24;
|
|
samples = ntohl(*((unsigned int *)(data)));
|
|
samples &= 0xFFFF;
|
|
|
|
if (rtp_debug_test_addr(&remote_address)) {
|
|
ast_verbose("Got RTP RFC2833 from %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6d, mark %d, event %08x, end %d, duration %-5.5u) \n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
payloadtype, seqno, timestamp, len, (mark?1:0), event, ((event_end & 0x80)?1:0), samples);
|
|
}
|
|
|
|
/* Print out debug if turned on */
|
|
if (rtpdebug)
|
|
ast_debug(0, "- RTP 2833 Event: %08x (len = %d)\n", event, len);
|
|
|
|
/* Figure out what digit was pressed */
|
|
if (event < 10) {
|
|
resp = '0' + event;
|
|
} else if (event < 11) {
|
|
resp = '*';
|
|
} else if (event < 12) {
|
|
resp = '#';
|
|
} else if (event < 16) {
|
|
resp = 'A' + (event - 12);
|
|
} else if (event < 17) { /* Event 16: Hook flash */
|
|
resp = 'X';
|
|
} else {
|
|
/* Not a supported event */
|
|
ast_debug(1, "Ignoring RTP 2833 Event: %08x. Not a DTMF Digit.\n", event);
|
|
return;
|
|
}
|
|
|
|
if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE)) {
|
|
if ((rtp->last_end_timestamp != timestamp) || (rtp->resp && rtp->resp != resp)) {
|
|
rtp->resp = resp;
|
|
rtp->dtmf_timeout = 0;
|
|
f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE)));
|
|
f->len = 0;
|
|
rtp->last_end_timestamp = timestamp;
|
|
AST_LIST_INSERT_TAIL(frames, f, frame_list);
|
|
}
|
|
} else {
|
|
/* The duration parameter measures the complete
|
|
duration of the event (from the beginning) - RFC2833.
|
|
Account for the fact that duration is only 16 bits long
|
|
(about 8 seconds at 8000 Hz) and can wrap is digit
|
|
is hold for too long. */
|
|
unsigned int new_duration = rtp->dtmf_duration;
|
|
unsigned int last_duration = new_duration & 0xFFFF;
|
|
|
|
if (last_duration > 64000 && samples < last_duration) {
|
|
new_duration += 0xFFFF + 1;
|
|
}
|
|
new_duration = (new_duration & ~0xFFFF) | samples;
|
|
|
|
if (event_end & 0x80) {
|
|
/* End event */
|
|
if ((rtp->last_seqno != seqno) && (timestamp > rtp->last_end_timestamp)) {
|
|
rtp->last_end_timestamp = timestamp;
|
|
rtp->dtmf_duration = new_duration;
|
|
rtp->resp = resp;
|
|
f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0));
|
|
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
|
|
rtp->resp = 0;
|
|
rtp->dtmf_duration = rtp->dtmf_timeout = 0;
|
|
AST_LIST_INSERT_TAIL(frames, f, frame_list);
|
|
} else if (rtpdebug) {
|
|
ast_debug(1, "Dropping duplicate or out of order DTMF END frame (seqno: %u, ts %u, digit %c)\n",
|
|
seqno, timestamp, resp);
|
|
}
|
|
} else {
|
|
/* Begin/continuation */
|
|
|
|
/* The second portion of the seqno check is to not mistakenly
|
|
* stop accepting DTMF if the seqno rolls over beyond
|
|
* 65535.
|
|
*/
|
|
if ((rtp->last_seqno > seqno && rtp->last_seqno - seqno < 50)
|
|
|| timestamp <= rtp->last_end_timestamp) {
|
|
/* Out of order frame. Processing this can cause us to
|
|
* improperly duplicate incoming DTMF, so just drop
|
|
* this.
|
|
*/
|
|
if (rtpdebug) {
|
|
ast_debug(1, "Dropping out of order DTMF frame (seqno %u, ts %u, digit %c)\n",
|
|
seqno, timestamp, resp);
|
|
}
|
|
return;
|
|
}
|
|
|
|
if (rtp->resp && rtp->resp != resp) {
|
|
/* Another digit already began. End it */
|
|
f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0));
|
|
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
|
|
rtp->resp = 0;
|
|
rtp->dtmf_duration = rtp->dtmf_timeout = 0;
|
|
AST_LIST_INSERT_TAIL(frames, f, frame_list);
|
|
}
|
|
|
|
if (rtp->resp) {
|
|
/* Digit continues */
|
|
rtp->dtmf_duration = new_duration;
|
|
} else {
|
|
/* New digit began */
|
|
rtp->resp = resp;
|
|
f = ast_frdup(create_dtmf_frame(instance, AST_FRAME_DTMF_BEGIN, 0));
|
|
rtp->dtmf_duration = samples;
|
|
AST_LIST_INSERT_TAIL(frames, f, frame_list);
|
|
}
|
|
|
|
rtp->dtmf_timeout = timestamp + rtp->dtmf_duration + dtmftimeout;
|
|
}
|
|
|
|
rtp->last_seqno = seqno;
|
|
}
|
|
|
|
rtp->dtmfsamples = samples;
|
|
|
|
return;
|
|
}
|
|
|
|
static struct ast_frame *process_dtmf_cisco(struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, struct ast_sockaddr *addr, int payloadtype, int mark)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
unsigned int event, flags, power;
|
|
char resp = 0;
|
|
unsigned char seq;
|
|
struct ast_frame *f = NULL;
|
|
|
|
if (len < 4) {
|
|
return NULL;
|
|
}
|
|
|
|
/* The format of Cisco RTP DTMF packet looks like next:
|
|
+0 - sequence number of DTMF RTP packet (begins from 1,
|
|
wrapped to 0)
|
|
+1 - set of flags
|
|
+1 (bit 0) - flaps by different DTMF digits delimited by audio
|
|
or repeated digit without audio???
|
|
+2 (+4,+6,...) - power level? (rises from 0 to 32 at begin of tone
|
|
then falls to 0 at its end)
|
|
+3 (+5,+7,...) - detected DTMF digit (0..9,*,#,A-D,...)
|
|
Repeated DTMF information (bytes 4/5, 6/7) is history shifted right
|
|
by each new packet and thus provides some redudancy.
|
|
|
|
Sample of Cisco RTP DTMF packet is (all data in hex):
|
|
19 07 00 02 12 02 20 02
|
|
showing end of DTMF digit '2'.
|
|
|
|
The packets
|
|
27 07 00 02 0A 02 20 02
|
|
28 06 20 02 00 02 0A 02
|
|
shows begin of new digit '2' with very short pause (20 ms) after
|
|
previous digit '2'. Bit +1.0 flips at begin of new digit.
|
|
|
|
Cisco RTP DTMF packets comes as replacement of audio RTP packets
|
|
so its uses the same sequencing and timestamping rules as replaced
|
|
audio packets. Repeat interval of DTMF packets is 20 ms and not rely
|
|
on audio framing parameters. Marker bit isn't used within stream of
|
|
DTMFs nor audio stream coming immediately after DTMF stream. Timestamps
|
|
are not sequential at borders between DTMF and audio streams,
|
|
*/
|
|
|
|
seq = data[0];
|
|
flags = data[1];
|
|
power = data[2];
|
|
event = data[3] & 0x1f;
|
|
|
|
if (rtpdebug)
|
|
ast_debug(0, "Cisco DTMF Digit: %02x (len=%d, seq=%d, flags=%02x, power=%u, history count=%d)\n", event, len, seq, flags, power, (len - 4) / 2);
|
|
if (event < 10) {
|
|
resp = '0' + event;
|
|
} else if (event < 11) {
|
|
resp = '*';
|
|
} else if (event < 12) {
|
|
resp = '#';
|
|
} else if (event < 16) {
|
|
resp = 'A' + (event - 12);
|
|
} else if (event < 17) {
|
|
resp = 'X';
|
|
}
|
|
if ((!rtp->resp && power) || (rtp->resp && (rtp->resp != resp))) {
|
|
rtp->resp = resp;
|
|
/* Why we should care on DTMF compensation at reception? */
|
|
if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE)) {
|
|
f = create_dtmf_frame(instance, AST_FRAME_DTMF_BEGIN, 0);
|
|
rtp->dtmfsamples = 0;
|
|
}
|
|
} else if ((rtp->resp == resp) && !power) {
|
|
f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE));
|
|
f->samples = rtp->dtmfsamples * (rtp_get_rate(rtp->lastrxformat) / 1000);
|
|
rtp->resp = 0;
|
|
} else if (rtp->resp == resp) {
|
|
rtp->dtmfsamples += 20 * (rtp_get_rate(rtp->lastrxformat) / 1000);
|
|
}
|
|
|
|
rtp->dtmf_timeout = 0;
|
|
|
|
return f;
|
|
}
|
|
|
|
static struct ast_frame *process_cn_rfc3389(struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, struct ast_sockaddr *addr, int payloadtype, int mark)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
/* Convert comfort noise into audio with various codecs. Unfortunately this doesn't
|
|
totally help us out becuase we don't have an engine to keep it going and we are not
|
|
guaranteed to have it every 20ms or anything */
|
|
if (rtpdebug) {
|
|
ast_debug(0, "- RTP 3389 Comfort noise event: Format %s (len = %d)\n",
|
|
ast_format_get_name(rtp->lastrxformat), len);
|
|
}
|
|
|
|
if (ast_test_flag(rtp, FLAG_3389_WARNING)) {
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
|
|
ast_log(LOG_NOTICE, "Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client address: %s\n",
|
|
ast_sockaddr_stringify(&remote_address));
|
|
ast_set_flag(rtp, FLAG_3389_WARNING);
|
|
}
|
|
|
|
/* Must have at least one byte */
|
|
if (!len) {
|
|
return NULL;
|
|
}
|
|
if (len < 24) {
|
|
rtp->f.data.ptr = rtp->rawdata + AST_FRIENDLY_OFFSET;
|
|
rtp->f.datalen = len - 1;
|
|
rtp->f.offset = AST_FRIENDLY_OFFSET;
|
|
memcpy(rtp->f.data.ptr, data + 1, len - 1);
|
|
} else {
|
|
rtp->f.data.ptr = NULL;
|
|
