res_rtp_asterisk.c File Reference

Supports RTP and RTCP with Symmetric RTP support for NAT traversal. More...

#include "asterisk.h"
#include <sys/time.h>
#include <signal.h>
#include <fcntl.h>
#include <openssl/opensslconf.h>
#include <openssl/opensslv.h>
#include <pjlib.h>
#include <pjlib-util.h>
#include <pjnath.h>
#include <ifaddrs.h>
#include "asterisk/conversions.h"
#include "asterisk/options.h"
#include "asterisk/logger_category.h"
#include "asterisk/stun.h"
#include "asterisk/pbx.h"
#include "asterisk/frame.h"
#include "asterisk/format_cache.h"
#include "asterisk/channel.h"
#include "asterisk/acl.h"
#include "asterisk/config.h"
#include "asterisk/lock.h"
#include "asterisk/utils.h"
#include "asterisk/cli.h"
#include "asterisk/manager.h"
#include "asterisk/unaligned.h"
#include "asterisk/module.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/smoother.h"
#include "asterisk/uuid.h"
#include "asterisk/test.h"
#include "asterisk/data_buffer.h"
#include "asterisk/res_pjproject.h"
#include "asterisk/security_events.h"

Include dependency graph for res_rtp_asterisk.c:

Go to the source code of this file.

Data Structures
struct	ast_ice_host_candidate
	Structure which contains ICE host candidate mapping information. More...
struct	ast_rtcp
	Structure defining an RTCP session. More...
struct	ast_rtp
	RTP session description. More...
struct	ast_rtp_ioqueue_thread
	Structure which contains ioqueue thread information. More...
struct	ast_rtp_rtcp_nack_payload
	Structure for storing RTP packets for retransmission. More...
struct	frame_list
struct	host_candidates
	List of ICE host candidate mappings. More...
struct	ice_wrap
struct	ioqueues
	List of ioqueue threads. More...
struct	optional_ts
struct	rtp_drop_packets_data
struct	rtp_learning_info
	RTP learning mode tracking information. More...
struct	rtp_red
struct	rtp_ssrc_mapping
	Structure used for mapping an incoming SSRC to an RTP instance. More...
struct	rtp_transport_wide_cc_packet_statistics
	Packet statistics (used for transport-cc) More...
struct	rtp_transport_wide_cc_statistics
	Statistics information (used for transport-cc) More...

Macros
#define	CALC_LEARNING_MIN_DURATION(count)   (((count) - 1) * 9 - 5)
	Calculate the min learning duration in ms. More...
#define	DEFAULT_DTLS_MTU   1200
#define	DEFAULT_DTMF_TIMEOUT   (150 * (8000 / 1000))
#define	DEFAULT_ICESUPPORT   1
#define	DEFAULT_LEARNING_MIN_DURATION   CALC_LEARNING_MIN_DURATION(DEFAULT_LEARNING_MIN_SEQUENTIAL)
#define	DEFAULT_LEARNING_MIN_SEQUENTIAL   4
#define	DEFAULT_RTP_END   31000
#define	DEFAULT_RTP_RECV_BUFFER_SIZE   20
#define	DEFAULT_RTP_SEND_BUFFER_SIZE   250
#define	DEFAULT_RTP_START   5000
#define	DEFAULT_SRTP_REPLAY_PROTECTION   1
#define	DEFAULT_STRICT_RTP   STRICT_RTP_YES
#define	DEFAULT_STUN_SOFTWARE_ATTRIBUTE   1
#define	DEFAULT_TURN_PORT   3478
#define	DTMF_SAMPLE_RATE_MS   8
#define	FLAG_3389_WARNING   (1 << 0)
#define	FLAG_DTMF_COMPENSATE   (1 << 4)
#define	FLAG_NAT_ACTIVE   (3 << 1)
#define	FLAG_NAT_INACTIVE   (0 << 1)
#define	FLAG_NAT_INACTIVE_NOWARN   (1 << 1)
#define	FLAG_NEED_MARKER_BIT   (1 << 3)
#define	FLAG_REQ_LOCAL_BRIDGE_BIT   (1 << 5)
#define	MAX_TIMESTAMP_SKEW   640
#define	MAXIMUM_RTP_PORT   65535
#define	MAXIMUM_RTP_RECV_BUFFER_SIZE   (DEFAULT_RTP_RECV_BUFFER_SIZE + 20)
#define	MAXIMUM_RTP_SEND_BUFFER_SIZE   (DEFAULT_RTP_SEND_BUFFER_SIZE + 200)
#define	MINIMUM_RTP_PORT   1024
#define	MISSING_SEQNOS_ADDED_TRIGGER   2
#define	OLD_PACKET_COUNT   1000
#define	RTCP_DEFAULT_INTERVALMS   5000
#define	RTCP_FB_NACK_BLOCK_WORD_LENGTH   2
#define	RTCP_FB_REMB_BLOCK_WORD_LENGTH   4
#define	RTCP_HEADER_SSRC_LENGTH   2
#define	RTCP_LENGTH_MASK   0xFFFF
#define	RTCP_LENGTH_SHIFT   0
#define	RTCP_MAX_INTERVALMS   60000
#define	RTCP_MIN_INTERVALMS   500
#define	RTCP_PADDING_MASK   0x01
#define	RTCP_PADDING_SHIFT   29
#define	RTCP_PAYLOAD_TYPE_MASK   0xFF
#define	RTCP_PAYLOAD_TYPE_SHIFT   16
#define	RTCP_PT_APP   204
#define	RTCP_PT_BYE   203
#define	RTCP_PT_FUR   192
#define	RTCP_PT_PSFB   AST_RTP_RTCP_PSFB
#define	RTCP_PT_RR   AST_RTP_RTCP_RR
#define	RTCP_PT_SDES   202
#define	RTCP_PT_SR   AST_RTP_RTCP_SR
#define	RTCP_REPORT_COUNT_MASK   0x1F
#define	RTCP_REPORT_COUNT_SHIFT   24
#define	RTCP_RR_BLOCK_WORD_LENGTH   6
#define	RTCP_SR_BLOCK_WORD_LENGTH   5
#define	RTCP_VALID_MASK   (RTCP_VERSION_MASK_SHIFTED | (((RTCP_PAYLOAD_TYPE_MASK & ~0x1)) << RTCP_PAYLOAD_TYPE_SHIFT))
#define	RTCP_VALID_VALUE   (RTCP_VERSION_SHIFTED | (RTCP_PT_SR << RTCP_PAYLOAD_TYPE_SHIFT))
#define	RTCP_VERSION   2U
#define	RTCP_VERSION_MASK   0x03
#define	RTCP_VERSION_MASK_SHIFTED   (RTCP_VERSION_MASK << RTCP_VERSION_SHIFT)
#define	RTCP_VERSION_SHIFT   30
#define	RTCP_VERSION_SHIFTED   (RTCP_VERSION << RTCP_VERSION_SHIFT)
#define	RTP_DTLS_ESTABLISHED   -37
#define	RTP_MTU   1200
#define	RTP_SEQ_MOD   (1<<16)
#define	SEQNO_CYCLE_OVER   65536
#define	SRTP_MASTER_KEY_LEN   16
#define	SRTP_MASTER_LEN   (SRTP_MASTER_KEY_LEN + SRTP_MASTER_SALT_LEN)
#define	SRTP_MASTER_SALT_LEN   14
#define	SSRC_MAPPING_ELEM_CMP(elem, value)   ((elem).instance == (value))
	SSRC mapping comparator for AST_VECTOR_REMOVE_CMP_UNORDERED() More...
#define	STRICT_RTP_LEARN_TIMEOUT   5000
	Strict RTP learning timeout time in milliseconds. More...
#define	TRANSPORT_SOCKET_RTCP   1
#define	TRANSPORT_SOCKET_RTP   0
#define	TRANSPORT_TURN_RTCP   3
#define	TRANSPORT_TURN_RTP   2
#define	TURN_STATE_WAIT_TIME   2000
#define	ZFONE_PROFILE_ID   0x505a

Enumerations
enum	strict_rtp_mode { STRICT_RTP_NO = 0, STRICT_RTP_YES, STRICT_RTP_SEQNO }
enum	strict_rtp_state { STRICT_RTP_OPEN = 0, STRICT_RTP_LEARN, STRICT_RTP_CLOSED }

Functions
static void	__init_pj_thread_storage (void)
static void	__reg_module (void)
static struct ast_rtp_instance *	__rtp_find_instance_by_ssrc (struct ast_rtp_instance *instance, struct ast_rtp *rtp, unsigned int ssrc, int source)
static int	__rtp_recvfrom (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int rtcp)
static int	__rtp_sendto (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int rtcp, int *via_ice, int use_srtp)
static void	__unreg_module (void)
static void	acl_change_stasis_cb (void *data, struct stasis_subscription *sub, struct stasis_message *message)
static void	ast2pj_rtp_ice_role (enum ast_rtp_ice_role ast_role, enum pj_ice_sess_role *pj_role)
struct ast_module *	AST_MODULE_SELF_SYM (void)
static unsigned int	ast_rtcp_calc_interval (struct ast_rtp *rtp)
static int	ast_rtcp_calculate_sr_rr_statistics (struct ast_rtp_instance *instance, struct ast_rtp_rtcp_report *rtcp_report, struct ast_sockaddr remote_address, int ice, int sr)
static int	ast_rtcp_generate_compound_prefix (struct ast_rtp_instance *instance, unsigned char *rtcpheader, struct ast_rtp_rtcp_report *report, int *sr)
static int	ast_rtcp_generate_nack (struct ast_rtp_instance *instance, unsigned char *rtcpheader)
static int	ast_rtcp_generate_report (struct ast_rtp_instance *instance, unsigned char *rtcpheader, struct ast_rtp_rtcp_report *rtcp_report, int *sr)
static int	ast_rtcp_generate_sdes (struct ast_rtp_instance *instance, unsigned char *rtcpheader, struct ast_rtp_rtcp_report *rtcp_report)
static struct ast_frame *	ast_rtcp_interpret (struct ast_rtp_instance *instance, struct ast_srtp *srtp, const unsigned char *rtcpdata, size_t size, struct ast_sockaddr *addr)
static struct ast_frame *	ast_rtcp_read (struct ast_rtp_instance *instance)
static int	ast_rtcp_write (const void *data)
	Write a RTCP packet to the far end. More...
static int	ast_rtp_bundle (struct ast_rtp_instance *child, struct ast_rtp_instance *parent)
static void	ast_rtp_change_source (struct ast_rtp_instance *instance)
static int	ast_rtp_destroy (struct ast_rtp_instance *instance)
static int	ast_rtp_dtmf_begin (struct ast_rtp_instance *instance, char digit)
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 int	ast_rtp_dtmf_continuation (struct ast_rtp_instance *instance)
static int	ast_rtp_dtmf_end (struct ast_rtp_instance *instance, char digit)
static int	ast_rtp_dtmf_end_with_duration (struct ast_rtp_instance *instance, char digit, unsigned int duration)
static enum ast_rtp_dtmf_mode	ast_rtp_dtmf_mode_get (struct ast_rtp_instance *instance)
static int	ast_rtp_dtmf_mode_set (struct ast_rtp_instance *instance, enum ast_rtp_dtmf_mode dtmf_mode)
static int	ast_rtp_extension_enable (struct ast_rtp_instance *instance, enum ast_rtp_extension extension)
static int	ast_rtp_fd (struct ast_rtp_instance *instance, int rtcp)
static const char *	ast_rtp_get_cname (struct ast_rtp_instance *instance)
static unsigned int	ast_rtp_get_ssrc (struct ast_rtp_instance *instance)
static int	ast_rtp_get_stat (struct ast_rtp_instance *instance, struct ast_rtp_instance_stats *stats, enum ast_rtp_instance_stat stat)
static void	ast_rtp_ice_add_cand (struct ast_rtp_instance *instance, 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)
static void	ast_rtp_ice_add_remote_candidate (struct ast_rtp_instance *instance, const struct ast_rtp_engine_ice_candidate *candidate)
static void	ast_rtp_ice_candidate_destroy (void *obj)
	Destructor for locally created ICE candidates. More...
static void	ast_rtp_ice_change_components (struct ast_rtp_instance *instance, int num_components)
static struct ao2_container *	ast_rtp_ice_get_local_candidates (struct ast_rtp_instance *instance)
static const char *	ast_rtp_ice_get_password (struct ast_rtp_instance *instance)
static const char *	ast_rtp_ice_get_ufrag (struct ast_rtp_instance *instance)
static void	ast_rtp_ice_lite (struct ast_rtp_instance *instance)
static void	ast_rtp_ice_set_authentication (struct ast_rtp_instance *instance, const char *ufrag, const char *password)
static void	ast_rtp_ice_set_role (struct ast_rtp_instance *instance, enum ast_rtp_ice_role role)
static void	ast_rtp_ice_start (struct ast_rtp_instance *instance)
static void	ast_rtp_ice_start_media (pj_ice_sess *ice, pj_status_t status)
static void	ast_rtp_ice_stop (struct ast_rtp_instance *instance)
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)
static struct ast_frame *	ast_rtp_interpret (struct ast_rtp_instance *instance, struct ast_srtp *srtp, const struct ast_sockaddr *remote_address, unsigned char *read_area, int length, int prev_seqno, unsigned int bundled)
static int	ast_rtp_local_bridge (struct ast_rtp_instance *instance0, struct ast_rtp_instance *instance1)
static int	ast_rtp_new (struct ast_rtp_instance *instance, struct ast_sched_context *sched, struct ast_sockaddr *addr, void *data)
static void	ast_rtp_on_ice_complete (pj_ice_sess *ice, pj_status_t status)
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)
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)
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)
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)
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)
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)
static void	ast_rtp_on_valid_pair (pj_ice_sess *ice)
static void	ast_rtp_prop_set (struct ast_rtp_instance *instance, enum ast_rtp_property property, int value)
static int	ast_rtp_qos_set (struct ast_rtp_instance *instance, int tos, int cos, const char *desc)
static struct ast_frame *	ast_rtp_read (struct ast_rtp_instance *instance, int rtcp)
static void	ast_rtp_remote_address_set (struct ast_rtp_instance *instance, struct ast_sockaddr *addr)
static int	ast_rtp_rtcp_handle_nack (struct ast_rtp_instance *instance, unsigned int *nackdata, unsigned int position, unsigned int length)
static int	ast_rtp_sendcng (struct ast_rtp_instance *instance, int level)
	generate comfort noice (CNG) More...
static void	ast_rtp_set_remote_ssrc (struct ast_rtp_instance *instance, unsigned int ssrc)
static void	ast_rtp_set_stream_num (struct ast_rtp_instance *instance, int stream_num)
static void	ast_rtp_stop (struct ast_rtp_instance *instance)
static void	ast_rtp_stun_request (struct ast_rtp_instance *instance, struct ast_sockaddr *suggestion, const char *username)
static void	ast_rtp_update_source (struct ast_rtp_instance *instance)
static int	ast_rtp_write (struct ast_rtp_instance *instance, struct ast_frame *frame)
static int	bridge_p2p_rtp_write (struct ast_rtp_instance *instance, struct ast_rtp_instance *instance1, unsigned int *rtpheader, int len, int hdrlen)
static void	calc_mean_and_standard_deviation (double new_sample, double *mean, double *std_dev, unsigned int *count)
static void	calc_rxstamp (struct timeval *tv, struct ast_rtp *rtp, unsigned int timestamp, int mark)
static unsigned int	calc_txstamp (struct ast_rtp *rtp, struct timeval *delivery)
static void	calculate_lost_packet_statistics (struct ast_rtp *rtp, unsigned int *lost_packets, int *fraction_lost)
static int	compare_by_value (int elem, int value)
	Helper function to compare an elem in a vector by value. More...
static struct ast_frame *	create_dtmf_frame (struct ast_rtp_instance *instance, enum ast_frame_type type, int compensate)
static int	create_new_socket (const char *type, int af)
static void	drop_packets_data_update (struct timeval tv)
static int	find_by_value (int elem, int value)
	Helper function to find an elem in a vector by value. More...
static char *	generate_random_string (char *buf, size_t size)
static size_t	get_recv_buffer_count (struct ast_rtp_instance *instance)
static size_t	get_recv_buffer_max (struct ast_rtp_instance *instance)
static size_t	get_send_buffer_count (struct ast_rtp_instance *instance)
static char *	handle_cli_rtcp_set_debug (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	handle_cli_rtcp_set_stats (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	handle_cli_rtp_drop_incoming_packets (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	handle_cli_rtp_set_debug (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static char *	handle_cli_rtp_settings (struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
static void	host_candidate_overrides_clear (void)
	Helper function which clears the ICE host candidate mapping. More...
static int	ice_candidate_cmp (void *obj, void *arg, int flags)
static int	ice_candidates_compare (struct ao2_container *left, struct ao2_container *right)
static int	ice_create (struct ast_rtp_instance *instance, struct ast_sockaddr *addr, int port, int replace)
static int	ice_reset_session (struct ast_rtp_instance *instance)
static void	ice_wrap_dtor (void *vdoomed)
	ao2 ICE wrapper object destructor. More...
static int	ioqueue_worker_thread (void *data)
	Worker thread for ioqueue and timerheap. More...
static int	load_module (void)
static void	ntp2timeval (unsigned int msw, unsigned int lsw, struct timeval *tv)
static void	pj2ast_rtp_ice_role (enum pj_ice_sess_role pj_role, enum ast_rtp_ice_role *ast_role)
static void	pj_thread_register_check (void)
	Function used to check if the calling thread is registered with pjlib. If it is not it will be registered. More...
static struct ast_frame *	process_cn_rfc3389 (struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, int payloadtype, int mark)
static struct ast_frame *	process_dtmf_cisco (struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, int payloadtype, int mark)
static void	process_dtmf_rfc2833 (struct ast_rtp_instance *instance, unsigned char *data, int len, unsigned int seqno, unsigned int timestamp, int payloadtype, int mark, struct frame_list *frames)
static void	put_unaligned_time24 (void *p, uint32_t time_msw, uint32_t time_lsw)
static struct ast_frame *	red_t140_to_red (struct rtp_red *red)
static int	red_write (const void *data)
	Write t140 redundacy frame. More...
static int	reload_module (void)
static int	rtcp_debug_test_addr (struct ast_sockaddr *addr)
static char *	rtcp_do_debug_ip (struct ast_cli_args *a)
static int	rtcp_mux (struct ast_rtp *rtp, const unsigned char *packet)
static const char *	rtcp_payload_subtype2str (unsigned int pt, unsigned int subtype)
static const char *	rtcp_payload_type2str (unsigned int pt)
static int	rtcp_recvfrom (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa)
static int	rtcp_sendto (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int *ice)
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)
static int	rtp_address_is_ice_blacklisted (const struct ast_sockaddr *address)
static int	rtp_allocate_transport (struct ast_rtp_instance *instance, struct ast_rtp *rtp)
static void	rtp_deallocate_transport (struct ast_rtp_instance *instance, struct ast_rtp *rtp)
static int	rtp_debug_test_addr (struct ast_sockaddr *addr)
static char *	rtp_do_debug_ip (struct ast_cli_args *a)
static struct ast_rtp_instance *	rtp_find_instance_by_media_source_ssrc (struct ast_rtp_instance *instance, struct ast_rtp *rtp, unsigned int ssrc)
static struct ast_rtp_instance *	rtp_find_instance_by_packet_source_ssrc (struct ast_rtp_instance *instance, struct ast_rtp *rtp, unsigned int ssrc)
static void	rtp_instance_parse_extmap_extensions (struct ast_rtp_instance *instance, struct ast_rtp *rtp, unsigned char *extension, int len)
static void	rtp_instance_parse_transport_wide_cc (struct ast_rtp_instance *instance, struct ast_rtp *rtp, unsigned char *data, int len)
static void	rtp_instance_unlock (struct ast_rtp_instance *instance)
static void	rtp_ioqueue_thread_destroy (struct ast_rtp_ioqueue_thread *ioqueue)
	Destroyer for ioqueue thread. More...
static struct ast_rtp_ioqueue_thread *	rtp_ioqueue_thread_get_or_create (void)
	Finder and allocator for an ioqueue thread. More...
static void	rtp_ioqueue_thread_remove (struct ast_rtp_ioqueue_thread *ioqueue)
	Removal function for ioqueue thread, determines if it should be terminated and destroyed. More...
static int	rtp_learning_rtp_seq_update (struct rtp_learning_info *info, uint16_t seq)
static void	rtp_learning_seq_init (struct rtp_learning_info *info, uint16_t seq)
static void	rtp_learning_start (struct ast_rtp *rtp)
	Start the strictrtp learning mode. More...
static int	rtp_raw_write (struct ast_rtp_instance *instance, struct ast_frame *frame, int codec)
static int	rtp_recvfrom (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa)
static int	rtp_red_buffer (struct ast_rtp_instance *instance, struct ast_frame *frame)
static int	rtp_red_init (struct ast_rtp_instance *instance, int buffer_time, int *payloads, int generations)
static int	rtp_reload (int reload, int by_external_config)
static int	rtp_sendto (struct ast_rtp_instance *instance, void *buf, size_t size, int flags, struct ast_sockaddr *sa, int *ice)
static void	rtp_terminate_pjproject (void)
static int	rtp_transport_wide_cc_feedback_produce (const void *data)
static void	rtp_transport_wide_cc_feedback_status_append (unsigned char *rtcpheader, int *packet_len, int *status_vector_chunk_bits, uint16_t *status_vector_chunk, int *run_length_chunk_count, int *run_length_chunk_status, int status)
static void	rtp_transport_wide_cc_feedback_status_vector_append (unsigned char *rtcpheader, int *packet_len, int *status_vector_chunk_bits, uint16_t *status_vector_chunk, int status)
static int	rtp_transport_wide_cc_packet_statistics_cmp (struct rtp_transport_wide_cc_packet_statistics a, struct rtp_transport_wide_cc_packet_statistics b)
static void	rtp_unload_acl (ast_rwlock_t *lock, struct ast_acl_list **acl)
static void	rtp_write_rtcp_fir (struct ast_rtp_instance *instance, struct ast_rtp *rtp, struct ast_sockaddr *remote_address)
static void	rtp_write_rtcp_psfb (struct ast_rtp_instance *instance, struct ast_rtp *rtp, struct ast_frame *frame, struct ast_sockaddr *remote_address)
static void	set_rtp_rtcp_schedid (struct ast_rtp_instance *instance, int id)
static int	should_drop_packets (struct ast_sockaddr *addr)
static int	stun_address_is_blacklisted (const struct ast_sockaddr *addr)
static int	timer_worker_thread (void *data)
	Worker thread for timerheap. More...
static void	timeval2ntp (struct timeval tv, unsigned int *msw, unsigned int *lsw)
static int	unload_module (void)
static void	update_address_with_ice_candidate (pj_ice_sess *ice, enum ast_rtp_ice_component_type component, struct ast_sockaddr *cand_address)
	Helper function which updates an ast_sockaddr with the candidate used for the component. More...
static void	update_jitter_stats (struct ast_rtp *rtp, unsigned int ia_jitter)
static void	update_lost_stats (struct ast_rtp *rtp, unsigned int lost_packets)
static int	update_rtt_stats (struct ast_rtp *rtp, unsigned int lsr, unsigned int dlsr)
static unsigned int	use_random (struct ast_cli_args *a, int pos, unsigned int index)

Variables
static struct ast_module_info	__mod_info = { .name = AST_MODULE, .flags = AST_MODFLAG_LOAD_ORDER , .description = "Asterisk RTP Stack" , .key = "This paragraph is copyright (c) 2006 by Digium, Inc. \In order for your module to load, it must return this \key via a function called \"key\". Any code which \includes this paragraph must be licensed under the GNU \General Public License version 2 or later (at your \option). In addition to Digium's general reservations \of rights, Digium expressly reserves the right to \allow other parties to license this paragraph under \different terms. Any use of Digium, Inc. trademarks or \logos (including \"Asterisk\" or \"Digium\") without \express written permission of Digium, Inc. is prohibited.\n" , .buildopt_sum = AST_BUILDOPT_SUM, .support_level = AST_MODULE_SUPPORT_CORE, .load = load_module, .unload = unload_module, .reload = reload_module, .load_pri = AST_MODPRI_CHANNEL_DEPEND, #ifdef 1 .requires = "res_pjproject", #endif }
static struct stasis_subscription *	acl_change_sub = NULL
static const struct ast_module_info *	ast_module_info = &__mod_info
static struct ast_rtp_engine_ice	ast_rtp_ice
static pj_ice_sess_cb	ast_rtp_ice_sess_cb
static struct ast_rtp_engine_test	ast_rtp_test
static pj_turn_sock_cb	ast_rtp_turn_rtcp_sock_cb
static pj_turn_sock_cb	ast_rtp_turn_rtp_sock_cb
static struct ast_rtp_engine	asterisk_rtp_engine
static pj_caching_pool	cachingpool
	Pool factory used by pjlib to allocate memory. More...
static struct ast_cli_entry	cli_rtp []
static struct rtp_drop_packets_data	drop_packets_data
static int	dtmftimeout = DEFAULT_DTMF_TIMEOUT
static struct host_candidates	host_candidates = { .first = NULL, .last = NULL, .lock = { PTHREAD_RWLOCK_INITIALIZER , NULL, {1, 0} } , }
static struct ast_acl_list *	ice_acl = NULL
static ast_rwlock_t	ice_acl_lock = AST_RWLOCK_INIT_VALUE
static int	icesupport = DEFAULT_ICESUPPORT
static struct ioqueues	ioqueues = { .first = NULL, .last = NULL, .lock = { PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP , NULL, {1, 0} } , }
static int	learning_min_duration = DEFAULT_LEARNING_MIN_DURATION
static int	learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL
static struct ast_sockaddr	lo6 = { .len = 0 }
static struct ast_threadstorage	pj_thread_storage = { .once = PTHREAD_ONCE_INIT , .key_init = __init_pj_thread_storage , .custom_init = NULL , }
static pj_pool_t *	pool
	Global memory pool for configuration and timers. More...
struct ast_srtp_res *	res_srtp
struct ast_srtp_policy_res *	res_srtp_policy
static struct ast_sockaddr	rtcpdebugaddr
static int	rtcpdebugport
static int	rtcpinterval = RTCP_DEFAULT_INTERVALMS
static int	rtcpstats
static struct ast_sockaddr	rtpdebugaddr
static int	rtpdebugport
static int	rtpend = DEFAULT_RTP_END
static int	rtpstart = DEFAULT_RTP_START
static int	srtp_replay_protection = DEFAULT_SRTP_REPLAY_PROTECTION
static int	strictrtp = DEFAULT_STRICT_RTP
static struct ast_acl_list *	stun_acl = NULL
static ast_rwlock_t	stun_acl_lock = AST_RWLOCK_INIT_VALUE
static int	stun_software_attribute = DEFAULT_STUN_SOFTWARE_ATTRIBUTE
static struct sockaddr_in	stunaddr
static pj_timer_heap_t *	timer_heap
	Global timer heap. More...
static int	timer_terminate
	Used to tell the timer thread to terminate. More...
static pj_thread_t *	timer_thread
	Thread executing the timer heap. More...
static pj_str_t	turnaddr
static pj_str_t	turnpassword
static int	turnport = DEFAULT_TURN_PORT
static pj_str_t	turnusername

Detailed Description

Supports RTP and RTCP with Symmetric RTP support for NAT traversal.

Author

Mark Spencer marks.nosp@m.ter@.nosp@m.digiu.nosp@m.m.co.nosp@m.m

Note

RTP is defined in RFC 3550.

Definition in file res_rtp_asterisk.c.

Macro Definition Documentation

CALC_LEARNING_MIN_DURATION

#define CALC_LEARNING_MIN_DURATION

(

count

)

(((count) - 1) * 9 - 5)

Calculate the min learning duration in ms.

The min supported packet size represents 10 ms and we need to account for some jitter and fast clocks while learning. Some messed up devices have very bad jitter for a small packet sample size. Jitter can also be introduced by the network itself.

So we'll allow packets to come in every 9ms on average for fast clocking with the last one coming in 5ms early for jitter.