rtp->f.offset = 0;
|
|
rtp->f.datalen = 0;
|
|
}
|
|
rtp->f.frametype = AST_FRAME_CNG;
|
|
rtp->f.subclass.integer = data[0] & 0x7f;
|
|
rtp->f.samples = 0;
|
|
rtp->f.delivery.tv_usec = rtp->f.delivery.tv_sec = 0;
|
|
|
|
return &rtp->f;
|
|
}
|
|
|
|
static int update_rtt_stats(struct ast_rtp *rtp, unsigned int lsr, unsigned int dlsr)
|
|
{
|
|
struct timeval now;
|
|
struct timeval rtt_tv;
|
|
unsigned int msw;
|
|
unsigned int lsw;
|
|
unsigned int rtt_msw;
|
|
unsigned int rtt_lsw;
|
|
unsigned int lsr_a;
|
|
unsigned int rtt;
|
|
double normdevrtt_current;
|
|
|
|
gettimeofday(&now, NULL);
|
|
timeval2ntp(now, &msw, &lsw);
|
|
|
|
lsr_a = ((msw & 0x0000ffff) << 16) | ((lsw & 0xffff0000) >> 16);
|
|
rtt = lsr_a - lsr - dlsr;
|
|
rtt_msw = (rtt & 0xffff0000) >> 16;
|
|
rtt_lsw = (rtt & 0x0000ffff) << 16;
|
|
rtt_tv.tv_sec = rtt_msw;
|
|
rtt_tv.tv_usec = ((rtt_lsw << 6) / 3650) - (rtt_lsw >> 12) - (rtt_lsw >> 8);
|
|
rtp->rtcp->rtt = (double)rtt_tv.tv_sec + ((double)rtt_tv.tv_usec / 1000000);
|
|
if (lsr_a - dlsr < lsr) {
|
|
return 1;
|
|
}
|
|
|
|
rtp->rtcp->accumulated_transit += rtp->rtcp->rtt;
|
|
if (rtp->rtcp->rtt_count == 0 || rtp->rtcp->minrtt > rtp->rtcp->rtt) {
|
|
rtp->rtcp->minrtt = rtp->rtcp->rtt;
|
|
}
|
|
if (rtp->rtcp->maxrtt < rtp->rtcp->rtt) {
|
|
rtp->rtcp->maxrtt = rtp->rtcp->rtt;
|
|
}
|
|
|
|
normdevrtt_current = normdev_compute(rtp->rtcp->normdevrtt,
|
|
rtp->rtcp->rtt,
|
|
rtp->rtcp->rtt_count);
|
|
rtp->rtcp->stdevrtt = stddev_compute(rtp->rtcp->stdevrtt,
|
|
rtp->rtcp->rtt,
|
|
rtp->rtcp->normdevrtt,
|
|
normdevrtt_current,
|
|
rtp->rtcp->rtt_count);
|
|
rtp->rtcp->normdevrtt = normdevrtt_current;
|
|
rtp->rtcp->rtt_count++;
|
|
|
|
return 0;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Update RTCP interarrival jitter stats
|
|
*/
|
|
static void update_jitter_stats(struct ast_rtp *rtp, unsigned int ia_jitter)
|
|
{
|
|
double reported_jitter;
|
|
double reported_normdev_jitter_current;
|
|
|
|
rtp->rtcp->reported_jitter = ia_jitter;
|
|
reported_jitter = (double) rtp->rtcp->reported_jitter;
|
|
if (rtp->rtcp->reported_jitter_count == 0) {
|
|
rtp->rtcp->reported_minjitter = reported_jitter;
|
|
}
|
|
if (reported_jitter < rtp->rtcp->reported_minjitter) {
|
|
rtp->rtcp->reported_minjitter = reported_jitter;
|
|
}
|
|
if (reported_jitter > rtp->rtcp->reported_maxjitter) {
|
|
rtp->rtcp->reported_maxjitter = reported_jitter;
|
|
}
|
|
reported_normdev_jitter_current = normdev_compute(rtp->rtcp->reported_normdev_jitter, reported_jitter, rtp->rtcp->reported_jitter_count);
|
|
rtp->rtcp->reported_stdev_jitter = stddev_compute(rtp->rtcp->reported_stdev_jitter, reported_jitter, rtp->rtcp->reported_normdev_jitter, reported_normdev_jitter_current, rtp->rtcp->reported_jitter_count);
|
|
rtp->rtcp->reported_normdev_jitter = reported_normdev_jitter_current;
|
|
}
|
|
|
|
/*!
|
|
* \internal
|
|
* \brief Update RTCP lost packet stats
|
|
*/
|
|
static void update_lost_stats(struct ast_rtp *rtp, unsigned int lost_packets)
|
|
{
|
|
double reported_lost;
|
|
double reported_normdev_lost_current;
|
|
|
|
rtp->rtcp->reported_lost = lost_packets;
|
|
reported_lost = (double)rtp->rtcp->reported_lost;
|
|
if (rtp->rtcp->reported_jitter_count == 0) {
|
|
rtp->rtcp->reported_minlost = reported_lost;
|
|
}
|
|
if (reported_lost < rtp->rtcp->reported_minlost) {
|
|
rtp->rtcp->reported_minlost = reported_lost;
|
|
}
|
|
if (reported_lost > rtp->rtcp->reported_maxlost) {
|
|
rtp->rtcp->reported_maxlost = reported_lost;
|
|
}
|
|
reported_normdev_lost_current = normdev_compute(rtp->rtcp->reported_normdev_lost, reported_lost, rtp->rtcp->reported_jitter_count);
|
|
rtp->rtcp->reported_stdev_lost = stddev_compute(rtp->rtcp->reported_stdev_lost, reported_lost, rtp->rtcp->reported_normdev_lost, reported_normdev_lost_current, rtp->rtcp->reported_jitter_count);
|
|
rtp->rtcp->reported_normdev_lost = reported_normdev_lost_current;
|
|
}
|
|
|
|
static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_sockaddr addr;
|
|
unsigned char rtcpdata[8192 + AST_FRIENDLY_OFFSET];
|
|
unsigned int *rtcpheader = (unsigned int *)(rtcpdata + AST_FRIENDLY_OFFSET);
|
|
int res, packetwords, position = 0;
|
|
int report_counter = 0;
|
|
struct ast_rtp_rtcp_report_block *report_block;
|
|
struct ast_frame *f = &ast_null_frame;
|
|
char *str_local_address;
|
|
char *str_remote_address;
|
|
struct ast_sockaddr local_address = { { 0,} };
|
|
struct ast_sockaddr real_local_address = { { 0, } };
|
|
struct ast_sockaddr real_remote_address = { { 0, } };
|
|
|
|
/* Read in RTCP data from the socket */
|
|
if ((res = rtcp_recvfrom(instance, rtcpdata + AST_FRIENDLY_OFFSET,
|
|
sizeof(rtcpdata) - AST_FRIENDLY_OFFSET,
|
|
0, &addr)) < 0) {
|
|
ast_assert(errno != EBADF);
|
|
if (errno != EAGAIN) {
|
|
ast_log(LOG_WARNING, "RTCP Read error: %s. Hanging up.\n",
|
|
(errno) ? strerror(errno) : "Unspecified");
|
|
return NULL;
|
|
}
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
/* If this was handled by the ICE session don't do anything further */
|
|
if (!res) {
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
if (!*(rtcpdata + AST_FRIENDLY_OFFSET)) {
|
|
struct sockaddr_in addr_tmp;
|
|
struct ast_sockaddr addr_v4;
|
|
|
|
if (ast_sockaddr_is_ipv4(&addr)) {
|
|
ast_sockaddr_to_sin(&addr, &addr_tmp);
|
|
} else if (ast_sockaddr_ipv4_mapped(&addr, &addr_v4)) {
|
|
ast_debug(1, "Using IPv6 mapped address %s for STUN\n",
|
|
ast_sockaddr_stringify(&addr));
|
|
ast_sockaddr_to_sin(&addr_v4, &addr_tmp);
|
|
} else {
|
|
ast_debug(1, "Cannot do STUN for non IPv4 address %s\n",
|
|
ast_sockaddr_stringify(&addr));
|
|
return &ast_null_frame;
|
|
}
|
|
if ((ast_stun_handle_packet(rtp->rtcp->s, &addr_tmp, rtcpdata + AST_FRIENDLY_OFFSET, res, NULL, NULL) == AST_STUN_ACCEPT)) {
|
|
ast_sockaddr_from_sin(&addr, &addr_tmp);
|
|
ast_sockaddr_copy(&rtp->rtcp->them, &addr);
|
|
}
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
packetwords = res / 4;
|
|
|
|
if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
|
|
/* Send to whoever sent to us */
|
|
if (ast_sockaddr_cmp(&rtp->rtcp->them, &addr)) {
|
|
ast_sockaddr_copy(&rtp->rtcp->them, &addr);
|
|
if (rtpdebug) {
|
|
ast_debug(0, "RTCP NAT: Got RTCP from other end. Now sending to address %s\n",
|
|
ast_sockaddr_stringify(&rtp->rtcp->them));
|
|
}
|
|
}
|
|
}
|
|
|
|
ast_debug(1, "Got RTCP report of %d bytes\n", res);
|
|
|
|
ast_rtp_instance_get_local_address(instance, &local_address);
|
|
|
|
while (position < packetwords) {
|
|
int i, pt, rc;
|
|
unsigned int length;
|
|
struct ast_json *message_blob;
|
|
RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report, NULL, ao2_cleanup);
|
|
|
|
i = position;
|
|
length = ntohl(rtcpheader[i]);
|
|
pt = (length & 0xff0000) >> 16;
|
|
rc = (length & 0x1f000000) >> 24;
|
|
length &= 0xffff;
|
|
|
|
rtcp_report = ast_rtp_rtcp_report_alloc(rc);
|
|
if (!rtcp_report) {
|
|
return &ast_null_frame;
|
|
}
|
|
rtcp_report->reception_report_count = rc;
|
|
rtcp_report->ssrc = ntohl(rtcpheader[i + 1]);
|
|
|
|
if ((i + length) > packetwords) {
|
|
if (rtpdebug) {
|
|
ast_debug(1, "RTCP Read too short\n");
|
|
}
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
if (rtcp_debug_test_addr(&addr)) {
|
|
ast_verbose("\n\nGot RTCP from %s\n",
|
|
ast_sockaddr_stringify(&addr));
|
|
ast_verbose("PT: %d(%s)\n", pt, (pt == RTCP_PT_SR) ? "Sender Report" :
|
|
(pt == RTCP_PT_RR) ? "Receiver Report" :
|
|
(pt == RTCP_PT_FUR) ? "H.261 FUR" : "Unknown");
|
|
ast_verbose("Reception reports: %d\n", rc);
|
|
ast_verbose("SSRC of sender: %u\n", rtcp_report->ssrc);
|
|
}
|
|
|
|
i += 2; /* Advance past header and ssrc */
|
|
switch (pt) {
|
|
case RTCP_PT_SR:
|
|
gettimeofday(&rtp->rtcp->rxlsr, NULL);
|
|
rtp->rtcp->themrxlsr = ((ntohl(rtcpheader[i]) & 0x0000ffff) << 16) | ((ntohl(rtcpheader[i + 1]) & 0xffff0000) >> 16);
|
|
rtp->rtcp->spc = ntohl(rtcpheader[i + 3]);
|
|
rtp->rtcp->soc = ntohl(rtcpheader[i + 4]);
|
|
|
|
rtcp_report->type = RTCP_PT_SR;
|
|
rtcp_report->sender_information.packet_count = rtp->rtcp->spc;
|
|
rtcp_report->sender_information.octet_count = rtp->rtcp->soc;
|
|
ntp2timeval((unsigned int)ntohl(rtcpheader[i]),
|
|
(unsigned int)ntohl(rtcpheader[i + 1]),
|
|
&rtcp_report->sender_information.ntp_timestamp);
|
|
rtcp_report->sender_information.rtp_timestamp = ntohl(rtcpheader[i + 2]);
|
|
if (rtcp_debug_test_addr(&addr)) {
|
|
ast_verbose("NTP timestamp: %u.%010u\n",
|
|
(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_sec,
|
|
(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec * 4096);
|
|
ast_verbose("RTP timestamp: %u\n", rtcp_report->sender_information.rtp_timestamp);
|
|
ast_verbose("SPC: %u\tSOC: %u\n",
|
|
rtcp_report->sender_information.packet_count,
|
|
rtcp_report->sender_information.octet_count);
|
|
}
|
|
i += 5;
|
|
/* Intentional fall through */
|
|
case RTCP_PT_RR:
|
|
if (rtcp_report->type != RTCP_PT_SR) {
|
|
rtcp_report->type = RTCP_PT_RR;
|
|
}
|
|
|
|
if (rc > 0) {
|
|
/* Don't handle multiple reception reports (rc > 1) yet */
|
|
report_block = ast_calloc(1, sizeof(*report_block));
|
|
if (!report_block) {
|
|
return &ast_null_frame;
|
|
}
|
|
rtcp_report->report_block[report_counter] = report_block;
|
|
report_block->source_ssrc = ntohl(rtcpheader[i]);
|
|
report_block->lost_count.packets = ntohl(rtcpheader[i + 1]) & 0x00ffffff;
|
|
report_block->lost_count.fraction = ((ntohl(rtcpheader[i + 1]) & 0xff000000) >> 24);
|
|
report_block->highest_seq_no = ntohl(rtcpheader[i + 2]);
|
|
report_block->ia_jitter = ntohl(rtcpheader[i + 3]);
|
|
report_block->lsr = ntohl(rtcpheader[i + 4]);
|
|
report_block->dlsr = ntohl(rtcpheader[i + 5]);
|
|
if (report_block->lsr
|
|
&& update_rtt_stats(rtp, report_block->lsr, report_block->dlsr)
|
|
&& rtcp_debug_test_addr(&addr)) {
|
|
struct timeval now;
|
|
unsigned int lsr_now, lsw, msw;
|
|
gettimeofday(&now, NULL);
|
|
timeval2ntp(now, &msw, &lsw);
|
|
lsr_now = (((msw & 0xffff) << 16) | ((lsw & 0xffff0000) >> 16));
|
|
ast_verbose("Internal RTCP NTP clock skew detected: "
|
|
"lsr=%u, now=%u, dlsr=%u (%u:%03ums), "
|
|
"diff=%u\n",
|
|
report_block->lsr, lsr_now, report_block->dlsr, report_block->dlsr / 65536,
|
|
(report_block->dlsr % 65536) * 1000 / 65536,
|
|
report_block->dlsr - (lsr_now - report_block->lsr));
|
|
}
|
|
update_jitter_stats(rtp, report_block->ia_jitter);
|
|
update_lost_stats(rtp, report_block->lost_count.packets);
|
|
rtp->rtcp->reported_jitter_count++;
|
|
|
|
if (rtcp_debug_test_addr(&addr)) {
|
|
ast_verbose(" Fraction lost: %d\n", report_block->lost_count.fraction);
|
|
ast_verbose(" Packets lost so far: %u\n", report_block->lost_count.packets);
|
|
ast_verbose(" Highest sequence number: %u\n", report_block->highest_seq_no & 0x0000ffff);
|
|
ast_verbose(" Sequence number cycles: %u\n", report_block->highest_seq_no >> 16);
|
|
ast_verbose(" Interarrival jitter: %u\n", report_block->ia_jitter);
|
|
ast_verbose(" Last SR(our NTP): %lu.%010lu\n",(unsigned long)(report_block->lsr) >> 16,((unsigned long)(report_block->lsr) << 16) * 4096);
|
|
ast_verbose(" DLSR: %4.4f (sec)\n",(double)report_block->dlsr / 65536.0);
|
|
ast_verbose(" RTT: %4.4f(sec)\n", rtp->rtcp->rtt);
|
|
}
|
|
report_counter++;
|
|
}
|
|
/* If and when we handle more than one report block, this should occur outside
|
|
* this loop.
|
|
*/
|
|
if (!ast_find_ourip(&real_local_address, &local_address, 0)) {
|
|
str_local_address = ast_strdupa(ast_sockaddr_stringify(&real_local_address));
|
|
} else {
|
|
str_local_address = ast_strdupa(ast_sockaddr_stringify(&local_address));
|
|
}
|
|
|
|
if (!ast_find_ourip(&real_remote_address, &addr, 0)) {
|
|
str_remote_address = ast_strdupa(ast_sockaddr_stringify(&real_remote_address));
|
|
} else {
|
|
str_remote_address = ast_strdupa(ast_sockaddr_stringify(&addr));
|
|
}
|
|
|
|
message_blob = ast_json_pack("{s: s, s: s, s: f}",
|
|
"from", str_remote_address,
|
|
"to", str_local_address,
|
|
"rtt", rtp->rtcp->rtt);
|
|
ast_rtp_publish_rtcp_message(instance, ast_rtp_rtcp_received_type(),
|
|
rtcp_report,
|
|
message_blob);
|
|
ast_json_unref(message_blob);
|
|
break;
|
|
case RTCP_PT_FUR:
|
|
/* Handle RTCP FIR as FUR */
|
|
case RTCP_PT_PSFB:
|
|
if (rtcp_debug_test_addr(&addr)) {
|
|
ast_verbose("Received an RTCP Fast Update Request\n");
|
|
}
|
|
rtp->f.frametype = AST_FRAME_CONTROL;
|
|
rtp->f.subclass.integer = AST_CONTROL_VIDUPDATE;
|
|
rtp->f.datalen = 0;
|
|
rtp->f.samples = 0;
|
|
rtp->f.mallocd = 0;
|
|
rtp->f.src = "RTP";
|
|
f = &rtp->f;
|
|
break;
|
|
case RTCP_PT_SDES:
|
|
if (rtcp_debug_test_addr(&addr)) {
|
|
ast_verbose("Received an SDES from %s\n",
|
|
ast_sockaddr_stringify(&rtp->rtcp->them));
|
|
}
|
|
break;
|
|
case RTCP_PT_BYE:
|
|
if (rtcp_debug_test_addr(&addr)) {
|
|
ast_verbose("Received a BYE from %s\n",
|
|
ast_sockaddr_stringify(&rtp->rtcp->them));
|
|
}
|
|
break;
|
|
default:
|
|
ast_debug(1, "Unknown RTCP packet (pt=%d) received from %s\n",
|
|
pt, ast_sockaddr_stringify(&rtp->rtcp->them));
|
|
break;
|
|
}
|
|
position += (length + 1);
|
|
}
|
|
rtp->rtcp->rtcp_info = 1;
|
|
|
|
return f;
|
|
}
|
|
|
|
static int bridge_p2p_rtp_write(struct ast_rtp_instance *instance, unsigned int *rtpheader, int len, int hdrlen)
|
|
{
|
|
struct ast_rtp_instance *instance1 = ast_rtp_instance_get_bridged(instance);
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance), *bridged = ast_rtp_instance_get_data(instance1);
|
|
int res = 0, payload = 0, bridged_payload = 0, mark;
|
|
RAII_VAR(struct ast_rtp_payload_type *, payload_type, NULL, ao2_cleanup);
|
|
int reconstruct = ntohl(rtpheader[0]);
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
int ice;
|
|
|
|
/* Get fields from packet */
|
|
payload = (reconstruct & 0x7f0000) >> 16;
|
|
mark = (((reconstruct & 0x800000) >> 23) != 0);
|
|
|
|
/* Check what the payload value should be */
|
|
payload_type = ast_rtp_codecs_get_payload(ast_rtp_instance_get_codecs(instance), payload);
|
|
if (!payload_type) {
|
|
return -1;
|
|
}
|
|
|
|
/* Otherwise adjust bridged payload to match */
|
|
bridged_payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance1), payload_type->asterisk_format, payload_type->format, payload_type->rtp_code);
|
|
|
|
/* If no codec could be matched between instance and instance1, then somehow things were made incompatible while we were still bridged. Bail. */
|
|
if (bridged_payload < 0) {
|
|
return -1;
|
|
}
|
|
|
|
/* If the payload coming in is not one of the negotiated ones then send it to the core, this will cause formats to change and the bridge to break */
|
|
if (ast_rtp_codecs_find_payload_code(ast_rtp_instance_get_codecs(instance1), bridged_payload) == -1) {
|
|
ast_debug(1, "Unsupported payload type received \n");
|
|
return -1;
|
|
}
|
|
|
|
/* If the marker bit has been explicitly set turn it on */
|
|
if (ast_test_flag(rtp, FLAG_NEED_MARKER_BIT)) {
|
|
mark = 1;
|
|
ast_clear_flag(rtp, FLAG_NEED_MARKER_BIT);
|
|
}
|
|
|
|
/* Reconstruct part of the packet */
|
|
reconstruct &= 0xFF80FFFF;
|
|
reconstruct |= (bridged_payload << 16);
|
|
reconstruct |= (mark << 23);
|
|
rtpheader[0] = htonl(reconstruct);
|
|
|
|
ast_rtp_instance_get_remote_address(instance1, &remote_address);
|
|
|
|
if (ast_sockaddr_isnull(&remote_address)) {
|
|
ast_debug(5, "Remote address is null, most likely RTP has been stopped\n");
|
|
return 0;
|
|
}
|
|
|
|
/* Send the packet back out */
|
|
res = rtp_sendto(instance1, (void *)rtpheader, len, 0, &remote_address, &ice);
|
|
if (res < 0) {
|
|
if (!ast_rtp_instance_get_prop(instance1, AST_RTP_PROPERTY_NAT) || (ast_rtp_instance_get_prop(instance1, AST_RTP_PROPERTY_NAT) && (ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_ACTIVE))) {
|
|
ast_log(LOG_WARNING,
|
|
"RTP Transmission error of packet to %s: %s\n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
strerror(errno));
|
|
} else if (((ast_test_flag(bridged, FLAG_NAT_ACTIVE) == FLAG_NAT_INACTIVE) || rtpdebug) && !ast_test_flag(bridged, FLAG_NAT_INACTIVE_NOWARN)) {
|
|
if (option_debug || rtpdebug) {
|
|
ast_log(LOG_WARNING,
|
|
"RTP NAT: Can't write RTP to private "
|
|
"address %s, waiting for other end to "
|
|
"send audio...\n",
|
|
ast_sockaddr_stringify(&remote_address));
|
|
}
|
|
ast_set_flag(bridged, FLAG_NAT_INACTIVE_NOWARN);