Definition at line 154 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_DTLS_MTU

#define DEFAULT_DTLS_MTU   1200

Definition at line 188 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_DTMF_TIMEOUT

#define DEFAULT_DTMF_TIMEOUT   (150 * (8000 / 1000))

samples

Definition at line 137 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_ICESUPPORT

#define DEFAULT_ICESUPPORT   1

Definition at line 186 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_LEARNING_MIN_DURATION

#define DEFAULT_LEARNING_MIN_DURATION   CALC_LEARNING_MIN_DURATION(DEFAULT_LEARNING_MIN_SEQUENTIAL)

Definition at line 155 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_LEARNING_MIN_SEQUENTIAL

#define DEFAULT_LEARNING_MIN_SEQUENTIAL   4

Definition at line 141 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_RTP_END

#define DEFAULT_RTP_END   31000

Default maximum port number to end allocating RTP ports at

Definition at line 100 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_RTP_RECV_BUFFER_SIZE

#define DEFAULT_RTP_RECV_BUFFER_SIZE   20

The initial size of the RTP receiver buffer

Definition at line 111 of file res_rtp_asterisk.c.

Referenced by ast_rtp_prop_set().

DEFAULT_RTP_SEND_BUFFER_SIZE

#define DEFAULT_RTP_SEND_BUFFER_SIZE   250

The initial size of the RTP send buffer

Definition at line 109 of file res_rtp_asterisk.c.

Referenced by ast_rtp_prop_set().

DEFAULT_RTP_START

#define DEFAULT_RTP_START   5000

Default port number to start allocating RTP ports from

Definition at line 99 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_SRTP_REPLAY_PROTECTION

#define DEFAULT_SRTP_REPLAY_PROTECTION   1

Definition at line 185 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_STRICT_RTP

#define DEFAULT_STRICT_RTP   STRICT_RTP_YES

Enabled by default

Definition at line 184 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_STUN_SOFTWARE_ATTRIBUTE

#define DEFAULT_STUN_SOFTWARE_ATTRIBUTE   1

Definition at line 187 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DEFAULT_TURN_PORT

#define DEFAULT_TURN_PORT   3478

Definition at line 105 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

DTMF_SAMPLE_RATE_MS

#define DTMF_SAMPLE_RATE_MS   8

DTMF samples per millisecond

Definition at line 135 of file res_rtp_asterisk.c.

Referenced by ast_rtp_dtmf_begin(), ast_rtp_dtmf_continuation(), and ast_rtp_dtmf_end_with_duration().

FLAG_3389_WARNING

#define FLAG_3389_WARNING   (1 << 0)

Definition at line 283 of file res_rtp_asterisk.c.

Referenced by process_cn_rfc3389().

FLAG_DTMF_COMPENSATE

#define FLAG_DTMF_COMPENSATE   (1 << 4)

Definition at line 288 of file res_rtp_asterisk.c.

FLAG_NAT_ACTIVE

#define FLAG_NAT_ACTIVE   (3 << 1)

Definition at line 284 of file res_rtp_asterisk.c.

Referenced by ast_rtp_read(), bridge_p2p_rtp_write(), and rtp_raw_write().

FLAG_NAT_INACTIVE

#define FLAG_NAT_INACTIVE   (0 << 1)

Definition at line 285 of file res_rtp_asterisk.c.

Referenced by bridge_p2p_rtp_write(), and rtp_raw_write().

FLAG_NAT_INACTIVE_NOWARN

#define FLAG_NAT_INACTIVE_NOWARN   (1 << 1)

Definition at line 286 of file res_rtp_asterisk.c.

Referenced by bridge_p2p_rtp_write(), and rtp_raw_write().

FLAG_NEED_MARKER_BIT

#define FLAG_NEED_MARKER_BIT   (1 << 3)

Definition at line 287 of file res_rtp_asterisk.c.

Referenced by ast_rtp_change_source(), ast_rtp_local_bridge(), ast_rtp_stop(), ast_rtp_update_source(), bridge_p2p_rtp_write(), and rtp_raw_write().

FLAG_REQ_LOCAL_BRIDGE_BIT

#define FLAG_REQ_LOCAL_BRIDGE_BIT   (1 << 5)

Definition at line 289 of file res_rtp_asterisk.c.

Referenced by ast_rtp_local_bridge(), and bridge_p2p_rtp_write().

MAX_TIMESTAMP_SKEW

#define MAX_TIMESTAMP_SKEW   640

Definition at line 92 of file res_rtp_asterisk.c.

Referenced by rtp_raw_write().

MAXIMUM_RTP_PORT

#define MAXIMUM_RTP_PORT   65535

Maximum port number to accept

Definition at line 103 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

MAXIMUM_RTP_RECV_BUFFER_SIZE

#define MAXIMUM_RTP_RECV_BUFFER_SIZE   (DEFAULT_RTP_RECV_BUFFER_SIZE + 20)

Maximum RTP receive buffer size

Definition at line 112 of file res_rtp_asterisk.c.

Referenced by ast_rtp_read().

MAXIMUM_RTP_SEND_BUFFER_SIZE

#define MAXIMUM_RTP_SEND_BUFFER_SIZE   (DEFAULT_RTP_SEND_BUFFER_SIZE + 200)

Maximum RTP send buffer size

Definition at line 110 of file res_rtp_asterisk.c.

Referenced by ast_rtp_rtcp_handle_nack().

MINIMUM_RTP_PORT

#define MINIMUM_RTP_PORT   1024

Minimum port number to accept

Definition at line 102 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

MISSING_SEQNOS_ADDED_TRIGGER

#define MISSING_SEQNOS_ADDED_TRIGGER   2

The number of immediate missing packets that will trigger an immediate NACK

Definition at line 114 of file res_rtp_asterisk.c.

Referenced by ast_rtp_read().

OLD_PACKET_COUNT

#define OLD_PACKET_COUNT   1000

The number of previous packets that are considered old

Definition at line 113 of file res_rtp_asterisk.c.

Referenced by ast_rtp_read().

RTCP_DEFAULT_INTERVALMS

#define RTCP_DEFAULT_INTERVALMS   5000

Default milli-seconds between RTCP reports we send

Definition at line 95 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

RTCP_FB_NACK_BLOCK_WORD_LENGTH

#define RTCP_FB_NACK_BLOCK_WORD_LENGTH   2

Definition at line 6098 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_FB_REMB_BLOCK_WORD_LENGTH

#define RTCP_FB_REMB_BLOCK_WORD_LENGTH   4

Definition at line 6097 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_HEADER_SSRC_LENGTH

#define RTCP_HEADER_SSRC_LENGTH   2

Definition at line 6096 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_LENGTH_MASK

#define RTCP_LENGTH_MASK   0xFFFF

Definition at line 6061 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_LENGTH_SHIFT

#define RTCP_LENGTH_SHIFT   0

Definition at line 6070 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_MAX_INTERVALMS

#define RTCP_MAX_INTERVALMS   60000

Max milli-seconds between RTCP reports we send

Definition at line 97 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

RTCP_MIN_INTERVALMS

#define RTCP_MIN_INTERVALMS   500

Min milli-seconds between RTCP reports we send

Definition at line 96 of file res_rtp_asterisk.c.

Referenced by rtp_reload().

RTCP_PADDING_MASK

#define RTCP_PADDING_MASK   0x01

Definition at line 6064 of file res_rtp_asterisk.c.

RTCP_PADDING_SHIFT

#define RTCP_PADDING_SHIFT   29

Definition at line 6073 of file res_rtp_asterisk.c.

RTCP_PAYLOAD_TYPE_MASK

#define RTCP_PAYLOAD_TYPE_MASK   0xFF

Definition at line 6062 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_PAYLOAD_TYPE_SHIFT

#define RTCP_PAYLOAD_TYPE_SHIFT   16

Definition at line 6071 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_PT_APP

#define RTCP_PT_APP   204

Application defined (From RFC3550)

Definition at line 129 of file res_rtp_asterisk.c.

RTCP_PT_BYE

#define RTCP_PT_BYE   203

Goodbye (To remove SSRC's from tables) (From RFC3550)

Definition at line 127 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret(), and rtcp_payload_type2str().

RTCP_PT_FUR

#define RTCP_PT_FUR   192

Full INTRA-frame Request / Fast Update Request (From RFC2032)

Definition at line 119 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret(), and rtcp_payload_type2str().

RTCP_PT_PSFB

#define RTCP_PT_PSFB   AST_RTP_RTCP_PSFB

Payload Specific Feed Back (From RFC4585 also RFC5104)

Definition at line 132 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret(), rtcp_payload_subtype2str(), rtcp_payload_type2str(), rtp_write_rtcp_fir(), and rtp_write_rtcp_psfb().

RTCP_PT_RR

#define RTCP_PT_RR   AST_RTP_RTCP_RR

Receiver Report (From RFC3550)

Definition at line 123 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_generate_report(), ast_rtcp_interpret(), and rtcp_payload_type2str().

RTCP_PT_SDES

#define RTCP_PT_SDES   202

Source Description (From RFC3550)

Definition at line 125 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_generate_sdes(), ast_rtcp_interpret(), and rtcp_payload_type2str().

RTCP_PT_SR

#define RTCP_PT_SR   AST_RTP_RTCP_SR

Sender Report (From RFC3550)

Definition at line 121 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_generate_report(), ast_rtcp_interpret(), and rtcp_payload_type2str().

RTCP_REPORT_COUNT_MASK

#define RTCP_REPORT_COUNT_MASK   0x1F

Definition at line 6063 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_REPORT_COUNT_SHIFT

#define RTCP_REPORT_COUNT_SHIFT   24

Definition at line 6072 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_RR_BLOCK_WORD_LENGTH

#define RTCP_RR_BLOCK_WORD_LENGTH   6

Definition at line 6095 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_SR_BLOCK_WORD_LENGTH

#define RTCP_SR_BLOCK_WORD_LENGTH   5

Definition at line 6094 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_VALID_MASK

#define RTCP_VALID_MASK   (RTCP_VERSION_MASK_SHIFTED | (((RTCP_PAYLOAD_TYPE_MASK & ~0x1)) << RTCP_PAYLOAD_TYPE_SHIFT))

Definition at line 6091 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_VALID_VALUE

#define RTCP_VALID_VALUE   (RTCP_VERSION_SHIFTED | (RTCP_PT_SR << RTCP_PAYLOAD_TYPE_SHIFT))

Definition at line 6092 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_VERSION

#define RTCP_VERSION   2U

Definition at line 6076 of file res_rtp_asterisk.c.

RTCP_VERSION_MASK

#define RTCP_VERSION_MASK   0x03

Definition at line 6065 of file res_rtp_asterisk.c.

RTCP_VERSION_MASK_SHIFTED

#define RTCP_VERSION_MASK_SHIFTED   (RTCP_VERSION_MASK << RTCP_VERSION_SHIFT)

Definition at line 6078 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTCP_VERSION_SHIFT

#define RTCP_VERSION_SHIFT   30

Definition at line 6074 of file res_rtp_asterisk.c.

RTCP_VERSION_SHIFTED

#define RTCP_VERSION_SHIFTED   (RTCP_VERSION << RTCP_VERSION_SHIFT)

Definition at line 6077 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_interpret().

RTP_DTLS_ESTABLISHED

#define RTP_DTLS_ESTABLISHED   -37

Definition at line 161 of file res_rtp_asterisk.c.

Referenced by __rtp_recvfrom(), ast_rtcp_read(), and ast_rtp_read().

RTP_MTU

#define RTP_MTU   1200

Definition at line 134 of file res_rtp_asterisk.c.

RTP_SEQ_MOD

#define RTP_SEQ_MOD   (1<<16)

A sequence number can't be more than 16 bits

Definition at line 94 of file res_rtp_asterisk.c.

Referenced by ast_rtp_interpret(), and rtp_instance_parse_transport_wide_cc().

SEQNO_CYCLE_OVER

#define SEQNO_CYCLE_OVER   65536

The number after the maximum allowed sequence number

Definition at line 116 of file res_rtp_asterisk.c.

Referenced by ast_rtcp_generate_nack(), and ast_rtp_read().

SRTP_MASTER_KEY_LEN

#define SRTP_MASTER_KEY_LEN   16

Definition at line 157 of file res_rtp_asterisk.c.

Referenced by res_sdp_crypto_alloc(), and rtcp_debug_test_addr().

SRTP_MASTER_LEN

#define SRTP_MASTER_LEN   (SRTP_MASTER_KEY_LEN + SRTP_MASTER_SALT_LEN)

Definition at line 159 of file res_rtp_asterisk.c.

Referenced by rtcp_debug_test_addr().

SRTP_MASTER_SALT_LEN

#define SRTP_MASTER_SALT_LEN   14

Definition at line 158 of file res_rtp_asterisk.c.

Referenced by rtcp_debug_test_addr().

SSRC_MAPPING_ELEM_CMP

#define SSRC_MAPPING_ELEM_CMP	(		elem,
			value
	)		((elem).instance == (value))

SSRC mapping comparator for AST_VECTOR_REMOVE_CMP_UNORDERED()

Parameters

elem	Element to compare against
value	Value to compare with the vector element.

Returns

0 if element does not match.

Non-zero if element matches.

Definition at line 4045 of file res_rtp_asterisk.c.

Referenced by ast_rtp_bundle(), and ast_rtp_destroy().

STRICT_RTP_LEARN_TIMEOUT

#define STRICT_RTP_LEARN_TIMEOUT   5000

Strict RTP learning timeout time in milliseconds.

Note

Set to 5 seconds to allow reinvite chains for direct media to settle before media actually starts to arrive. There may be a reinvite collision involved on the other leg.

Definition at line 182 of file res_rtp_asterisk.c.

Referenced by ast_rtp_read().

TRANSPORT_SOCKET_RTCP

#define TRANSPORT_SOCKET_RTCP   1

Definition at line 292 of file res_rtp_asterisk.c.

Referenced by __rtp_recvfrom(), ast_rtp_on_ice_rx_data(), ast_rtp_on_ice_tx_pkt(), ast_rtp_prop_set(), and ice_create().

TRANSPORT_SOCKET_RTP

#define TRANSPORT_SOCKET_RTP   0

Definition at line 291 of file res_rtp_asterisk.c.

Referenced by __rtp_recvfrom(), ast_rtp_on_ice_rx_data(), ast_rtp_on_ice_tx_pkt(), and ice_create().

TRANSPORT_TURN_RTCP

#define TRANSPORT_TURN_RTCP   3

Definition at line 294 of file res_rtp_asterisk.c.

Referenced by ast_rtp_ice_turn_request(), ast_rtp_on_ice_rx_data(), ast_rtp_on_ice_tx_pkt(), and ast_rtp_on_turn_rx_rtcp_data().

TRANSPORT_TURN_RTP

#define TRANSPORT_TURN_RTP   2

Definition at line 293 of file res_rtp_asterisk.c.

Referenced by ast_rtp_ice_turn_request(), ast_rtp_on_ice_rx_data(), ast_rtp_on_ice_tx_pkt(), and ast_rtp_on_turn_rx_rtp_data().

TURN_STATE_WAIT_TIME

#define TURN_STATE_WAIT_TIME   2000

Definition at line 107 of file res_rtp_asterisk.c.

Referenced by ast_rtp_ice_turn_request(), ice_reset_session(), and rtp_deallocate_transport().

ZFONE_PROFILE_ID

#define ZFONE_PROFILE_ID   0x505a

Definition at line 139 of file res_rtp_asterisk.c.

Enumeration Type Documentation

strict_rtp_mode

enum strict_rtp_mode

Enumerator
STRICT_RTP_NO
STRICT_RTP_YES	Don't adhere to any strict RTP rules
STRICT_RTP_SEQNO	Strict RTP that restricts packets based on time and sequence number

Definition at line 169 of file res_rtp_asterisk.c.

 {
 STRICT_RTP_NO = 0, /*! Don't adhere to any strict RTP rules */
 STRICT_RTP_YES, /*! Strict RTP that restricts packets based on time and sequence number */
 STRICT_RTP_SEQNO, /*! Strict RTP that restricts packets based on sequence number */
};

strict_rtp_state

enum strict_rtp_state

Enumerator
STRICT_RTP_OPEN
STRICT_RTP_LEARN	No RTP packets should be dropped, all sources accepted
STRICT_RTP_CLOSED	Accept next packet as source

Definition at line 163 of file res_rtp_asterisk.c.

 {
 STRICT_RTP_OPEN = 0, /*! No RTP packets should be dropped, all sources accepted */
 STRICT_RTP_LEARN, /*! Accept next packet as source */
 STRICT_RTP_CLOSED, /*! Drop all RTP packets not coming from source that was learned */
};

Function Documentation

__init_pj_thread_storage()

static void __init_pj_thread_storage ( void  )

static

Definition at line 862 of file res_rtp_asterisk.c.

{

__reg_module()

static void __reg_module ( void  )

static

Definition at line 9742 of file res_rtp_asterisk.c.

__rtp_find_instance_by_ssrc()

static struct ast_rtp_instance* __rtp_find_instance_by_ssrc ( struct ast_rtp_instance *  instance, struct ast_rtp *  rtp, unsigned int  ssrc, int  source  )

static

Precondition

instance is locked

Definition at line 5869 of file res_rtp_asterisk.c.

References ast_rtp_get_ssrc(), AST_VECTOR_GET_ADDR, AST_VECTOR_SIZE, rtp_ssrc_mapping::instance, NULL, rtp_ssrc_mapping::ssrc, rtp_ssrc_mapping::ssrc_valid, ast_rtp::themssrc, and ast_rtp::themssrc_valid.

Referenced by rtp_find_instance_by_media_source_ssrc(), and rtp_find_instance_by_packet_source_ssrc().

{
 int index;

 if (!AST_VECTOR_SIZE(&rtp->ssrc_mapping)) {
 /* This instance is not bundled */
 return instance;
 }

 /* Find the bundled child instance */
 for (index = 0; index < AST_VECTOR_SIZE(&rtp->ssrc_mapping); ++index) {
 struct rtp_ssrc_mapping *mapping = AST_VECTOR_GET_ADDR(&rtp->ssrc_mapping, index);
 unsigned int mapping_ssrc = source ? ast_rtp_get_ssrc(mapping->instance) : mapping->ssrc;

 if (mapping->ssrc_valid && mapping_ssrc == ssrc) {
 return mapping->instance;
 }
 }

 /* Does the SSRC match the bundled parent? */
 if (rtp->themssrc_valid && rtp->themssrc == ssrc) {
 return instance;
 }
 return NULL;
}

__rtp_recvfrom()

static int __rtp_recvfrom ( struct ast_rtp_instance *  instance, void *  buf, size_t  size, int  flags, struct ast_sockaddr *  sa, int  rtcp  )

static

Precondition

instance is locked

Definition at line 3084 of file res_rtp_asterisk.c.

References ao2_lock, ao2_ref, ao2_unlock, ast_debug_dtls, ast_log, ast_recvfrom(), AST_RTP_DTLS_CONNECTION_EXISTING, AST_RTP_DTLS_SETUP_ACTPASS, AST_RTP_DTLS_SETUP_PASSIVE, AST_RTP_ICE_COMPONENT_RTCP, AST_RTP_ICE_COMPONENT_RTP, ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_rtp_instance_get_test(), ast_rtp_instance_set_remote_address, ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), buf, error(), in, len(), LOG_ERROR, LOG_WARNING, ast_rtp_engine_test::packets_to_drop, pj_thread_register_check(), ice_wrap::real_ice, RTP_DTLS_ESTABLISHED, ast_rtp::s, status, TRANSPORT_SOCKET_RTCP, and TRANSPORT_SOCKET_RTP.

Referenced by rtcp_recvfrom(), and rtp_recvfrom().

{
 int len;
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
 char *in = buf;
#endif
#ifdef HAVE_PJPROJECT
 struct ast_sockaddr *loop = rtcp ? &rtp->rtcp_loop : &rtp->rtp_loop;
#endif
#ifdef TEST_FRAMEWORK
 struct ast_rtp_engine_test *test = ast_rtp_instance_get_test(instance);
#endif

 if ((len = ast_recvfrom(rtcp ? rtp->rtcp->s : rtp->s, buf, size, flags, sa)) < 0) {
 return len;
 }

#ifdef TEST_FRAMEWORK
 if (test && test->packets_to_drop > 0) {
 test->packets_to_drop--;
 return 0;
 }
#endif

#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_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;
 }

 ast_debug_dtls(3, "(%p) DTLS - __rtp_recvfrom rtp=%p - Got SSL packet '%d'\n", instance, rtp, *in);

 /*
 * A race condition is prevented between dtls_perform_handshake()
 * and this function because both functions have to get the
 * instance lock before they can do anything. The
 * dtls_perform_handshake() function needs to start the timer
 * before we stop it below.
 */

 /* Before we feed data into OpenSSL ensure that the timeout timer is either stopped or completed */
 ao2_unlock(instance);
 dtls_srtp_stop_timeout_timer(instance, rtp, rtcp);
 ao2_lock(instance);

 /* 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);
 }

 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;
 }

 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;
 /* Use the keying material to set up key/salt information */
 if ((res = dtls_srtp_setup(rtp, instance, rtcp))) {
 return res;
 }
 /* Notify that dtls has been established */
 res = RTP_DTLS_ESTABLISHED;

 ast_debug_dtls(3, "(%p) DTLS - __rtp_recvfrom rtp=%p - established'\n", instance, rtp);
 } 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;
 struct ice_wrap *ice;

 pj_thread_register_check();

 pj_sockaddr_parse(pj_AF_UNSPEC(), 0, &combined, &address);

 /* Release the instance lock to avoid deadlock with PJPROJECT group lock */
 ice = rtp->ice;
 ao2_ref(ice, +1);
 ao2_unlock(instance);
 status = pj_ice_sess_on_rx_pkt(ice->real_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));
 ao2_ref(ice, -1);
 ao2_lock(instance);
 if (status != PJ_SUCCESS) {
 char err_buf[100];

 pj_strerror(status, err_buf, sizeof(err_buf));
 ast_log(LOG_WARNING, "PJ ICE Rx error status code: %d '%s'.\n",
 (int)status, err_buf);
 return -1;
 }
 if (!rtp->passthrough) {
 /* If a unidirectional ICE negotiation occurs then lock on to the source of the
 * ICE traffic and use it as the target. This will occur if the remote side only
 * wants to receive media but never send to us.
 */
 if (!rtp->ice_active_remote_candidates && !rtp->ice_proposed_remote_candidates) {
 if (rtcp) {
 ast_sockaddr_copy(&rtp->rtcp->them, sa);
 } else {
 ast_rtp_instance_set_remote_address(instance, sa);
 }
 }
 return 0;
 }
 rtp->passthrough = 0;
 }
#endif

 return len;
}

__rtp_sendto()

static int __rtp_sendto ( struct ast_rtp_instance *  instance, void *  buf, size_t  size, int  flags, struct ast_sockaddr *  sa, int  rtcp, int *  via_ice, int  use_srtp  )

static

Precondition

instance is locked

Definition at line 3247 of file res_rtp_asterisk.c.

References ao2_lock, ao2_ref, ao2_unlock, AST_RTP_ICE_COMPONENT_RTCP, AST_RTP_ICE_COMPONENT_RTP, ast_rtp_instance_get_data(), ast_rtp_instance_get_srtp(), ast_rtp_instance_set_last_tx(), ast_sendto(), buf, len(), NULL, pj_thread_register_check(), ast_srtp_res::protect, ice_wrap::real_ice, ast_rtp::s, and status.

Referenced by rtcp_sendto(), and rtp_sendto().

{
 int len = size;
 void *temp = buf;
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 struct ast_rtp_instance *transport = rtp->bundled ? rtp->bundled : instance;
 struct ast_rtp *transport_rtp = ast_rtp_instance_get_data(transport);
 struct ast_srtp *srtp = ast_rtp_instance_get_srtp(transport, rtcp);
 int res;

 *via_ice = 0;

 if (use_srtp && res_srtp && srtp && res_srtp->protect(srtp, &temp, &len, rtcp) < 0) {
 return -1;
 }

#ifdef HAVE_PJPROJECT
 if (transport_rtp->ice) {
 enum ast_rtp_ice_component_type component = rtcp ? AST_RTP_ICE_COMPONENT_RTCP : AST_RTP_ICE_COMPONENT_RTP;
 pj_status_t status;
 struct ice_wrap *ice;

 /* If RTCP is sharing the same socket then use the same component */
 if (rtcp && rtp->rtcp->s == rtp->s) {
 component = AST_RTP_ICE_COMPONENT_RTP;
 }

 pj_thread_register_check();

 /* Release the instance lock to avoid deadlock with PJPROJECT group lock */
 ice = transport_rtp->ice;
 ao2_ref(ice, +1);
 if (instance == transport) {
 ao2_unlock(instance);
 }
 status = pj_ice_sess_send_data(ice->real_ice, component, temp, len);
 ao2_ref(ice, -1);
 if (instance == transport) {
 ao2_lock(instance);
 }
 if (status == PJ_SUCCESS) {
 *via_ice = 1;
 return len;
 }
 }
#endif

 res = ast_sendto(rtcp ? transport_rtp->rtcp->s : transport_rtp->s, temp, len, flags, sa);
 if (res > 0) {
 ast_rtp_instance_set_last_tx(instance, time(NULL));
 }

 return res;
}

__unreg_module()

static void __unreg_module ( void  )

static

Definition at line 9742 of file res_rtp_asterisk.c.

acl_change_stasis_cb()

static void acl_change_stasis_cb ( void *  data, struct stasis_subscription *  sub, struct stasis_message *  message  )

static

Definition at line 9613 of file res_rtp_asterisk.c.

References ast_named_acl_change_type(), rtp_reload(), and stasis_message_type().

Referenced by rtp_learning_start().

{
 if (stasis_message_type(message) != ast_named_acl_change_type()) {
 return;
 }

 /* There is no simple way to just reload the ACLs, so just execute a forced reload. */
 rtp_reload(1, 1);
}

ast2pj_rtp_ice_role()

static void ast2pj_rtp_ice_role ( enum ast_rtp_ice_role  ast_role, enum pj_ice_sess_role *  pj_role  )

static

Definition at line 927 of file res_rtp_asterisk.c.

References AST_RTP_ICE_ROLE_CONTROLLED, and AST_RTP_ICE_ROLE_CONTROLLING.

Referenced by ice_reset_session().

{
 switch (ast_role) {
 case AST_RTP_ICE_ROLE_CONTROLLED:
 *pj_role = PJ_ICE_SESS_ROLE_CONTROLLED;
 break;
 case AST_RTP_ICE_ROLE_CONTROLLING:
 *pj_role = PJ_ICE_SESS_ROLE_CONTROLLING;
 break;
 }
}

AST_MODULE_SELF_SYM()

struct ast_module* AST_MODULE_SELF_SYM

(

void

)

Definition at line 9742 of file res_rtp_asterisk.c.

ast_rtcp_calc_interval()

static unsigned int ast_rtcp_calc_interval ( struct ast_rtp *  rtp )

static

Todo:

XXX Do a more reasonable calculation on this one Look in RFC 3550 Section A.7 for an example

Definition at line 3323 of file res_rtp_asterisk.c.

References rtcpinterval.

Referenced by ast_rtp_interpret(), and rtp_raw_write().

{
 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;
}

ast_rtcp_calculate_sr_rr_statistics()

static int ast_rtcp_calculate_sr_rr_statistics ( struct ast_rtp_instance *  instance, struct ast_rtp_rtcp_report *  rtcp_report, struct ast_sockaddr  remote_address, int  ice, int  sr  )

static

Definition at line 4579 of file res_rtp_asterisk.c.

References ast_json_pack(), ast_json_unref(), ast_rtp_get_rate(), ast_rtp_instance_get_data(), ast_rtp_publish_rtcp_message(), ast_rtp_rtcp_sent_type(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_verbose(), ast_rtp_rtcp_report_block::dlsr, ast_rtp::f, ast_frame_subclass::format, ast_rtp_rtcp_report_block::fraction, ast_rtp_rtcp_report_block::highest_seq_no, ast_rtp_rtcp_report_block::ia_jitter, ast_rtp_rtcp_report_block::lost_count, ast_rtp_rtcp_report_block::lsr, ast_rtp_rtcp_report::ntp_timestamp, NULL, ast_rtp_rtcp_report::octet_count, ast_rtp_rtcp_report::packet_count, ast_rtp_rtcp_report_block::packets, RAII_VAR, ast_rtp_rtcp_report::report_block, rtcp_debug_test_addr(), ast_rtp_rtcp_report::rtp_timestamp, ast_rtp_rtcp_report::sender_information, ast_rtp_rtcp_report_block::source_ssrc, ast_rtp_rtcp_report::ssrc, and ast_frame::subclass.