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
if (rtp_debug_test_addr(&remote_address)) {
|
|
ast_verbose("Sent RTP P2P packet to %s%s (type %-2.2d, len %-6.6d)\n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
ice ? " (via ICE)" : "",
|
|
bridged_payload, len - hdrlen);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtcp)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_sockaddr addr;
|
|
int res, hdrlen = 12, version, payloadtype, padding, mark, ext, cc, prev_seqno;
|
|
unsigned int *rtpheader = (unsigned int*)(rtp->rawdata + AST_FRIENDLY_OFFSET), seqno, ssrc, timestamp;
|
|
RAII_VAR(struct ast_rtp_payload_type *, payload, NULL, ao2_cleanup);
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
struct frame_list frames;
|
|
|
|
/* If this is actually RTCP let's hop on over and handle it */
|
|
if (rtcp) {
|
|
if (rtp->rtcp) {
|
|
return ast_rtcp_read(instance);
|
|
}
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
/* If we are currently sending DTMF to the remote party send a continuation packet */
|
|
if (rtp->sending_digit) {
|
|
ast_rtp_dtmf_continuation(instance);
|
|
}
|
|
|
|
/* Actually read in the data from the socket */
|
|
if ((res = rtp_recvfrom(instance, rtp->rawdata + AST_FRIENDLY_OFFSET,
|
|
sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET, 0,
|
|
&addr)) < 0) {
|
|
ast_assert(errno != EBADF);
|
|
if (errno != EAGAIN) {
|
|
ast_log(LOG_WARNING, "RTP Read error: %s. Hanging up.\n",
|
|
(errno) ? strerror(errno) : "Unspecified");
|
|
return NULL;
|
|
}
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
/* If this was handled by the ICE session don't do anything */
|
|
if (!res) {
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
/* Make sure the data that was read in is actually enough to make up an RTP packet */
|
|
if (res < hdrlen) {
|
|
/* If this is a keepalive containing only nulls, don't bother with a warning */
|
|
int i;
|
|
for (i = 0; i < res; ++i) {
|
|
if (rtp->rawdata[AST_FRIENDLY_OFFSET + i] != '\0') {
|
|
ast_log(LOG_WARNING, "RTP Read too short\n");
|
|
return &ast_null_frame;
|
|
}
|
|
}
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
/* Get fields and verify this is an RTP packet */
|
|
seqno = ntohl(rtpheader[0]);
|
|
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
|
|
if (!(version = (seqno & 0xC0000000) >> 30)) {
|
|
struct sockaddr_in addr_tmp;
|
|
struct ast_sockaddr addr_v4;
|
|
if (ast_sockaddr_is_ipv4(&addr)) {
|
|
ast_sockaddr_to_sin(&addr, &addr_tmp);
|
|
} else if (ast_sockaddr_ipv4_mapped(&addr, &addr_v4)) {
|
|
ast_debug(1, "Using IPv6 mapped address %s for STUN\n",
|
|
ast_sockaddr_stringify(&addr));
|
|
ast_sockaddr_to_sin(&addr_v4, &addr_tmp);
|
|
} else {
|
|
ast_debug(1, "Cannot do STUN for non IPv4 address %s\n",
|
|
ast_sockaddr_stringify(&addr));
|
|
return &ast_null_frame;
|
|
}
|
|
if ((ast_stun_handle_packet(rtp->s, &addr_tmp, rtp->rawdata + AST_FRIENDLY_OFFSET, res, NULL, NULL) == AST_STUN_ACCEPT) &&
|
|
ast_sockaddr_isnull(&remote_address)) {
|
|
ast_sockaddr_from_sin(&addr, &addr_tmp);
|
|
ast_rtp_instance_set_remote_address(instance, &addr);
|
|
}
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
/* If strict RTP protection is enabled see if we need to learn the remote address or if we need to drop the packet */
|
|
if (rtp->strict_rtp_state == STRICT_RTP_LEARN) {
|
|
ast_debug(1, "%p -- Probation learning mode pass with source address %s\n", rtp, ast_sockaddr_stringify(&addr));
|
|
/* For now, we always copy the address. */
|
|
ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
|
|
|
|
/* Send the rtp and the seqno from header to rtp_learning_rtp_seq_update to see whether we can exit or not*/
|
|
if (rtp_learning_rtp_seq_update(&rtp->rtp_source_learn, seqno)) {
|
|
ast_debug(1, "%p -- Probation at seq %d with %d to go; discarding frame\n",
|
|
rtp, rtp->rtp_source_learn.max_seq, rtp->rtp_source_learn.packets);
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
ast_verb(4, "%p -- Probation passed - setting RTP source address to %s\n", rtp, ast_sockaddr_stringify(&addr));
|
|
rtp->strict_rtp_state = STRICT_RTP_CLOSED;
|
|
}
|
|
if (rtp->strict_rtp_state == STRICT_RTP_CLOSED) {
|
|
if (!ast_sockaddr_cmp(&rtp->strict_rtp_address, &addr)) {
|
|
/* Always reset the alternate learning source */
|
|
rtp_learning_seq_init(&rtp->alt_source_learn, seqno);
|
|
} else {
|
|
/* Start trying to learn from the new address. If we pass a probationary period with
|
|
* it, that means we've stopped getting RTP from the original source and we should
|
|
* switch to it.
|
|
*/
|
|
if (rtp_learning_rtp_seq_update(&rtp->alt_source_learn, seqno)) {
|
|
ast_debug(1, "%p -- Received RTP packet from %s, dropping due to strict RTP protection. Will switch to it in %d packets\n",
|
|
rtp, ast_sockaddr_stringify(&addr), rtp->alt_source_learn.packets);
|
|
return &ast_null_frame;
|
|
}
|
|
ast_verb(4, "%p -- Switching RTP source address to %s\n", rtp, ast_sockaddr_stringify(&addr));
|
|
ast_sockaddr_copy(&rtp->strict_rtp_address, &addr);
|
|
}
|
|
}
|
|
|
|
/* If symmetric RTP is enabled see if the remote side is not what we expected and change where we are sending audio */
|
|
if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_NAT)) {
|
|
if (ast_sockaddr_cmp(&remote_address, &addr)) {
|
|
/* do not update the originally given address, but only the remote */
|
|
ast_rtp_instance_set_incoming_source_address(instance, &addr);
|
|
ast_sockaddr_copy(&remote_address, &addr);
|
|
if (rtp->rtcp) {
|
|
ast_sockaddr_copy(&rtp->rtcp->them, &addr);
|
|
ast_sockaddr_set_port(&rtp->rtcp->them, ast_sockaddr_port(&addr) + 1);
|
|
}
|
|
rtp->rxseqno = 0;
|
|
ast_set_flag(rtp, FLAG_NAT_ACTIVE);
|
|
if (rtpdebug)
|
|
ast_debug(0, "RTP NAT: Got audio from other end. Now sending to address %s\n",
|
|
ast_sockaddr_stringify(&remote_address));
|
|
}
|
|
}
|
|
|
|
/* If we are directly bridged to another instance send the audio directly out */
|
|
if (ast_rtp_instance_get_bridged(instance) && !bridge_p2p_rtp_write(instance, rtpheader, res, hdrlen)) {
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
/* If the version is not what we expected by this point then just drop the packet */
|
|
if (version != 2) {
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
/* Pull out the various other fields we will need */
|
|
payloadtype = (seqno & 0x7f0000) >> 16;
|
|
padding = seqno & (1 << 29);
|
|
mark = seqno & (1 << 23);
|
|
ext = seqno & (1 << 28);
|
|
cc = (seqno & 0xF000000) >> 24;
|
|
seqno &= 0xffff;
|
|
timestamp = ntohl(rtpheader[1]);
|
|
ssrc = ntohl(rtpheader[2]);
|
|
|
|
AST_LIST_HEAD_INIT_NOLOCK(&frames);
|
|
/* Force a marker bit and change SSRC if the SSRC changes */
|
|
if (rtp->rxssrc && rtp->rxssrc != ssrc) {
|
|
struct ast_frame *f, srcupdate = {
|
|
AST_FRAME_CONTROL,
|
|
.subclass.integer = AST_CONTROL_SRCCHANGE,
|
|
};
|
|
|
|
if (!mark) {
|
|
if (rtpdebug) {
|
|
ast_debug(1, "Forcing Marker bit, because SSRC has changed\n");
|
|
}
|
|
mark = 1;
|
|
}
|
|
|
|
f = ast_frisolate(&srcupdate);
|
|
AST_LIST_INSERT_TAIL(&frames, f, frame_list);
|
|
|
|
rtp->seedrxseqno = 0;
|
|
rtp->rxcount = 0;
|
|
rtp->cycles = 0;
|
|
rtp->lastrxseqno = 0;
|
|
rtp->last_seqno = 0;
|
|
rtp->last_end_timestamp = 0;
|
|
if (rtp->rtcp) {
|
|
rtp->rtcp->expected_prior = 0;
|
|
rtp->rtcp->received_prior = 0;
|
|
}
|
|
}
|
|
|
|
rtp->rxssrc = ssrc;
|
|
|
|
/* Remove any padding bytes that may be present */
|
|
if (padding) {
|
|
res -= rtp->rawdata[AST_FRIENDLY_OFFSET + res - 1];
|
|
}
|
|
|
|
/* Skip over any CSRC fields */
|
|
if (cc) {
|
|
hdrlen += cc * 4;
|
|
}
|
|
|
|
/* Look for any RTP extensions, currently we do not support any */
|
|
if (ext) {
|
|
hdrlen += (ntohl(rtpheader[hdrlen/4]) & 0xffff) << 2;
|
|
hdrlen += 4;
|
|
if (option_debug) {
|
|
unsigned int profile;
|
|
profile = (ntohl(rtpheader[3]) & 0xffff0000) >> 16;
|
|
if (profile == 0x505a)
|
|
ast_debug(1, "Found Zfone extension in RTP stream - zrtp - not supported.\n");
|
|
else
|
|
ast_debug(1, "Found unknown RTP Extensions %x\n", profile);
|
|
}
|
|
}
|
|
|
|
/* Make sure after we potentially mucked with the header length that it is once again valid */