Referenced by ast_rtcp_write(), ast_rtp_read(), rtp_write_rtcp_fir(), and rtp_write_rtcp_psfb().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 struct ast_rtp_rtcp_report_block *report_block = NULL;
 RAII_VAR(struct ast_json *, message_blob, NULL, ast_json_unref);

 if (!rtp || !rtp->rtcp) {
 return 0;
 }

 if (ast_sockaddr_isnull(&rtp->rtcp->them)) {
 return 0;
 }

 if (!rtcp_report) {
 return -1;
 }

 report_block = rtcp_report->report_block[0];

 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.%06u\n",
 (unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_sec,
 (unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec);
 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 / ast_rtp_get_rate(rtp->f.subclass.format));
 ast_verbose(" Their last SR: %u\n", report_block->lsr);
 ast_verbose(" DLSR: %4.4f (sec)\n\n", (double)(report_block->dlsr / 65536.0));
 }
 }

 message_blob = ast_json_pack("{s: s, s: s}",
 "to", ast_sockaddr_stringify(&remote_address),
 "from", rtp->rtcp->local_addr_str);
 ast_rtp_publish_rtcp_message(instance, ast_rtp_rtcp_sent_type(),
 rtcp_report, message_blob);

 return 1;
}

ast_rtcp_generate_compound_prefix()

static int ast_rtcp_generate_compound_prefix ( struct ast_rtp_instance *  instance, unsigned char *  rtcpheader, struct ast_rtp_rtcp_report *  report, int *  sr  )

static

Definition at line 4699 of file res_rtp_asterisk.c.

References ast_debug_rtcp, ast_rtcp_generate_report(), and ast_rtcp_generate_sdes().

Referenced by ast_rtcp_write(), ast_rtp_read(), rtp_write_rtcp_fir(), and rtp_write_rtcp_psfb().

{
 int packet_len = 0;
 int res;

 /* Every RTCP packet needs to be sent out with a SR/RR and SDES prefixing it.
 * At the end of this function, rtcpheader should contain both of those packets,
 * and will return the length of the overall packet. This can be used to determine
 * where further packets can be inserted in the compound packet.
 */
 res = ast_rtcp_generate_report(instance, rtcpheader, report, sr);

 if (res == 0 || res == 1) {
 ast_debug_rtcp(1, "(%p) RTCP failed to generate %s report!\n", instance, sr ? "SR" : "RR");
 return 0;
 }

 packet_len += res;

 res = ast_rtcp_generate_sdes(instance, rtcpheader + packet_len, report);

 if (res == 0 || res == 1) {
 ast_debug_rtcp(1, "(%p) RTCP failed to generate SDES!\n", instance);
 return 0;
 }

 return packet_len + res;
}

ast_rtcp_generate_nack()

static int ast_rtcp_generate_nack ( struct ast_rtp_instance *  instance, unsigned char *  rtcpheader  )

static

Definition at line 4729 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_data(), AST_RTP_RTCP_FMT_NACK, AST_RTP_RTCP_RTPFB, ast_sockaddr_isnull(), AST_VECTOR_GET_CMP, AST_VECTOR_SIZE, ast_rtp::expectedrxseqno, find_by_value(), put_unaligned_uint32(), SEQNO_CYCLE_OVER, ast_rtp::ssrc, and ast_rtp::themssrc.

Referenced by ast_rtp_read().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 int packet_len;
 int blp_index = -1;
 int current_seqno;
 unsigned int fci = 0;
 size_t remaining_missing_seqno;

 if (!rtp || !rtp->rtcp) {
 return 0;
 }

 if (ast_sockaddr_isnull(&rtp->rtcp->them)) {
 return 0;
 }

 current_seqno = rtp->expectedrxseqno;
 remaining_missing_seqno = AST_VECTOR_SIZE(&rtp->missing_seqno);
 packet_len = 12; /* The header length is 12 (version line, packet source SSRC, media source SSRC) */

 /* If there are no missing sequence numbers then don't bother sending a NACK needlessly */
 if (!remaining_missing_seqno) {
 return 0;
 }

 /* This iterates through the possible forward sequence numbers seeing which ones we
 * have no packet for, adding it to the NACK until we are out of missing packets.
 */
 while (remaining_missing_seqno) {
 int *missing_seqno;

 /* On the first entry to this loop blp_index will be -1, so this will become 0
 * and the sequence number will be placed into the packet as the PID.
 */
 blp_index++;

 missing_seqno = AST_VECTOR_GET_CMP(&rtp->missing_seqno, current_seqno,
 find_by_value);
 if (missing_seqno) {
 /* We hit the max blp size, reset */
 if (blp_index >= 17) {
 put_unaligned_uint32(rtcpheader + packet_len, htonl(fci));
 fci = 0;
 blp_index = 0;
 packet_len += 4;
 }

 if (blp_index == 0) {
 fci |= (current_seqno << 16);
 } else {
 fci |= (1 << (blp_index - 1));
 }

 /* Since we've used a missing sequence number, we're down one */
 remaining_missing_seqno--;
 }

 /* Handle cycling of the sequence number */
 current_seqno++;
 if (current_seqno == SEQNO_CYCLE_OVER) {
 current_seqno = 0;
 }
 }

 put_unaligned_uint32(rtcpheader + packet_len, htonl(fci));
 packet_len += 4;

 /* Length MUST be 2+n, where n is the number of NACKs. Same as length in words minus 1 */
 put_unaligned_uint32(rtcpheader, htonl((2 << 30) | (AST_RTP_RTCP_FMT_NACK << 24)
 | (AST_RTP_RTCP_RTPFB << 16) | ((packet_len / 4) - 1)));
 put_unaligned_uint32(rtcpheader + 4, htonl(rtp->ssrc));
 put_unaligned_uint32(rtcpheader + 8, htonl(rtp->themssrc));

 return packet_len;
}

ast_rtcp_generate_report()

static int ast_rtcp_generate_report ( struct ast_rtp_instance *  instance, unsigned char *  rtcpheader, struct ast_rtp_rtcp_report *  rtcp_report, int *  sr  )

static

Definition at line 4491 of file res_rtp_asterisk.c.

References ast_calloc, ast_rtp_get_rate(), ast_rtp_instance_get_data(), ast_sockaddr_isnull(), ast_tvzero(), calculate_lost_packet_statistics(), ast_rtp_rtcp_report_block::dlsr, ast_rtp::f, ast_frame_subclass::format, ast_rtp_rtcp_report_block::fraction, ast_rtp_rtcp_report_block::highest_seq_no, ast_rtp_rtcp_report_block::ia_jitter, ast_rtp::lastrxseqno, ast_rtp::lastts, len(), ast_rtp_rtcp_report_block::lost_count, ast_rtp_rtcp_report_block::lsr, ast_rtp_rtcp_report::ntp_timestamp, NULL, ast_rtp_rtcp_report::octet_count, ast_rtp_rtcp_report::packet_count, ast_rtp_rtcp_report_block::packets, put_unaligned_uint32(), ast_rtp_rtcp_report::reception_report_count, ast_rtp_rtcp_report::report_block, RTCP_PT_RR, RTCP_PT_SR, ast_rtp_rtcp_report::rtp_timestamp, ast_rtp_rtcp_report::sender_information, ast_rtp_rtcp_report_block::source_ssrc, ast_rtp_rtcp_report::ssrc, ast_rtp::ssrc, ast_frame::subclass, ast_rtp::themssrc, ast_rtp::themssrc_valid, timersub(), timeval2ntp(), and ast_rtp_rtcp_report::type.

Referenced by ast_rtcp_generate_compound_prefix().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 int len = 0;
 struct timeval now;
 unsigned int now_lsw;
 unsigned int now_msw;
 unsigned int lost_packets;
 int fraction_lost;
 struct timeval dlsr = { 0, };
 struct ast_rtp_rtcp_report_block *report_block = NULL;

 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;
 }

 *sr = rtp->txcount > rtp->rtcp->lastsrtxcount ? 1 : 0;

 /* Compute statistics */
 calculate_lost_packet_statistics(rtp, &lost_packets, &fraction_lost);

 gettimeofday(&now, NULL);
 rtcp_report->reception_report_count = rtp->themssrc_valid ? 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_valid) {
 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 * ast_rtp_get_rate(rtp->f.subclass.format));
 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);
 put_unaligned_uint32(rtcpheader + 4, htonl(rtcp_report->ssrc)); /* Our SSRC */
 len += 8;
 if (*sr) {
 put_unaligned_uint32(rtcpheader + len, htonl(now_msw)); /* now, MSW. gettimeofday() + SEC_BETWEEN_1900_AND_1970 */
 put_unaligned_uint32(rtcpheader + len + 4, htonl(now_lsw)); /* now, LSW */
 put_unaligned_uint32(rtcpheader + len + 8, htonl(rtcp_report->sender_information.rtp_timestamp));
 put_unaligned_uint32(rtcpheader + len + 12, htonl(rtcp_report->sender_information.packet_count));
 put_unaligned_uint32(rtcpheader + len + 16, htonl(rtcp_report->sender_information.octet_count));
 len += 20;
 }
 if (report_block) {
 put_unaligned_uint32(rtcpheader + len, htonl(report_block->source_ssrc)); /* Their SSRC */
 put_unaligned_uint32(rtcpheader + len + 4, htonl((report_block->lost_count.fraction << 24) | report_block->lost_count.packets));
 put_unaligned_uint32(rtcpheader + len + 8, htonl(report_block->highest_seq_no));
 put_unaligned_uint32(rtcpheader + len + 12, htonl(report_block->ia_jitter));
 put_unaligned_uint32(rtcpheader + len + 16, htonl(report_block->lsr));
 put_unaligned_uint32(rtcpheader + len + 20, htonl(report_block->dlsr));
 len += 24;
 }

 put_unaligned_uint32(rtcpheader, htonl((2 << 30) | (rtcp_report->reception_report_count << 24)
 | ((*sr ? RTCP_PT_SR : RTCP_PT_RR) << 16) | ((len/4)-1)));

 return len;
}

ast_rtcp_generate_sdes()

static int ast_rtcp_generate_sdes ( struct ast_rtp_instance *  instance, unsigned char *  rtcpheader, struct ast_rtp_rtcp_report *  rtcp_report  )

static

Definition at line 4641 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_data(), ast_sockaddr_isnull(), AST_UUID_STR_LEN, ast_rtp::cname, len(), put_unaligned_uint32(), RTCP_PT_SDES, and ast_rtp_rtcp_report::ssrc.

Referenced by ast_rtcp_generate_compound_prefix().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 int len = 0;
 uint16_t sdes_packet_len_bytes;
 uint16_t sdes_packet_len_rounded;

 if (!rtp || !rtp->rtcp) {
 return 0;
 }

 if (ast_sockaddr_isnull(&rtp->rtcp->them)) {
 return 0;
 }

 if (!rtcp_report) {
 return -1;
 }

 sdes_packet_len_bytes =
 4 + /* RTCP Header */
 4 + /* SSRC */
 1 + /* Type (CNAME) */
 1 + /* Text Length */
 AST_UUID_STR_LEN /* Text and NULL terminator */
 ;

 /* Round to 32 bit boundary */
 sdes_packet_len_rounded = (sdes_packet_len_bytes + 3) & ~0x3;

 put_unaligned_uint32(rtcpheader, htonl((2 << 30) | (1 << 24) | (RTCP_PT_SDES << 16) | ((sdes_packet_len_rounded / 4) - 1)));
 put_unaligned_uint32(rtcpheader + 4, htonl(rtcp_report->ssrc));
 rtcpheader[8] = 0x01; /* CNAME */
 rtcpheader[9] = AST_UUID_STR_LEN - 1; /* Number of bytes of text */
 memcpy(rtcpheader + 10, rtp->cname, AST_UUID_STR_LEN);
 len += 10 + AST_UUID_STR_LEN;

 /* Padding - Note that we don't set the padded bit on the packet. From
 * RFC 3550 Section 6.5:
 *
 * No length octet follows the null item type octet, but additional null
 * octets MUST be included if needd to pad until the next 32-bit
 * boundary. Note that this padding is separate from that indicated by
 * the P bit in the RTCP header.
 *
 * These bytes will already be zeroed out during array initialization.
 */
 len += (sdes_packet_len_rounded - sdes_packet_len_bytes);

 return len;
}

ast_rtcp_interpret()

static struct ast_frame* ast_rtcp_interpret ( struct ast_rtp_instance *  instance, struct ast_srtp *  srtp, const unsigned char *  rtcpdata, size_t  size, struct ast_sockaddr *  addr  )

static

True if we have seen an acceptable SSRC to learn the remote RTCP address

True if the ssrc value we have is valid and not garbage because it doesn't exist.

Always use packet source SSRC to find the rtp instance unless explicitly told not to.

Definition at line 6100 of file res_rtp_asterisk.c.