|
|
if (res < hdrlen) {
|
|
ast_log(LOG_WARNING, "RTP Read too short (%d, expecting %d\n", res, hdrlen);
|
|
return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
|
|
}
|
|
|
|
rtp->rxcount++;
|
|
if (rtp->rxcount == 1) {
|
|
rtp->seedrxseqno = seqno;
|
|
}
|
|
|
|
/* Do not schedule RR if RTCP isn't run */
|
|
if (rtp->rtcp && !ast_sockaddr_isnull(&rtp->rtcp->them) && rtp->rtcp->schedid < 1) {
|
|
/* Schedule transmission of Receiver Report */
|
|
ao2_ref(instance, +1);
|
|
rtp->rtcp->schedid = ast_sched_add(rtp->sched, ast_rtcp_calc_interval(rtp), ast_rtcp_write, instance);
|
|
if (rtp->rtcp->schedid < 0) {
|
|
ao2_ref(instance, -1);
|
|
ast_log(LOG_WARNING, "scheduling RTCP transmission failed.\n");
|
|
}
|
|
}
|
|
if ((int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */
|
|
rtp->cycles += RTP_SEQ_MOD;
|
|
|
|
prev_seqno = rtp->lastrxseqno;
|
|
rtp->lastrxseqno = seqno;
|
|
|
|
if (!rtp->themssrc) {
|
|
rtp->themssrc = ntohl(rtpheader[2]); /* Record their SSRC to put in future RR */
|
|
}
|
|
|
|
if (rtp_debug_test_addr(&addr)) {
|
|
ast_verbose("Got RTP packet from %s (type %-2.2d, seq %-6.6u, ts %-6.6u, len %-6.6d)\n",
|
|
ast_sockaddr_stringify(&addr),
|
|
payloadtype, seqno, timestamp,res - hdrlen);
|
|
}
|
|
|
|
payload = ast_rtp_codecs_get_payload(ast_rtp_instance_get_codecs(instance), payloadtype);
|
|
|
|
/* If the payload is not actually an Asterisk one but a special one pass it off to the respective handler */
|
|
if (!payload->asterisk_format) {
|
|
struct ast_frame *f = NULL;
|
|
if (payload->rtp_code == AST_RTP_DTMF) {
|
|
/* process_dtmf_rfc2833 may need to return multiple frames. We do this
|
|
* by passing the pointer to the frame list to it so that the method
|
|
* can append frames to the list as needed.
|
|
*/
|
|
process_dtmf_rfc2833(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark, &frames);
|
|
} else if (payload->rtp_code == AST_RTP_CISCO_DTMF) {
|
|
f = process_dtmf_cisco(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark);
|
|
} else if (payload->rtp_code == AST_RTP_CN) {
|
|
f = process_cn_rfc3389(instance, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen, seqno, timestamp, &addr, payloadtype, mark);
|
|
} else {
|
|
ast_log(LOG_NOTICE, "Unknown RTP codec %d received from '%s'\n",
|
|
payloadtype,
|
|
ast_sockaddr_stringify(&remote_address));
|
|
}
|
|
|
|
if (f) {
|
|
AST_LIST_INSERT_TAIL(&frames, f, frame_list);
|
|
}
|
|
/* Even if no frame was returned by one of the above methods,
|
|
* we may have a frame to return in our frame list
|
|
*/
|
|
if (!AST_LIST_EMPTY(&frames)) {
|
|
return AST_LIST_FIRST(&frames);
|
|
}
|
|
return &ast_null_frame;
|
|
}
|
|
|
|
ao2_replace(rtp->lastrxformat, payload->format);
|
|
ao2_replace(rtp->f.subclass.format, payload->format);
|
|
switch (ast_format_get_type(rtp->f.subclass.format)) {
|
|
case AST_MEDIA_TYPE_AUDIO:
|
|
rtp->f.frametype = AST_FRAME_VOICE;
|
|
break;
|
|
case AST_MEDIA_TYPE_VIDEO:
|
|
rtp->f.frametype = AST_FRAME_VIDEO;
|
|
break;
|
|
case AST_MEDIA_TYPE_TEXT:
|
|
rtp->f.frametype = AST_FRAME_TEXT;
|
|
break;
|
|
case AST_MEDIA_TYPE_IMAGE:
|
|
/* Fall through */
|
|
default:
|
|
ast_log(LOG_WARNING, "Unknown or unsupported media type: %s\n",
|
|
ast_codec_media_type2str(ast_format_get_type(rtp->f.subclass.format)));
|
|
return &ast_null_frame;
|
|
}
|
|
rtp->rxseqno = seqno;
|
|
|
|
if (rtp->dtmf_timeout && rtp->dtmf_timeout < timestamp) {
|
|
rtp->dtmf_timeout = 0;
|
|
|
|
if (rtp->resp) {
|
|
struct ast_frame *f;
|
|
f = create_dtmf_frame(instance, AST_FRAME_DTMF_END, 0);
|
|
f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, rtp_get_rate(f->subclass.format)), ast_tv(0, 0));
|
|
rtp->resp = 0;
|
|
rtp->dtmf_timeout = rtp->dtmf_duration = 0;
|
|
AST_LIST_INSERT_TAIL(&frames, f, frame_list);
|
|
return AST_LIST_FIRST(&frames);
|
|
}
|
|
}
|
|
|
|
rtp->lastrxts = timestamp;
|
|
|
|
rtp->f.src = "RTP";
|
|
rtp->f.mallocd = 0;
|
|
rtp->f.datalen = res - hdrlen;
|
|
rtp->f.data.ptr = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
|
|
rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
|
|
rtp->f.seqno = seqno;
|
|
|
|
if ((ast_format_cmp(rtp->f.subclass.format, ast_format_t140) == AST_FORMAT_CMP_EQUAL)
|
|
&& ((int)seqno - (prev_seqno + 1) > 0)
|
|
&& ((int)seqno - (prev_seqno + 1) < 10)) {
|
|
unsigned char *data = rtp->f.data.ptr;
|
|
|
|
memmove(rtp->f.data.ptr+3, rtp->f.data.ptr, rtp->f.datalen);
|
|
rtp->f.datalen +=3;
|
|
*data++ = 0xEF;
|
|
*data++ = 0xBF;
|
|
*data = 0xBD;
|
|
}
|
|
|
|
if (ast_format_cmp(rtp->f.subclass.format, ast_format_t140_red) == AST_FORMAT_CMP_EQUAL) {
|
|
unsigned char *data = rtp->f.data.ptr;
|
|
unsigned char *header_end;
|
|
int num_generations;
|
|
int header_length;
|
|
int len;
|
|
int diff =(int)seqno - (prev_seqno+1); /* if diff = 0, no drop*/
|
|
int x;
|
|
|
|
ao2_replace(rtp->f.subclass.format, ast_format_t140);
|
|
header_end = memchr(data, ((*data) & 0x7f), rtp->f.datalen);
|
|
if (header_end == NULL) {
|
|
return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
|
|
}
|
|
header_end++;
|
|
|
|
header_length = header_end - data;
|
|
num_generations = header_length / 4;
|
|
len = header_length;
|
|
|
|
if (!diff) {
|
|
for (x = 0; x < num_generations; x++)
|
|
len += data[x * 4 + 3];
|
|
|
|
if (!(rtp->f.datalen - len))
|
|
return AST_LIST_FIRST(&frames) ? AST_LIST_FIRST(&frames) : &ast_null_frame;
|
|
|
|
rtp->f.data.ptr += len;
|
|
rtp->f.datalen -= len;
|
|
} else if (diff > num_generations && diff < 10) {
|
|
len -= 3;
|
|
rtp->f.data.ptr += len;
|
|
rtp->f.datalen -= len;
|
|
|
|
data = rtp->f.data.ptr;
|
|
*data++ = 0xEF;
|
|
*data++ = 0xBF;
|
|
*data = 0xBD;
|
|
} else {
|
|
for ( x = 0; x < num_generations - diff; x++)
|
|
len += data[x * 4 + 3];
|
|
|
|
rtp->f.data.ptr += len;
|
|
rtp->f.datalen -= len;
|
|
}
|
|
}
|
|
|
|
if (ast_format_get_type(rtp->f.subclass.format) == AST_MEDIA_TYPE_AUDIO) {
|
|
rtp->f.samples = ast_codec_samples_count(&rtp->f);
|
|
if (ast_format_cache_is_slinear(rtp->f.subclass.format)) {
|
|
ast_frame_byteswap_be(&rtp->f);
|
|
}
|
|
calc_rxstamp(&rtp->f.delivery, rtp, timestamp, mark);
|
|
/* Add timing data to let ast_generic_bridge() put the frame into a jitterbuf */
|
|
ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
|
|
rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass.format) / 1000);
|
|
rtp->f.len = rtp->f.samples / ((ast_format_get_sample_rate(rtp->f.subclass.format) / 1000));
|
|
} else if (ast_format_get_type(rtp->f.subclass.format) == AST_MEDIA_TYPE_VIDEO) {
|
|
/* 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;
|
|
/* Pass the RTP marker bit as bit */
|
|
rtp->f.subclass.frame_ending = mark;
|
|
} else if (ast_format_get_type(rtp->f.subclass.format) == AST_MEDIA_TYPE_TEXT) {
|
|
/* TEXT -- samples is # of samples vs. 1000 */
|
|
if (!rtp->lastitexttimestamp)
|
|
rtp->lastitexttimestamp = timestamp;
|
|
rtp->f.samples = timestamp - rtp->lastitexttimestamp;
|
|
rtp->lastitexttimestamp = timestamp;
|
|
rtp->f.delivery.tv_sec = 0;
|
|
rtp->f.delivery.tv_usec = 0;
|
|
} else {
|
|
ast_log(LOG_WARNING, "Unknown or unsupported media type: %s\n",
|
|
ast_codec_media_type2str(ast_format_get_type(rtp->f.subclass.format)));
|
|
return &ast_null_frame;;
|
|
}
|
|
|
|
AST_LIST_INSERT_TAIL(&frames, &rtp->f, frame_list);
|
|
return AST_LIST_FIRST(&frames);
|
|
}
|
|
|
|
static void ast_rtp_prop_set(struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (property == AST_RTP_PROPERTY_RTCP) {
|
|
if (value) {
|
|
if (rtp->rtcp) {
|
|
ast_debug(1, "Ignoring duplicate RTCP property on RTP instance '%p'\n", instance);
|
|
return;
|
|
}
|
|
/* Setup RTCP to be activated on the next RTP write */
|
|
if (!(rtp->rtcp = ast_calloc(1, sizeof(*rtp->rtcp)))) {
|
|
return;
|
|
}
|
|
|
|
/* Grab the IP address and port we are going to use */
|
|
ast_rtp_instance_get_local_address(instance, &rtp->rtcp->us);
|
|
ast_sockaddr_set_port(&rtp->rtcp->us,
|
|
ast_sockaddr_port(&rtp->rtcp->us) + 1);
|
|
|
|
if ((rtp->rtcp->s =
|
|
create_new_socket("RTCP",
|
|
ast_sockaddr_is_ipv4(&rtp->rtcp->us) ?