References ao2_cleanup, ao2_lock, ao2_unlock, ast_calloc, AST_CONTROL_VIDUPDATE, ast_debug, ast_debug_rtcp, ast_debug_rtp_packet_is_allowed, AST_FRAME_CONTROL, AST_FRAME_RTCP, AST_FRIENDLY_OFFSET, ast_json_pack(), ast_json_unref(), ast_null_frame, ast_rtp_instance_get_data(), ast_rtp_instance_get_prop(), ast_rtp_instance_get_test(), AST_RTP_PROPERTY_NAT, AST_RTP_PROPERTY_REMB, ast_rtp_publish_rtcp_message(), AST_RTP_RTCP_FMT_FIR, AST_RTP_RTCP_FMT_NACK, AST_RTP_RTCP_FMT_PLI, AST_RTP_RTCP_FMT_REMB, ast_rtp_rtcp_handle_nack(), ast_rtp_rtcp_received_type(), ast_rtp_rtcp_report_alloc(), AST_RTP_RTCP_RTPFB, ast_sockaddr_cmp(), ast_sockaddr_copy(), ast_sockaddr_stringify(), ast_verbose(), ast_rtp_rtcp_feedback_remb::br_exp, ast_rtp_rtcp_feedback_remb::br_mantissa, ast_frame::data, ast_frame::datalen, ast_frame::delivery, ast_rtp_rtcp_report_block::dlsr, ast_rtp::f, ast_rtp_rtcp_feedback::fmt, ast_rtp_rtcp_report_block::fraction, ast_frame::frametype, ast_rtp_rtcp_report_block::highest_seq_no, ast_rtp_rtcp_report_block::ia_jitter, ast_frame_subclass::integer, ast_rtp_rtcp_report_block::lost_count, ast_rtp_rtcp_report_block::lsr, ast_frame::mallocd, ntp2timeval(), NULL, ast_frame::offset, ast_rtp_rtcp_report_block::packets, ast_frame::ptr, RAII_VAR, ast_rtp_rtcp_feedback::remb, ast_rtp_rtcp_report::report_block, rtcp_debug_test_addr(), RTCP_FB_NACK_BLOCK_WORD_LENGTH, RTCP_FB_REMB_BLOCK_WORD_LENGTH, RTCP_HEADER_SSRC_LENGTH, RTCP_LENGTH_MASK, RTCP_LENGTH_SHIFT, rtcp_payload_subtype2str(), rtcp_payload_type2str(), RTCP_PAYLOAD_TYPE_MASK, RTCP_PAYLOAD_TYPE_SHIFT, RTCP_PT_BYE, RTCP_PT_FUR, RTCP_PT_PSFB, RTCP_PT_RR, RTCP_PT_SDES, RTCP_PT_SR, RTCP_REPORT_COUNT_MASK, RTCP_REPORT_COUNT_SHIFT, RTCP_RR_BLOCK_WORD_LENGTH, RTCP_SR_BLOCK_WORD_LENGTH, RTCP_VALID_MASK, RTCP_VALID_VALUE, RTCP_VERSION_MASK_SHIFTED, RTCP_VERSION_SHIFTED, rtp_find_instance_by_media_source_ssrc(), rtp_find_instance_by_packet_source_ssrc(), ast_frame::samples, ast_rtp_engine_test::sdes_received, ast_rtp_rtcp_report_block::source_ssrc, ast_frame::src, srtp_replay_protection, ast_frame::stream_num, STRICT_RTP_OPEN, ast_frame::subclass, ast_rtp::themssrc, ast_rtp::themssrc_valid, timeval2ntp(), ast_srtp_res::unprotect, update_jitter_stats(), update_lost_stats(), and update_rtt_stats().

Referenced by ast_rtcp_read(), and ast_rtp_read().

{
 struct ast_rtp_instance *transport = instance;
 struct ast_rtp *transport_rtp = ast_rtp_instance_get_data(instance);
 int len = size;
 unsigned int *rtcpheader = (unsigned int *)(rtcpdata);
 unsigned int packetwords;
 unsigned int position;
 unsigned int first_word;
 /*! True if we have seen an acceptable SSRC to learn the remote RTCP address */
 unsigned int ssrc_seen;
 struct ast_rtp_rtcp_report_block *report_block;
 struct ast_frame *f = &ast_null_frame;
#ifdef TEST_FRAMEWORK
 struct ast_rtp_engine_test *test_engine;
#endif

 /* If this is encrypted then decrypt the payload */
 if ((*rtcpheader & 0xC0) && res_srtp && srtp && res_srtp->unprotect(
 srtp, rtcpheader, &len, 1 | (srtp_replay_protection << 1)) < 0) {
 return &ast_null_frame;
 }

 packetwords = len / 4;

 ast_debug_rtcp(1, "(%p) RTCP got report of %d bytes from %s\n",
 instance, len, ast_sockaddr_stringify(addr));

 /*
 * Validate the RTCP packet according to an adapted and slightly
 * modified RFC3550 validation algorithm.
 */
 if (packetwords < RTCP_HEADER_SSRC_LENGTH) {
 ast_debug_rtcp(1, "(%p) RTCP %p -- from %s: Frame size (%u words) is too short\n",
 instance, transport_rtp, ast_sockaddr_stringify(addr), packetwords);
 return &ast_null_frame;
 }
 position = 0;
 first_word = ntohl(rtcpheader[position]);
 if ((first_word & RTCP_VALID_MASK) != RTCP_VALID_VALUE) {
 ast_debug_rtcp(1, "(%p) RTCP %p -- from %s: Failed first packet validity check\n",
 instance, transport_rtp, ast_sockaddr_stringify(addr));
 return &ast_null_frame;
 }
 do {
 position += ((first_word >> RTCP_LENGTH_SHIFT) & RTCP_LENGTH_MASK) + 1;
 if (packetwords <= position) {
 break;
 }
 first_word = ntohl(rtcpheader[position]);
 } while ((first_word & RTCP_VERSION_MASK_SHIFTED) == RTCP_VERSION_SHIFTED);
 if (position != packetwords) {
 ast_debug_rtcp(1, "(%p) RTCP %p -- from %s: Failed packet version or length check\n",
 instance, transport_rtp, ast_sockaddr_stringify(addr));
 return &ast_null_frame;
 }

 /*
 * Note: RFC3605 points out that true NAT (vs NAPT) can cause RTCP
 * to have a different IP address and port than RTP. Otherwise, when
 * strictrtp is enabled we could reject RTCP packets not coming from
 * the learned RTP IP address if it is available.
 */

 /*
 * strictrtp safety needs SSRC to match before we use the
 * sender's address for symmetrical RTP to send our RTCP
 * reports.
 *
 * If strictrtp is not enabled then claim to have already seen
 * a matching SSRC so we'll accept this packet's address for
 * symmetrical RTP.
 */
 ssrc_seen = transport_rtp->strict_rtp_state == STRICT_RTP_OPEN;

 position = 0;
 while (position < packetwords) {
 unsigned int i;
 unsigned int pt;
 unsigned int rc;
 unsigned int ssrc;
 /*! True if the ssrc value we have is valid and not garbage because it doesn't exist. */
 unsigned int ssrc_valid;
 unsigned int length;
 unsigned int min_length;
 /*! Always use packet source SSRC to find the rtp instance unless explicitly told not to. */
 unsigned int use_packet_source = 1;

 struct ast_json *message_blob;
 RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report, NULL, ao2_cleanup);
 struct ast_rtp_instance *child;
 struct ast_rtp *rtp;
 struct ast_rtp_rtcp_feedback *feedback;

 i = position;
 first_word = ntohl(rtcpheader[i]);
 pt = (first_word >> RTCP_PAYLOAD_TYPE_SHIFT) & RTCP_PAYLOAD_TYPE_MASK;
 rc = (first_word >> RTCP_REPORT_COUNT_SHIFT) & RTCP_REPORT_COUNT_MASK;
 /* RFC3550 says 'length' is the number of words in the packet - 1 */
 length = ((first_word >> RTCP_LENGTH_SHIFT) & RTCP_LENGTH_MASK) + 1;

 /* Check expected RTCP packet record length */
 min_length = RTCP_HEADER_SSRC_LENGTH;
 switch (pt) {
 case RTCP_PT_SR:
 min_length += RTCP_SR_BLOCK_WORD_LENGTH;
 /* fall through */
 case RTCP_PT_RR:
 min_length += (rc * RTCP_RR_BLOCK_WORD_LENGTH);
 use_packet_source = 0;
 break;
 case RTCP_PT_FUR:
 break;
 case AST_RTP_RTCP_RTPFB:
 switch (rc) {
 case AST_RTP_RTCP_FMT_NACK:
 min_length += RTCP_FB_NACK_BLOCK_WORD_LENGTH;
 break;
 default:
 break;
 }
 use_packet_source = 0;
 break;
 case RTCP_PT_PSFB:
 switch (rc) {
 case AST_RTP_RTCP_FMT_REMB:
 min_length += RTCP_FB_REMB_BLOCK_WORD_LENGTH;
 break;
 default:
 break;
 }
 break;
 case RTCP_PT_SDES:
 case RTCP_PT_BYE:
 /*
 * There may not be a SSRC/CSRC present. The packet is
 * useless but still valid if it isn't present.
 *
 * We don't know what min_length should be so disable the check
 */
 min_length = length;
 break;
 default:
 ast_debug_rtcp(1, "(%p) RTCP %p -- from %s: %u(%s) skipping record\n",
 instance, transport_rtp, ast_sockaddr_stringify(addr), pt, rtcp_payload_type2str(pt));
 if (rtcp_debug_test_addr(addr)) {
 ast_verbose("\n");
 ast_verbose("RTCP from %s: %u(%s) skipping record\n",
 ast_sockaddr_stringify(addr), pt, rtcp_payload_type2str(pt));
 }
 position += length;
 continue;
 }
 if (length < min_length) {
 ast_debug_rtcp(1, "(%p) RTCP %p -- from %s: %u(%s) length field less than expected minimum. Min:%u Got:%u\n",
 instance, transport_rtp, ast_sockaddr_stringify(addr), pt, rtcp_payload_type2str(pt),
 min_length - 1, length - 1);
 return &ast_null_frame;
 }

 /* Get the RTCP record SSRC if defined for the record */
 ssrc_valid = 1;
 switch (pt) {
 case RTCP_PT_SR:
 case RTCP_PT_RR:
 rtcp_report = ast_rtp_rtcp_report_alloc(rc);
 if (!rtcp_report) {
 return &ast_null_frame;
 }
 rtcp_report->reception_report_count = rc;

 ssrc = ntohl(rtcpheader[i + 2]);
 rtcp_report->ssrc = ssrc;
 break;
 case RTCP_PT_FUR:
 case RTCP_PT_PSFB:
 ssrc = ntohl(rtcpheader[i + 1]);
 break;
 case AST_RTP_RTCP_RTPFB:
 ssrc = ntohl(rtcpheader[i + 2]);
 break;
 case RTCP_PT_SDES:
 case RTCP_PT_BYE:
 default:
 ssrc = 0;
 ssrc_valid = 0;
 break;
 }

 if (rtcp_debug_test_addr(addr)) {
 const char *subtype = rtcp_payload_subtype2str(pt, rc);

 ast_verbose("\n");
 ast_verbose("RTCP from %s\n", ast_sockaddr_stringify(addr));
 ast_verbose("PT: %u (%s)\n", pt, rtcp_payload_type2str(pt));
 if (subtype) {
 ast_verbose("Packet Subtype: %u (%s)\n", rc, subtype);
 } else {
 ast_verbose("Reception reports: %u\n", rc);
 }
 ast_verbose("SSRC of sender: %u\n", ssrc);
 }

 /* Determine the appropriate instance for this */
 if (ssrc_valid) {
 /*
 * Depending on the payload type, either the packet source or media source
 * SSRC is used.
 */
 if (use_packet_source) {
 child = rtp_find_instance_by_packet_source_ssrc(transport, transport_rtp, ssrc);
 } else {
 child = rtp_find_instance_by_media_source_ssrc(transport, transport_rtp, ssrc);
 }
 if (child && child != transport) {
 /*
 * It is safe to hold the child lock while holding the parent lock.
 * We guarantee that the locking order is always parent->child or
 * that the child lock is not held when acquiring the parent lock.
 */
 ao2_lock(child);
 instance = child;
 rtp = ast_rtp_instance_get_data(instance);
 } else {
 /* The child is the parent! We don't need to unlock it. */
 child = NULL;
 rtp = transport_rtp;
 }
 } else {
 child = NULL;
 rtp = transport_rtp;
 }

 if (ssrc_valid && rtp->themssrc_valid) {
 /*
 * If the SSRC is 1, we still need to handle RTCP since this could be a
 * special case. For example, if we have a unidirectional video stream, the
 * SSRC may be set to 1 by the browser (in the case of chromium), and requests
 * will still need to be processed so that video can flow as expected. This
 * should only be done for PLI and FUR, since there is not a way to get the
 * appropriate rtp instance when the SSRC is 1.
 */
 int exception = (ssrc == 1 && !((pt == RTCP_PT_PSFB && rc == AST_RTP_RTCP_FMT_PLI) || pt == RTCP_PT_FUR));
 if ((ssrc != rtp->themssrc && use_packet_source && ssrc != 1)
 || exception) {
 /*
 * Skip over this RTCP record as it does not contain the
 * correct SSRC. We should not act upon RTCP records
 * for a different stream.
 */
 position += length;
 ast_debug_rtcp(1, "(%p) RTCP %p -- from %s: Skipping record, received SSRC '%u' != expected '%u'\n",
 instance, rtp, ast_sockaddr_stringify(addr), ssrc, rtp->themssrc);
 if (child) {
 ao2_unlock(child);
 }
 continue;
 }
 ssrc_seen = 1;
 }

 if (ssrc_seen && 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 (ast_debug_rtp_packet_is_allowed) {
 ast_debug(0, "(%p) RTCP NAT: Got RTCP from other end. Now sending to address %s\n",
 instance, ast_sockaddr_stringify(addr));
 }
 }
 }

 i += RTCP_HEADER_SSRC_LENGTH; /* 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.%06u\n",
 (unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_sec,
 (unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec);
 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 += RTCP_SR_BLOCK_WORD_LENGTH;
 /* 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) {
 if (child) {
 ao2_unlock(child);
 }
 return &ast_null_frame;
 }
 rtcp_report->report_block[0] = 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);

 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);
 }
 }
 /* If and when we handle more than one report block, this should occur outside
 * this loop.
 */

 message_blob = ast_json_pack("{s: s, s: s, s: f}",
 "from", ast_sockaddr_stringify(addr),
 "to", transport_rtp->rtcp->local_addr_str,
 "rtt", rtp->rtcp->rtt);
 ast_rtp_publish_rtcp_message(instance, ast_rtp_rtcp_received_type(),
 rtcp_report,
 message_blob);
 ast_json_unref(message_blob);

 /* Return an AST_FRAME_RTCP frame with the ast_rtp_rtcp_report
 * object as a its data */
 transport_rtp->f.frametype = AST_FRAME_RTCP;
 transport_rtp->f.subclass.integer = pt;
 transport_rtp->f.data.ptr = rtp->rtcp->frame_buf + AST_FRIENDLY_OFFSET;
 memcpy(transport_rtp->f.data.ptr, rtcp_report, sizeof(struct ast_rtp_rtcp_report));
 transport_rtp->f.datalen = sizeof(struct ast_rtp_rtcp_report);
 if (rc > 0) {
 /* There's always a single report block stored, here */
 struct ast_rtp_rtcp_report *rtcp_report2;
 report_block = transport_rtp->f.data.ptr + transport_rtp->f.datalen + sizeof(struct ast_rtp_rtcp_report_block *);
 memcpy(report_block, rtcp_report->report_block[0], sizeof(struct ast_rtp_rtcp_report_block));
 rtcp_report2 = (struct ast_rtp_rtcp_report *)transport_rtp->f.data.ptr;
 rtcp_report2->report_block[0] = report_block;
 transport_rtp->f.datalen += sizeof(struct ast_rtp_rtcp_report_block);
 }
 transport_rtp->f.offset = AST_FRIENDLY_OFFSET;
 transport_rtp->f.samples = 0;
 transport_rtp->f.mallocd = 0;
 transport_rtp->f.delivery.tv_sec = 0;
 transport_rtp->f.delivery.tv_usec = 0;
 transport_rtp->f.src = "RTP";
 transport_rtp->f.stream_num = rtp->stream_num;
 f = &transport_rtp->f;
 break;
 case AST_RTP_RTCP_RTPFB:
 switch (rc) {
 case AST_RTP_RTCP_FMT_NACK:
 /* If retransmissions are not enabled ignore this message */
 if (!rtp->send_buffer) {
 break;
 }

 if (rtcp_debug_test_addr(addr)) {
 ast_verbose("Received generic RTCP NACK message\n");
 }

 ast_rtp_rtcp_handle_nack(instance, rtcpheader, position, length);
 break;
 default:
 break;
 }
 break;
 case RTCP_PT_FUR:
 /* Handle RTCP FUR as FIR by setting the format to 4 */
 rc = AST_RTP_RTCP_FMT_FIR;
 case RTCP_PT_PSFB:
 switch (rc) {
 case AST_RTP_RTCP_FMT_PLI:
 case AST_RTP_RTCP_FMT_FIR:
 if (rtcp_debug_test_addr(addr)) {
 ast_verbose("Received an RTCP Fast Update Request\n");
 }
 transport_rtp->f.frametype = AST_FRAME_CONTROL;
 transport_rtp->f.subclass.integer = AST_CONTROL_VIDUPDATE;
 transport_rtp->f.datalen = 0;
 transport_rtp->f.samples = 0;
 transport_rtp->f.mallocd = 0;
 transport_rtp->f.src = "RTP";
 f = &transport_rtp->f;
 break;
 case AST_RTP_RTCP_FMT_REMB:
 /* If REMB support is not enabled ignore this message */
 if (!ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_REMB)) {
 break;
 }

 if (rtcp_debug_test_addr(addr)) {
 ast_verbose("Received REMB report\n");
 }
 transport_rtp->f.frametype = AST_FRAME_RTCP;
 transport_rtp->f.subclass.integer = pt;
 transport_rtp->f.stream_num = rtp->stream_num;
 transport_rtp->f.data.ptr = rtp->rtcp->frame_buf + AST_FRIENDLY_OFFSET;
 feedback = transport_rtp->f.data.ptr;
 feedback->fmt = rc;

 /* We don't actually care about the SSRC information in the feedback message */
 first_word = ntohl(rtcpheader[i + 2]);
 feedback->remb.br_exp = (first_word >> 18) & ((1 << 6) - 1);
 feedback->remb.br_mantissa = first_word & ((1 << 18) - 1);

 transport_rtp->f.datalen = sizeof(struct ast_rtp_rtcp_feedback);
 transport_rtp->f.offset = AST_FRIENDLY_OFFSET;
 transport_rtp->f.samples = 0;
 transport_rtp->f.mallocd = 0;
 transport_rtp->f.delivery.tv_sec = 0;
 transport_rtp->f.delivery.tv_usec = 0;
 transport_rtp->f.src = "RTP";
 f = &transport_rtp->f;
 break;
 default:
 break;
 }
 break;
 case RTCP_PT_SDES:
 if (rtcp_debug_test_addr(addr)) {
 ast_verbose("Received an SDES from %s\n",
 ast_sockaddr_stringify(addr));
 }
#ifdef TEST_FRAMEWORK
 if ((test_engine = ast_rtp_instance_get_test(instance))) {
 test_engine->sdes_received = 1;
 }
#endif
 break;
 case RTCP_PT_BYE:
 if (rtcp_debug_test_addr(addr)) {
 ast_verbose("Received a BYE from %s\n",
 ast_sockaddr_stringify(addr));
 }
 break;
 default:
 break;
 }
 position += length;
 rtp->rtcp->rtcp_info = 1;

 if (child) {
 ao2_unlock(child);
 }
 }

 return f;
}

ast_rtcp_read()

static struct ast_frame* ast_rtcp_read ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 6590 of file res_rtp_asterisk.c.

References ast_assert, AST_CONTROL_SRCCHANGE, ast_debug_stun, AST_FRAME_CONTROL, AST_FRIENDLY_OFFSET, ast_log, ast_null_frame, ast_rtcp_interpret(), ast_rtp_instance_get_data(), ast_rtp_instance_get_srtp(), ast_sockaddr_copy(), ast_sockaddr_from_sin, ast_sockaddr_ipv4_mapped(), ast_sockaddr_is_ipv4(), ast_sockaddr_stringify(), ast_sockaddr_to_sin, AST_STUN_ACCEPT, ast_stun_handle_packet(), errno, ast_rtp::f, ast_frame::frametype, ast_frame_subclass::integer, LOG_WARNING, NULL, rtcp_recvfrom(), RTP_DTLS_ESTABLISHED, ast_rtp::s, and ast_frame::subclass.

Referenced by ast_rtp_read().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance, 1);
 struct ast_sockaddr addr;
 unsigned char rtcpdata[8192 + AST_FRIENDLY_OFFSET];
 unsigned char *read_area = rtcpdata + AST_FRIENDLY_OFFSET;
 size_t read_area_size = sizeof(rtcpdata) - AST_FRIENDLY_OFFSET;
 int res;

 /* Read in RTCP data from the socket */
 if ((res = rtcp_recvfrom(instance, read_area, read_area_size,
 0, &addr)) < 0) {
 if (res == RTP_DTLS_ESTABLISHED) {
 rtp->f.frametype = AST_FRAME_CONTROL;
 rtp->f.subclass.integer = AST_CONTROL_SRCCHANGE;
 return &rtp->f;
 }

 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 (!*read_area) {
 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_stun(2, "(%p) STUN using IPv6 mapped address %s\n",
 instance, ast_sockaddr_stringify(&addr));
 ast_sockaddr_to_sin(&addr_v4, &addr_tmp);
 } else {
 ast_debug_stun(2, "(%p) STUN cannot do for non IPv4 address %s\n",
 instance, ast_sockaddr_stringify(&addr));
 return &ast_null_frame;
 }
 if ((ast_stun_handle_packet(rtp->rtcp->s, &addr_tmp, read_area, res, NULL, NULL) == AST_STUN_ACCEPT)) {
 ast_sockaddr_from_sin(&addr, &addr_tmp);
 ast_sockaddr_copy(&rtp->rtcp->them, &addr);
 }
 return &ast_null_frame;
 }

 return ast_rtcp_interpret(instance, srtp, read_area, res, &addr);
}

ast_rtcp_write()

static int ast_rtcp_write ( const void *  data )

static

Write a 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

Scheduler callback

Definition at line 4814 of file res_rtp_asterisk.c.

References ao2_cleanup, ao2_lock, ao2_ref, ao2_unlock, ast_log, ast_rtcp_calculate_sr_rr_statistics(), ast_rtcp_generate_compound_prefix(), ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_rtp_rtcp_report_alloc(), ast_sockaddr_copy(), ast_sockaddr_stringify(), AST_UUID_STR_LEN, cleanup(), errno, LOG_ERROR, RAII_VAR, rtcp_sendto(), and ast_rtp::themssrc_valid.

Referenced by ast_rtp_interpret(), and rtp_raw_write().

{
 struct ast_rtp_instance *instance = (struct ast_rtp_instance *) data;
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 int res;
 int sr = 0;
 int packet_len = 0;
 int ice;
 struct ast_sockaddr remote_address = { { 0, } };
 unsigned char *rtcpheader;
 unsigned char bdata[AST_UUID_STR_LEN + 128] = ""; /* More than enough */
 RAII_VAR(struct ast_rtp_rtcp_report *, rtcp_report,
 ast_rtp_rtcp_report_alloc(rtp->themssrc_valid ? 1 : 0),
 ao2_cleanup);

 if (!rtp || !rtp->rtcp || rtp->rtcp->schedid == -1) {
 ao2_ref(instance, -1);
 return 0;
 }

 ao2_lock(instance);
 rtcpheader = bdata;

 res = ast_rtcp_generate_compound_prefix(instance, rtcpheader, rtcp_report, &sr);

 if (res == 0 || res == 1) {
 goto cleanup;
 }

 packet_len += res;

 if (rtp->bundled) {
 ast_rtp_instance_get_remote_address(instance, &remote_address);
 } else {
 ast_sockaddr_copy(&remote_address, &rtp->rtcp->them);
 }

 res = rtcp_sendto(instance, (unsigned int *)rtcpheader, packet_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));
 res = 0;
 } else {
 ast_rtcp_calculate_sr_rr_statistics(instance, rtcp_report, remote_address, ice, sr);
 }

cleanup:
 ao2_unlock(instance);

 if (!res) {
 /*
 * Not being rescheduled.
 */
 rtp->rtcp->schedid = -1;
 ao2_ref(instance, -1);
 }

 return res;
}

ast_rtp_bundle()

static int ast_rtp_bundle ( struct ast_rtp_instance *  child, struct ast_rtp_instance *  parent  )

static

Precondition

child is locked

Definition at line 8938 of file res_rtp_asterisk.c.

References ao2_bump, ao2_lock, ao2_ref, ao2_unlock, ast_debug_dtls, AST_RTP_DTLS_CONNECTION_EXISTING, ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, AST_RTP_INSTANCE_RTCP_STANDARD, ast_rtp_instance_set_remote_address, AST_VECTOR_APPEND, AST_VECTOR_ELEM_CLEANUP_NOOP, AST_VECTOR_REMOVE_CMP_UNORDERED, rtp_ssrc_mapping::instance, NULL, rtp_allocate_transport(), rtp_deallocate_transport(), rtp_ssrc_mapping::ssrc, ast_rtp::ssrc, SSRC_MAPPING_ELEM_CMP, rtp_ssrc_mapping::ssrc_valid, ast_rtp::themssrc, and ast_rtp::themssrc_valid.

{
 struct ast_rtp *child_rtp = ast_rtp_instance_get_data(child);
 struct ast_rtp *parent_rtp;
 struct rtp_ssrc_mapping mapping;
 struct ast_sockaddr them = { { 0, } };

 if (child_rtp->bundled == parent) {
 return 0;
 }

 /* If this instance was already bundled then remove the SSRC mapping */
 if (child_rtp->bundled) {
 struct ast_rtp *bundled_rtp;

 ao2_unlock(child);

 /* The child lock can't be held while accessing the parent */
 ao2_lock(child_rtp->bundled);
 bundled_rtp = ast_rtp_instance_get_data(child_rtp->bundled);
 AST_VECTOR_REMOVE_CMP_UNORDERED(&bundled_rtp->ssrc_mapping, child, SSRC_MAPPING_ELEM_CMP, AST_VECTOR_ELEM_CLEANUP_NOOP);
 ao2_unlock(child_rtp->bundled);

 ao2_lock(child);
 ao2_ref(child_rtp->bundled, -1);
 child_rtp->bundled = NULL;
 }

 if (!parent) {
 /* We transitioned away from bundle so we need our own transport resources once again */
 rtp_allocate_transport(child, child_rtp);
 return 0;
 }

 parent_rtp = ast_rtp_instance_get_data(parent);

 /* We no longer need any transport related resources as we will use our parent RTP instance instead */
 rtp_deallocate_transport(child, child_rtp);

 /* Children maintain a reference to the parent to guarantee that the transport doesn't go away on them */
 child_rtp->bundled = ao2_bump(parent);

 mapping.ssrc = child_rtp->themssrc;
 mapping.ssrc_valid = child_rtp->themssrc_valid;
 mapping.instance = child;

 ao2_unlock(child);

 ao2_lock(parent);

 AST_VECTOR_APPEND(&parent_rtp->ssrc_mapping, mapping);

#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
 /* If DTLS-SRTP is already in use then add the local SSRC to it, otherwise it will get added once DTLS
 * negotiation has been completed.
 */
 if (parent_rtp->dtls.connection == AST_RTP_DTLS_CONNECTION_EXISTING) {
 dtls_srtp_add_local_ssrc(parent_rtp, parent, 0, child_rtp->ssrc, 0);
 }
#endif

 /* Bundle requires that RTCP-MUX be in use so only the main remote address needs to match */
 ast_rtp_instance_get_remote_address(parent, &them);

 ao2_unlock(parent);

 ao2_lock(child);

 ast_rtp_instance_set_remote_address(child, &them);

 return 0;
}

ast_rtp_change_source()

static void ast_rtp_change_source ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 4360 of file res_rtp_asterisk.c.

References ast_debug_rtp, ast_random(), ast_rtp_instance_get_data(), ast_rtp_instance_get_srtp(), ast_set_flag, ast_srtp_res::change_source, ast_rtp::expectedrxseqno, FLAG_NEED_MARKER_BIT, ast_rtp::lastts, and ast_rtp::ssrc.

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 struct ast_srtp *srtp = ast_rtp_instance_get_srtp(instance, 0);
 struct ast_srtp *rtcp_srtp = ast_rtp_instance_get_srtp(instance, 1);
 unsigned int ssrc = ast_random();

 if (rtp->lastts) {
 /* 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_rtp(3, "(%p) RTP changing ssrc from %u to %u due to a source change\n",
 instance, rtp->ssrc, ssrc);

 if (srtp) {
 ast_debug_rtp(3, "(%p) RTP changing ssrc for SRTP from %u to %u\n",
 instance, rtp->ssrc, ssrc);
 res_srtp->change_source(srtp, rtp->ssrc, ssrc);
 if (rtcp_srtp != srtp) {
 res_srtp->change_source(rtcp_srtp, rtp->ssrc, ssrc);
 }
 }

 rtp->ssrc = ssrc;

 /* Since the source is changing, we don't know what sequence number to expect next */
 rtp->expectedrxseqno = -1;

 return;
}

ast_rtp_destroy()

static int ast_rtp_destroy ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 4048 of file res_rtp_asterisk.c.