|
|
AF_INET :
|
|
ast_sockaddr_is_ipv6(&rtp->rtcp->us) ?
|
|
AF_INET6 : -1)) < 0) {
|
|
ast_debug(1, "Failed to create a new socket for RTCP on instance '%p'\n", instance);
|
|
ast_free(rtp->rtcp);
|
|
rtp->rtcp = NULL;
|
|
return;
|
|
}
|
|
|
|
/* Try to actually bind to the IP address and port we are going to use for RTCP, if this fails we have to bail out */
|
|
if (ast_bind(rtp->rtcp->s, &rtp->rtcp->us)) {
|
|
ast_debug(1, "Failed to setup RTCP on RTP instance '%p'\n", instance);
|
|
close(rtp->rtcp->s);
|
|
ast_free(rtp->rtcp);
|
|
rtp->rtcp = NULL;
|
|
return;
|
|
}
|
|
|
|
ast_debug(1, "Setup RTCP on RTP instance '%p'\n", instance);
|
|
rtp->rtcp->schedid = -1;
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
if (rtp->ice) {
|
|
rtp_add_candidates_to_ice(instance, rtp, &rtp->rtcp->us, ast_sockaddr_port(&rtp->rtcp->us), AST_RTP_ICE_COMPONENT_RTCP, TRANSPORT_SOCKET_RTCP);
|
|
}
|
|
#endif
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
rtp->rtcp->dtls.timeout_timer = -1;
|
|
dtls_setup_rtcp(instance);
|
|
#endif
|
|
|
|
return;
|
|
} else {
|
|
if (rtp->rtcp) {
|
|
if (rtp->rtcp->schedid > 0) {
|
|
if (!ast_sched_del(rtp->sched, rtp->rtcp->schedid)) {
|
|
/* Successfully cancelled scheduler entry. */
|
|
ao2_ref(instance, -1);
|
|
} else {
|
|
/* Unable to cancel scheduler entry */
|
|
ast_debug(1, "Failed to tear down RTCP on RTP instance '%p'\n", instance);
|
|
return;
|
|
}
|
|
rtp->rtcp->schedid = -1;
|
|
}
|
|
close(rtp->rtcp->s);
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
if (rtp->rtcp->dtls.ssl) {
|
|
SSL_free(rtp->rtcp->dtls.ssl);
|
|
}
|
|
#endif
|
|
ast_free(rtp->rtcp);
|
|
rtp->rtcp = NULL;
|
|
}
|
|
return;
|
|
}
|
|
}
|
|
|
|
return;
|
|
}
|
|
|
|
static int ast_rtp_fd(struct ast_rtp_instance *instance, int rtcp)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
return rtcp ? (rtp->rtcp ? rtp->rtcp->s : -1) : rtp->s;
|
|
}
|
|
|
|
static void ast_rtp_remote_address_set(struct ast_rtp_instance *instance, struct ast_sockaddr *addr)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (rtp->rtcp) {
|
|
ast_debug(1, "Setting RTCP address on RTP instance '%p'\n", instance);
|
|
ast_sockaddr_copy(&rtp->rtcp->them, addr);
|
|
if (!ast_sockaddr_isnull(addr)) {
|
|
ast_sockaddr_set_port(&rtp->rtcp->them,
|
|
ast_sockaddr_port(addr) + 1);
|
|
}
|
|
}
|
|
|
|
rtp->rxseqno = 0;
|
|
|
|
if (strictrtp && rtp->strict_rtp_state != STRICT_RTP_OPEN) {
|
|
rtp->strict_rtp_state = STRICT_RTP_LEARN;
|
|
rtp_learning_seq_init(&rtp->rtp_source_learn, rtp->seqno);
|
|
}
|
|
|
|
return;
|
|
}
|
|
|
|
/*! \brief Write t140 redundacy frame
|
|
* \param data primary data to be buffered
|
|
*/
|
|
static int red_write(const void *data)
|
|
{
|
|
struct ast_rtp_instance *instance = (struct ast_rtp_instance*) data;
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
ast_rtp_write(instance, &rtp->red->t140);
|
|
|
|
return 1;
|
|
}
|
|
|
|
static int rtp_red_init(struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
int x;
|
|
|
|
if (!(rtp->red = ast_calloc(1, sizeof(*rtp->red)))) {
|
|
return -1;
|
|
}
|
|
|
|
rtp->red->t140.frametype = AST_FRAME_TEXT;
|
|
ao2_replace(rtp->red->t140.subclass.format, ast_format_t140_red);
|
|
rtp->red->t140.data.ptr = &rtp->red->buf_data;
|
|
|
|
rtp->red->t140.ts = 0;
|
|
rtp->red->t140red = rtp->red->t140;
|
|
rtp->red->t140red.data.ptr = &rtp->red->t140red_data;
|
|
rtp->red->t140red.datalen = 0;
|
|
rtp->red->ti = buffer_time;
|
|
rtp->red->num_gen = generations;
|
|
rtp->red->hdrlen = generations * 4 + 1;
|
|
rtp->red->prev_ts = 0;
|
|
|
|
for (x = 0; x < generations; x++) {
|
|
rtp->red->pt[x] = payloads[x];
|
|
rtp->red->pt[x] |= 1 << 7; /* mark redundant generations pt */
|
|
rtp->red->t140red_data[x*4] = rtp->red->pt[x];
|
|
}
|
|
rtp->red->t140red_data[x*4] = rtp->red->pt[x] = payloads[x]; /* primary pt */
|
|
rtp->red->schedid = ast_sched_add(rtp->sched, generations, red_write, instance);
|
|
|
|
rtp->red->t140.datalen = 0;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int rtp_red_buffer(struct ast_rtp_instance *instance, struct ast_frame *frame)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (frame->datalen > -1) {
|
|
struct rtp_red *red = rtp->red;
|
|
memcpy(&red->buf_data[red->t140.datalen], frame->data.ptr, frame->datalen);
|
|
red->t140.datalen += frame->datalen;
|
|
red->t140.ts = frame->ts;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int ast_rtp_local_bridge(struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance0);
|
|
|
|
ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int ast_rtp_get_stat(struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
if (!rtp->rtcp) {
|
|
return -1;
|
|
}
|
|
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXCOUNT, -1, stats->txcount, rtp->txcount);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXCOUNT, -1, stats->rxcount, rtp->rxcount);
|
|
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->txploss, rtp->rtcp->reported_lost);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->rxploss, rtp->rtcp->expected_prior - rtp->rtcp->received_prior);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_maxrxploss, rtp->rtcp->reported_maxlost);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_minrxploss, rtp->rtcp->reported_minlost);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_normdevrxploss, rtp->rtcp->reported_normdev_lost);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->remote_stdevrxploss, rtp->rtcp->reported_stdev_lost);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_maxrxploss, rtp->rtcp->maxrxlost);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_minrxploss, rtp->rtcp->minrxlost);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_normdevrxploss, rtp->rtcp->normdev_rxlost);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, stats->local_stdevrxploss, rtp->rtcp->stdev_rxlost);
|
|
AST_RTP_STAT_TERMINATOR(AST_RTP_INSTANCE_STAT_COMBINED_LOSS);
|
|
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_TXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->txjitter, rtp->rxjitter);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->rxjitter, rtp->rtcp->reported_jitter / (unsigned int) 65536.0);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_maxjitter, rtp->rtcp->reported_maxjitter);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_minjitter, rtp->rtcp->reported_minjitter);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_normdevjitter, rtp->rtcp->reported_normdev_jitter);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->remote_stdevjitter, rtp->rtcp->reported_stdev_jitter);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_maxjitter, rtp->rtcp->maxrxjitter);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_minjitter, rtp->rtcp->minrxjitter);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_normdevjitter, rtp->rtcp->normdev_rxjitter);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, stats->local_stdevjitter, rtp->rtcp->stdev_rxjitter);
|
|
AST_RTP_STAT_TERMINATOR(AST_RTP_INSTANCE_STAT_COMBINED_JITTER);
|
|
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->rtt, rtp->rtcp->rtt);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_MAX_RTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->maxrtt, rtp->rtcp->maxrtt);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_MIN_RTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->minrtt, rtp->rtcp->minrtt);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_NORMDEVRTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->normdevrtt, rtp->rtcp->normdevrtt);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_STDEVRTT, AST_RTP_INSTANCE_STAT_COMBINED_RTT, stats->stdevrtt, rtp->rtcp->stdevrtt);
|
|
AST_RTP_STAT_TERMINATOR(AST_RTP_INSTANCE_STAT_COMBINED_RTT);
|
|
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_LOCAL_SSRC, -1, stats->local_ssrc, rtp->ssrc);
|
|
AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_REMOTE_SSRC, -1, stats->remote_ssrc, rtp->themssrc);
|
|
AST_RTP_STAT_STRCPY(AST_RTP_INSTANCE_STAT_CHANNEL_UNIQUEID, -1, stats->channel_uniqueid, ast_rtp_instance_get_channel_id(instance));
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int ast_rtp_dtmf_compatible(struct ast_channel *chan0, struct ast_rtp_instance *instance0, struct ast_channel *chan1, struct ast_rtp_instance *instance1)
|
|
{
|
|
/* If both sides are not using the same method of DTMF transmission
|
|
* (ie: one is RFC2833, other is INFO... then we can not do direct media.
|
|
* --------------------------------------------------
|
|
* | DTMF Mode | HAS_DTMF | Accepts Begin Frames |
|
|
* |-----------|------------|-----------------------|
|
|
* | Inband | False | True |
|
|
* | RFC2833 | True | True |
|
|
* | SIP INFO | False | False |
|
|
* --------------------------------------------------
|
|
*/
|
|
return (((ast_rtp_instance_get_prop(instance0, AST_RTP_PROPERTY_DTMF) != ast_rtp_instance_get_prop(instance1, AST_RTP_PROPERTY_DTMF)) ||
|
|
(!ast_channel_tech(chan0)->send_digit_begin != !ast_channel_tech(chan1)->send_digit_begin)) ? 0 : 1);
|
|
}
|
|
|
|
static void ast_rtp_stun_request(struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct sockaddr_in suggestion_tmp;
|
|
|
|
ast_sockaddr_to_sin(suggestion, &suggestion_tmp);
|
|
ast_stun_request(rtp->s, &suggestion_tmp, username, NULL);
|
|
ast_sockaddr_from_sin(suggestion, &suggestion_tmp);
|
|
}
|
|
|
|
static void ast_rtp_stop(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_sockaddr addr = { {0,} };
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
AST_SCHED_DEL_UNREF(rtp->sched, rtp->rekeyid, ao2_ref(instance, -1));
|
|
|
|
dtls_srtp_stop_timeout_timer(instance, rtp, 0);
|
|
if (rtp->rtcp) {
|
|
dtls_srtp_stop_timeout_timer(instance, rtp, 1);
|
|
}
|
|
#endif
|
|
|
|
if (rtp->rtcp && rtp->rtcp->schedid > 0) {
|
|
if (!ast_sched_del(rtp->sched, rtp->rtcp->schedid)) {
|
|
/* successfully cancelled scheduler entry. */
|
|
ao2_ref(instance, -1);
|
|
}
|
|
rtp->rtcp->schedid = -1;
|
|
}
|
|
|
|
if (rtp->red) {
|
|
AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
|
|
free(rtp->red);
|
|
rtp->red = NULL;
|
|
}
|
|
|
|
ast_rtp_instance_set_remote_address(instance, &addr);
|
|
if (rtp->rtcp) {
|
|
ast_sockaddr_setnull(&rtp->rtcp->them);
|
|
}
|
|
|
|
ast_set_flag(rtp, FLAG_NEED_MARKER_BIT);
|
|
}
|
|
|
|
static int ast_rtp_qos_set(struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
return ast_set_qos(rtp->s, tos, cos, desc);
|
|
}
|
|
|
|
/*! \brief generate comfort noice (CNG) */
|
|
static int ast_rtp_sendcng(struct ast_rtp_instance *instance, int level)
|
|
{
|
|
unsigned int *rtpheader;
|
|
int hdrlen = 12;
|
|
int res, payload = 0;
|
|
char data[256];
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
struct ast_sockaddr remote_address = { {0,} };
|
|
int ice;
|
|
|
|
ast_rtp_instance_get_remote_address(instance, &remote_address);
|
|
|
|
if (ast_sockaddr_isnull(&remote_address)) {
|
|
return -1;
|
|
}
|
|
|
|
payload = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 0, NULL, AST_RTP_CN);
|
|
|
|
level = 127 - (level & 0x7f);
|
|
|
|
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) | (payload << 16) | (rtp->seqno));
|
|
rtpheader[1] = htonl(rtp->lastts);
|
|
rtpheader[2] = htonl(rtp->ssrc);
|
|
data[12] = level;
|
|
|
|
res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 1, 0, &remote_address, &ice);
|
|
|
|
if (res < 0) {
|
|
ast_log(LOG_ERROR, "RTP Comfort Noise Transmission error to %s: %s\n", ast_sockaddr_stringify(&remote_address), strerror(errno));
|
|
return res;
|
|
}
|
|
|
|
if (rtp_debug_test_addr(&remote_address)) {
|
|
ast_verbose("Sent Comfort Noise RTP packet to %s%s (type %-2.2d, seq %-6.6d, ts %-6.6u, len %-6.6d)\n",
|
|
ast_sockaddr_stringify(&remote_address),
|
|
ice ? " (via ICE)" : "",
|
|
AST_RTP_CN, rtp->seqno, rtp->lastdigitts, res - hdrlen);
|
|
}
|
|
|
|
rtp->seqno++;
|
|
|
|
return res;
|
|
}
|
|
|
|
#ifdef HAVE_OPENSSL_SRTP
|
|
static void dtls_perform_setup(struct dtls_details *dtls)
|
|
{
|
|
if (!dtls->ssl || !SSL_is_init_finished(dtls->ssl)) {
|
|
return;
|
|
}
|
|
|
|
SSL_clear(dtls->ssl);
|
|
if (dtls->dtls_setup == AST_RTP_DTLS_SETUP_PASSIVE) {
|
|
SSL_set_accept_state(dtls->ssl);
|
|
} else {
|
|
SSL_set_connect_state(dtls->ssl);
|
|
}
|
|
dtls->connection = AST_RTP_DTLS_CONNECTION_NEW;
|
|
}
|
|
|
|
static int ast_rtp_activate(struct ast_rtp_instance *instance)
|
|
{
|
|
struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
|
|
|
|
dtls_perform_setup(&rtp->dtls);
|
|
|
|
if (rtp->rtcp) {
|
|
dtls_perform_setup(&rtp->rtcp->dtls);
|
|
}
|
|
|
|
/* If ICE negotiation is enabled the DTLS Handshake will be performed upon completion of it */
|
|
#ifdef HAVE_PJPROJECT
|
|
if (rtp->ice) {
|
|
return 0;
|
|
}
|
|
#endif
|
|
|
|
dtls_perform_handshake(instance, &rtp->dtls, 0);
|
|
|
|
if (rtp->rtcp) {
|
|
dtls_perform_handshake(instance, &rtp->rtcp->dtls, 1);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
#endif
|
|
|
|
static char *rtp_do_debug_ip(struct ast_cli_args *a)
|
|
{
|
|
char *arg = ast_strdupa(a->argv[4]);
|
|
char *debughost = NULL;
|
|
char *debugport = NULL;
|
|
|
|
if (!ast_sockaddr_parse(&rtpdebugaddr, arg, 0) || !ast_sockaddr_split_hostport(arg, &debughost, &debugport, 0)) {
|
|
ast_cli(a->fd, "Lookup failed for '%s'\n", arg);
|
|
return CLI_FAILURE;
|
|
}
|
|
rtpdebugport = (!ast_strlen_zero(debugport) && debugport[0] != '0');
|
|
ast_cli(a->fd, "RTP Debugging Enabled for address: %s\n",
|
|
ast_sockaddr_stringify(&rtpdebugaddr));
|
|
rtpdebug = 1;
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static char *rtcp_do_debug_ip(struct ast_cli_args *a)
|
|
{
|
|
char *arg = ast_strdupa(a->argv[4]);
|
|
char *debughost = NULL;
|
|
char *debugport = NULL;
|
|
|
|
if (!ast_sockaddr_parse(&rtcpdebugaddr, arg, 0) || !ast_sockaddr_split_hostport(arg, &debughost, &debugport, 0)) {
|
|
ast_cli(a->fd, "Lookup failed for '%s'\n", arg);
|
|
return CLI_FAILURE;
|
|
}
|
|
rtcpdebugport = (!ast_strlen_zero(debugport) && debugport[0] != '0');
|
|
ast_cli(a->fd, "RTCP Debugging Enabled for address: %s\n",
|
|
ast_sockaddr_stringify(&rtcpdebugaddr));
|
|
rtcpdebug = 1;
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static char *handle_cli_rtp_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "rtp set debug {on|off|ip}";
|
|
e->usage =
|
|
"Usage: rtp set debug {on|off|ip host[:port]}\n"
|
|
" Enable/Disable dumping of all RTP packets. If 'ip' is\n"
|
|
" specified, limit the dumped packets to those to and from\n"
|
|
" the specified 'host' with optional port.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc == e->args) { /* set on or off */
|
|
if (!strncasecmp(a->argv[e->args-1], "on", 2)) {
|
|
rtpdebug = 1;
|
|
memset(&rtpdebugaddr, 0, sizeof(rtpdebugaddr));
|
|
ast_cli(a->fd, "RTP Debugging Enabled\n");
|
|
return CLI_SUCCESS;
|
|
} else if (!strncasecmp(a->argv[e->args-1], "off", 3)) {
|
|
rtpdebug = 0;
|
|
ast_cli(a->fd, "RTP Debugging Disabled\n");
|
|
return CLI_SUCCESS;
|
|
}
|
|
} else if (a->argc == e->args +1) { /* ip */
|
|
return rtp_do_debug_ip(a);
|
|
}
|
|
|
|
return CLI_SHOWUSAGE; /* default, failure */
|
|
}
|
|
|
|
static char *handle_cli_rtcp_set_debug(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "rtcp set debug {on|off|ip}";
|
|
e->usage =
|
|
"Usage: rtcp set debug {on|off|ip host[:port]}\n"
|
|
" Enable/Disable dumping of all RTCP packets. If 'ip' is\n"
|
|
" specified, limit the dumped packets to those to and from\n"
|
|
" the specified 'host' with optional port.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc == e->args) { /* set on or off */
|
|
if (!strncasecmp(a->argv[e->args-1], "on", 2)) {
|
|
rtcpdebug = 1;
|
|
memset(&rtcpdebugaddr, 0, sizeof(rtcpdebugaddr));
|
|
ast_cli(a->fd, "RTCP Debugging Enabled\n");
|
|
return CLI_SUCCESS;
|
|
} else if (!strncasecmp(a->argv[e->args-1], "off", 3)) {
|
|
rtcpdebug = 0;
|
|
ast_cli(a->fd, "RTCP Debugging Disabled\n");
|
|
return CLI_SUCCESS;
|
|
}
|
|
} else if (a->argc == e->args +1) { /* ip */
|
|
return rtcp_do_debug_ip(a);
|
|
}
|
|
|
|
return CLI_SHOWUSAGE; /* default, failure */
|
|
}
|
|
|
|
static char *handle_cli_rtcp_set_stats(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
|
|
{
|
|
switch (cmd) {
|
|
case CLI_INIT:
|
|
e->command = "rtcp set stats {on|off}";
|
|
e->usage =
|
|
"Usage: rtcp set stats {on|off}\n"
|
|
" Enable/Disable dumping of RTCP stats.\n";
|
|
return NULL;
|
|
case CLI_GENERATE:
|
|
return NULL;
|
|
}
|
|
|
|
if (a->argc != e->args)
|
|
return CLI_SHOWUSAGE;
|
|
|
|
if (!strncasecmp(a->argv[e->args-1], "on", 2))
|
|
rtcpstats = 1;
|
|
else if (!strncasecmp(a->argv[e->args-1], "off", 3))
|
|
rtcpstats = 0;
|
|
else
|
|
return CLI_SHOWUSAGE;
|
|
|
|
ast_cli(a->fd, "RTCP Stats %s\n", rtcpstats ? "Enabled" : "Disabled");
|
|
return CLI_SUCCESS;
|
|
}
|
|
|
|
static struct ast_cli_entry cli_rtp[] = {
|
|
AST_CLI_DEFINE(handle_cli_rtp_set_debug, "Enable/Disable RTP debugging"),
|
|
AST_CLI_DEFINE(handle_cli_rtcp_set_debug, "Enable/Disable RTCP debugging"),
|
|
AST_CLI_DEFINE(handle_cli_rtcp_set_stats, "Enable/Disable RTCP stats"),
|
|
};
|
|
|
|
static int rtp_reload(int reload)
|
|
{
|
|
struct ast_config *cfg;
|
|
const char *s;
|
|
struct ast_flags config_flags = { reload ? CONFIG_FLAG_FILEUNCHANGED : 0 };
|
|
|
|
cfg = ast_config_load2("rtp.conf", "rtp", config_flags);
|
|
if (cfg == CONFIG_STATUS_FILEMISSING || cfg == CONFIG_STATUS_FILEUNCHANGED || cfg == CONFIG_STATUS_FILEINVALID) {
|
|
return 0;
|
|
}
|
|
|
|
rtpstart = DEFAULT_RTP_START;
|
|
rtpend = DEFAULT_RTP_END;
|
|
dtmftimeout = DEFAULT_DTMF_TIMEOUT;
|
|
strictrtp = DEFAULT_STRICT_RTP;
|
|
learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL;
|
|
|
|
/** This resource is not "reloaded" so much as unloaded and loaded again.
|
|
* In the case of the TURN related variables, the memory referenced by a
|
|
* previously loaded instance *should* have been released when the
|
|
* corresponding pool was destroyed. If at some point in the future this
|
|
* resource were to support ACTUAL live reconfiguration and did NOT release
|
|
* the pool this will cause a small memory leak.
|
|
*/
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
icesupport = DEFAULT_ICESUPPORT;
|
|
turnport = DEFAULT_TURN_PORT;
|
|
memset(&stunaddr, 0, sizeof(stunaddr));
|
|
turnaddr = pj_str(NULL);
|
|
turnusername = pj_str(NULL);
|
|
turnpassword = pj_str(NULL);
|
|
#endif
|
|
|
|
if (cfg) {
|
|
if ((s = ast_variable_retrieve(cfg, "general", "rtpstart"))) {
|
|
rtpstart = atoi(s);
|
|
if (rtpstart < MINIMUM_RTP_PORT)
|
|
rtpstart = MINIMUM_RTP_PORT;
|
|
if (rtpstart > MAXIMUM_RTP_PORT)
|
|
rtpstart = MAXIMUM_RTP_PORT;
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "rtpend"))) {
|
|
rtpend = atoi(s);
|
|
if (rtpend < MINIMUM_RTP_PORT)
|
|
rtpend = MINIMUM_RTP_PORT;
|
|
if (rtpend > MAXIMUM_RTP_PORT)
|
|
rtpend = MAXIMUM_RTP_PORT;
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "rtcpinterval"))) {
|
|
rtcpinterval = atoi(s);
|
|
if (rtcpinterval == 0)
|
|
rtcpinterval = 0; /* Just so we're clear... it's zero */
|
|
if (rtcpinterval < RTCP_MIN_INTERVALMS)
|
|
rtcpinterval = RTCP_MIN_INTERVALMS; /* This catches negative numbers too */
|
|
if (rtcpinterval > RTCP_MAX_INTERVALMS)
|
|
rtcpinterval = RTCP_MAX_INTERVALMS;
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "rtpchecksums"))) {
|
|
#ifdef SO_NO_CHECK
|
|
nochecksums = ast_false(s) ? 1 : 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 > 64000)) {
|
|
ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n",
|
|
dtmftimeout, DEFAULT_DTMF_TIMEOUT);
|
|
dtmftimeout = DEFAULT_DTMF_TIMEOUT;
|
|
};
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "strictrtp"))) {
|
|
strictrtp = ast_true(s);
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "probation"))) {
|
|
if ((sscanf(s, "%d", &learning_min_sequential) <= 0) || learning_min_sequential <= 0) {
|
|
ast_log(LOG_WARNING, "Value for 'probation' could not be read, using default of '%d' instead\n",
|
|
DEFAULT_LEARNING_MIN_SEQUENTIAL);
|
|
}
|
|
}
|
|
#ifdef HAVE_PJPROJECT
|
|
if ((s = ast_variable_retrieve(cfg, "general", "icesupport"))) {
|
|
icesupport = ast_true(s);
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "stunaddr"))) {
|
|
stunaddr.sin_port = htons(STANDARD_STUN_PORT);
|
|
if (ast_parse_arg(s, PARSE_INADDR, &stunaddr)) {
|
|
ast_log(LOG_WARNING, "Invalid STUN server address: %s\n", s);
|
|
}
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "turnaddr"))) {
|
|
struct sockaddr_in addr;
|
|
addr.sin_port = htons(DEFAULT_TURN_PORT);
|
|
if (ast_parse_arg(s, PARSE_INADDR, &addr)) {
|
|
ast_log(LOG_WARNING, "Invalid TURN server address: %s\n", s);
|
|
} else {
|
|
pj_strdup2_with_null(pool, &turnaddr, ast_inet_ntoa(addr.sin_addr));
|
|
/* ntohs() is not a bug here. The port number is used in host byte order with
|
|
* a pjnat API. */
|
|
turnport = ntohs(addr.sin_port);
|
|
}
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "turnusername"))) {
|
|
pj_strdup2_with_null(pool, &turnusername, s);
|
|
}
|
|
if ((s = ast_variable_retrieve(cfg, "general", "turnpassword"))) {
|
|
pj_strdup2_with_null(pool, &turnpassword, s);
|
|
}
|
|
#endif
|
|
ast_config_destroy(cfg);
|
|
}
|
|
if (rtpstart >= rtpend) {
|
|
ast_log(LOG_WARNING, "Unreasonable values for RTP start/end port in rtp.conf\n");
|
|
rtpstart = DEFAULT_RTP_START;
|
|
rtpend = DEFAULT_RTP_END;
|
|
}
|
|
ast_verb(2, "RTP Allocating from port range %d -> %d\n", rtpstart, rtpend);
|
|
return 0;
|
|
}
|
|
|
|
static int reload_module(void)
|
|
{
|
|
rtp_reload(1);
|
|
return 0;
|
|
}
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
static void rtp_terminate_pjproject(void)
|
|
{
|
|
pj_thread_register_check();
|
|
|
|
if (timer_thread) {
|
|
timer_terminate = 1;
|
|
pj_thread_join(timer_thread);
|
|
pj_thread_destroy(timer_thread);
|
|
}
|
|
|
|
pj_caching_pool_destroy(&cachingpool);
|
|
pj_shutdown();
|
|
}
|
|
#endif
|
|
|
|
static int load_module(void)
|
|
{
|
|
#ifdef HAVE_PJPROJECT
|
|
pj_lock_t *lock;
|
|
|
|
if (pj_init() != PJ_SUCCESS) {
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (pjlib_util_init() != PJ_SUCCESS) {
|
|
rtp_terminate_pjproject();
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (pjnath_init() != PJ_SUCCESS) {
|
|
rtp_terminate_pjproject();
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
pj_caching_pool_init(&cachingpool, &pj_pool_factory_default_policy, 0);
|
|
|
|
pool = pj_pool_create(&cachingpool.factory, "timer", 512, 512, NULL);
|
|
|
|
if (pj_timer_heap_create(pool, 100, &timer_heap) != PJ_SUCCESS) {
|
|
rtp_terminate_pjproject();
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (pj_lock_create_recursive_mutex(pool, "rtp%p", &lock) != PJ_SUCCESS) {
|
|
rtp_terminate_pjproject();
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
pj_timer_heap_set_lock(timer_heap, lock, PJ_TRUE);
|
|
|
|
if (pj_thread_create(pool, "timer", &timer_worker_thread, NULL, 0, 0, &timer_thread) != PJ_SUCCESS) {
|
|
rtp_terminate_pjproject();
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
#endif
|
|
|
|
if (ast_rtp_engine_register(&asterisk_rtp_engine)) {
|
|
#ifdef HAVE_PJPROJECT
|
|
rtp_terminate_pjproject();
|
|
#endif
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
if (ast_cli_register_multiple(cli_rtp, ARRAY_LEN(cli_rtp))) {
|
|
#ifdef HAVE_PJPROJECT
|
|
ast_rtp_engine_unregister(&asterisk_rtp_engine);
|
|
rtp_terminate_pjproject();
|
|
#endif
|
|
return AST_MODULE_LOAD_DECLINE;
|
|
}
|
|
|
|
rtp_reload(0);
|
|
|
|
return AST_MODULE_LOAD_SUCCESS;
|
|
}
|
|
|
|
static int unload_module(void)
|
|
{
|
|
ast_rtp_engine_unregister(&asterisk_rtp_engine);
|
|
ast_cli_unregister_multiple(cli_rtp, ARRAY_LEN(cli_rtp));
|
|
|
|
#ifdef HAVE_PJPROJECT
|
|
pj_thread_register_check();
|
|
rtp_terminate_pjproject();
|
|
#endif
|
|
|
|
return 0;
|
|
}
|
|
|
|
AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Asterisk RTP Stack",
|
|
.support_level = AST_MODULE_SUPPORT_CORE,
|
|
.load = load_module,
|
|
.unload = unload_module,
|
|
.reload = reload_module,
|
|
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
|
|
);
|