References ao2_cleanup, ao2_lock, ao2_ref, ao2_unlock, ast_data_buffer_free(), ast_free, ast_rtp_instance_get_data(), AST_SCHED_DEL, ast_smoother_free(), AST_VECTOR_ELEM_CLEANUP_NOOP, AST_VECTOR_FREE, AST_VECTOR_REMOVE_CMP_UNORDERED, ast_rtp::f, ast_frame_subclass::format, NULL, rtp_deallocate_transport(), SSRC_MAPPING_ELEM_CMP, and ast_frame::subclass.

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);

 if (rtp->bundled) {
 struct ast_rtp *bundled_rtp;

 /* We can't hold our instance lock while removing ourselves from the parent */
 ao2_unlock(instance);

 ao2_lock(rtp->bundled);
 bundled_rtp = ast_rtp_instance_get_data(rtp->bundled);
 AST_VECTOR_REMOVE_CMP_UNORDERED(&bundled_rtp->ssrc_mapping, instance, SSRC_MAPPING_ELEM_CMP, AST_VECTOR_ELEM_CLEANUP_NOOP);
 ao2_unlock(rtp->bundled);

 ao2_lock(instance);
 ao2_ref(rtp->bundled, -1);
 }

 rtp_deallocate_transport(instance, rtp);

 /* Destroy the smoother that was smoothing out audio if present */
 if (rtp->smoother) {
 ast_smoother_free(rtp->smoother);
 }

 /* 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.
 */
 ast_free(rtp->rtcp->local_addr_str);
 ast_free(rtp->rtcp);
 }

 /* Destroy RED if it was being used */
 if (rtp->red) {
 ao2_unlock(instance);
 AST_SCHED_DEL(rtp->sched, rtp->red->schedid);
 ao2_lock(instance);
 ast_free(rtp->red);
 rtp->red = NULL;
 }

 /* Destroy the send buffer if it was being used */
 if (rtp->send_buffer) {
 ast_data_buffer_free(rtp->send_buffer);
 }

 /* Destroy the recv buffer if it was being used */
 if (rtp->recv_buffer) {
 ast_data_buffer_free(rtp->recv_buffer);
 }

 AST_VECTOR_FREE(&rtp->transport_wide_cc.packet_statistics);

 ao2_cleanup(rtp->lasttxformat);
 ao2_cleanup(rtp->lastrxformat);
 ao2_cleanup(rtp->f.subclass.format);
 AST_VECTOR_FREE(&rtp->ssrc_mapping);
 AST_VECTOR_FREE(&rtp->missing_seqno);

 /* Finally destroy ourselves */
 ast_free(rtp);

 return 0;
}

ast_rtp_dtmf_begin()

static int ast_rtp_dtmf_begin ( struct ast_rtp_instance *  instance, char  digit  )

static

Precondition

instance is locked

Definition at line 4134 of file res_rtp_asterisk.c.

References ast_log, ast_rtp_codecs_payload_code_tx(), AST_RTP_DTMF, ast_rtp_instance_get_codecs(), ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose(), calc_txstamp(), digit, DTMF_SAMPLE_RATE_MS, errno, ast_rtp::lastts, LOG_ERROR, LOG_WARNING, NULL, rtp_debug_test_addr(), rtp_sendto(), and ast_rtp::ssrc.

{
 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_tx(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;
}

ast_rtp_dtmf_compatible()

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

Precondition

Neither instance0 nor instance1 are locked

Definition at line 8723 of file res_rtp_asterisk.c.

References ast_channel_tech(), ast_rtp_instance_get_prop(), and AST_RTP_PROPERTY_DTMF.

{
 /* 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);
}

ast_rtp_dtmf_continuation()

static int ast_rtp_dtmf_continuation ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 4209 of file res_rtp_asterisk.c.

References ast_log, ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_verbose(), calc_txstamp(), DTMF_SAMPLE_RATE_MS, errno, ast_rtp::lastts, LOG_ERROR, NULL, rtp_debug_test_addr(), rtp_sendto(), and ast_rtp::ssrc.

Referenced by ast_rtp_interpret().

{
 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;
}

ast_rtp_dtmf_end()

static int ast_rtp_dtmf_end ( struct ast_rtp_instance *  instance, char  digit  )

static

Precondition

instance is locked

Definition at line 4342 of file res_rtp_asterisk.c.

References ast_rtp_dtmf_end_with_duration().

{
 return ast_rtp_dtmf_end_with_duration(instance, digit, 0);
}

ast_rtp_dtmf_end_with_duration()

static int ast_rtp_dtmf_end_with_duration ( struct ast_rtp_instance *  instance, char  digit, unsigned int  duration  )

static

Precondition

instance is locked

Definition at line 4255 of file res_rtp_asterisk.c.

References ast_debug_rtp, ast_log, ast_rtp_get_rate(), ast_rtp_instance_get_data(), ast_rtp_instance_get_remote_address, ast_smoother_free(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_tv(), ast_tvadd(), ast_tvnow(), ast_verbose(), calc_txstamp(), cleanup(), DTMF_SAMPLE_RATE_MS, errno, ast_rtp::f, ast_frame_subclass::format, ast_rtp::lastts, LOG_ERROR, LOG_WARNING, NULL, rtp_debug_test_addr(), rtp_sendto(), ast_rtp::ssrc, and ast_frame::subclass.

Referenced by ast_rtp_dtmf_end().

{
 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 * ast_rtp_get_rate(rtp->f.subclass.format) / 1000) > rtp->send_duration) {
 ast_debug_rtp(2, "(%p) RTP adjusting final end duration from %d to %u\n",
 instance, 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;

 /* Reset the smoother as the delivery time stored in it is now out of date */
 if (rtp->smoother) {
 ast_smoother_free(rtp->smoother);
 rtp->smoother = NULL;
 }
cleanup:
 rtp->sending_digit = 0;
 rtp->send_digit = 0;

 return res;
}

ast_rtp_dtmf_mode_get()

static enum ast_rtp_dtmf_mode ast_rtp_dtmf_mode_get ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 4127 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_data().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 return rtp->dtmfmode;
}

ast_rtp_dtmf_mode_set()

static int ast_rtp_dtmf_mode_set ( struct ast_rtp_instance *  instance, enum ast_rtp_dtmf_mode  dtmf_mode  )

static

Precondition

instance is locked

Definition at line 4119 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_data().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 rtp->dtmfmode = dtmf_mode;
 return 0;
}

ast_rtp_extension_enable()

static int ast_rtp_extension_enable ( struct ast_rtp_instance *  instance, enum ast_rtp_extension  extension  )

static

Definition at line 8926 of file res_rtp_asterisk.c.

References AST_RTP_EXTENSION_ABS_SEND_TIME, and AST_RTP_EXTENSION_TRANSPORT_WIDE_CC.

{
 switch (extension) {
 case AST_RTP_EXTENSION_ABS_SEND_TIME:
 case AST_RTP_EXTENSION_TRANSPORT_WIDE_CC:
 return 1;
 default:
 return 0;
 }
}

ast_rtp_fd()

static int ast_rtp_fd ( struct ast_rtp_instance *  instance, int  rtcp  )

static

Precondition

instance is locked

Definition at line 8489 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_data(), and ast_rtp::s.

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);

 return rtcp ? (rtp->rtcp ? rtp->rtcp->s : -1) : rtp->s;
}

ast_rtp_get_cname()

static const char * ast_rtp_get_cname ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 8873 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_data(), and ast_rtp::cname.

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);

 return rtp->cname;
}

ast_rtp_get_ssrc()

static unsigned int ast_rtp_get_ssrc ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 8865 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_data(), and ast_rtp::ssrc.

Referenced by __rtp_find_instance_by_ssrc().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);

 return rtp->ssrc;
}

ast_rtp_get_stat()

static int ast_rtp_get_stat ( struct ast_rtp_instance *  instance, struct ast_rtp_instance_stats *  stats, enum ast_rtp_instance_stat  stat  )

static

Precondition

instance is locked

Definition at line 8671 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_channel_id(), ast_rtp_instance_get_data(), AST_RTP_INSTANCE_STAT_CHANNEL_UNIQUEID, AST_RTP_INSTANCE_STAT_COMBINED_JITTER, AST_RTP_INSTANCE_STAT_COMBINED_LOSS, AST_RTP_INSTANCE_STAT_COMBINED_RTT, AST_RTP_INSTANCE_STAT_LOCAL_MAXJITTER, AST_RTP_INSTANCE_STAT_LOCAL_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_MINJITTER, AST_RTP_INSTANCE_STAT_LOCAL_MINRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_LOCAL_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_LOCAL_SSRC, AST_RTP_INSTANCE_STAT_LOCAL_STDEVJITTER, AST_RTP_INSTANCE_STAT_LOCAL_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_MAX_RTT, AST_RTP_INSTANCE_STAT_MIN_RTT, AST_RTP_INSTANCE_STAT_NORMDEVRTT, AST_RTP_INSTANCE_STAT_REMOTE_MAXJITTER, AST_RTP_INSTANCE_STAT_REMOTE_MAXRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_MINJITTER, AST_RTP_INSTANCE_STAT_REMOTE_MINRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVJITTER, AST_RTP_INSTANCE_STAT_REMOTE_NORMDEVRXPLOSS, AST_RTP_INSTANCE_STAT_REMOTE_SSRC, AST_RTP_INSTANCE_STAT_REMOTE_STDEVJITTER, AST_RTP_INSTANCE_STAT_REMOTE_STDEVRXPLOSS, AST_RTP_INSTANCE_STAT_RTT, AST_RTP_INSTANCE_STAT_RXCOUNT, AST_RTP_INSTANCE_STAT_RXJITTER, AST_RTP_INSTANCE_STAT_RXOCTETCOUNT, AST_RTP_INSTANCE_STAT_RXPLOSS, AST_RTP_INSTANCE_STAT_STDEVRTT, AST_RTP_INSTANCE_STAT_TXCOUNT, AST_RTP_INSTANCE_STAT_TXJITTER, AST_RTP_INSTANCE_STAT_TXOCTETCOUNT, AST_RTP_INSTANCE_STAT_TXPLOSS, AST_RTP_STAT_SET, AST_RTP_STAT_STRCPY, AST_RTP_STAT_TERMINATOR, ast_rtp_instance_stats::channel_uniqueid, ast_rtp_instance_stats::local_maxjitter, ast_rtp_instance_stats::local_maxrxploss, ast_rtp_instance_stats::local_minjitter, ast_rtp_instance_stats::local_minrxploss, ast_rtp_instance_stats::local_normdevjitter, ast_rtp_instance_stats::local_normdevrxploss, ast_rtp_instance_stats::local_ssrc, ast_rtp_instance_stats::local_stdevjitter, ast_rtp_instance_stats::local_stdevrxploss, ast_rtp_instance_stats::maxrtt, ast_rtp_instance_stats::minrtt, ast_rtp_instance_stats::normdevrtt, ast_rtp_instance_stats::remote_maxjitter, ast_rtp_instance_stats::remote_maxrxploss, ast_rtp_instance_stats::remote_minjitter, ast_rtp_instance_stats::remote_minrxploss, ast_rtp_instance_stats::remote_normdevjitter, ast_rtp_instance_stats::remote_normdevrxploss, ast_rtp_instance_stats::remote_ssrc, ast_rtp_instance_stats::remote_stdevjitter, ast_rtp_instance_stats::remote_stdevrxploss, ast_rtp_instance_stats::rtt, ast_rtp_instance_stats::rxcount, ast_rtp_instance_stats::rxjitter, ast_rtp_instance_stats::rxoctetcount, ast_rtp_instance_stats::rxploss, ast_rtp::ssrc, ast_rtp_instance_stats::stdevrtt, ast_rtp::themssrc, ast_rtp_instance_stats::txcount, ast_rtp_instance_stats::txjitter, ast_rtp_instance_stats::txoctetcount, and ast_rtp_instance_stats::txploss.

{
 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_TXOCTETCOUNT, -1, stats->txoctetcount, rtp->txoctetcount);
 AST_RTP_STAT_SET(AST_RTP_INSTANCE_STAT_RXOCTETCOUNT, -1, stats->rxoctetcount, rtp->rxoctetcount);

 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;
}

ast_rtp_ice_add_cand()

static void ast_rtp_ice_add_cand ( struct ast_rtp_instance *  instance, 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  )

static

Precondition

instance is locked

Definition at line 1239 of file res_rtp_asterisk.c.

References ast_rtp_engine_ice_candidate::address, ao2_alloc, AO2_ALLOC_OPT_LOCK_MUTEX, ao2_container_alloc_list, ao2_find, ao2_link, ao2_lock, ao2_ref, ao2_unlock, ast_debug_ice, ast_rtp_ice_candidate_destroy(), AST_RTP_ICE_CANDIDATE_TYPE_HOST, AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, AST_RTP_ICE_CANDIDATE_TYPE_SRFLX, ast_sockaddr_parse(), ast_sockaddr_set_port, ast_sockaddr_stringify(), ast_strdup, ast_strndup, ast_rtp_engine_ice_candidate::foundation, ice_candidate_cmp(), ast_rtp_engine_ice_candidate::id, NULL, OBJ_POINTER, pj_thread_register_check(), ast_rtp_engine_ice_candidate::priority, ice_wrap::real_ice, ast_rtp_engine_ice_candidate::relay_address, status, ast_rtp_engine_ice_candidate::transport, and ast_rtp_engine_ice_candidate::type.

Referenced by ast_rtp_ice_turn_request(), and rtp_add_candidates_to_ice().

{
 pj_str_t foundation;
 struct ast_rtp_engine_ice_candidate *candidate, *existing;
 struct ice_wrap *ice;
 char address[PJ_INET6_ADDRSTRLEN];
 pj_status_t status;

 if (!rtp->ice) {
 return;
 }

 pj_thread_register_check();

 pj_ice_calc_foundation(rtp->ice->real_ice->pool, &foundation, type, addr);

 if (!rtp->ice_local_candidates) {
 rtp->ice_local_candidates = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
 NULL, ice_candidate_cmp);
 if (!rtp->ice_local_candidates) {
 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;
 }

 /* Release the instance lock to avoid deadlock with PJPROJECT group lock */
 ice = rtp->ice;
 ao2_ref(ice, +1);
 ao2_unlock(instance);
 status = pj_ice_sess_add_cand(ice->real_ice, comp_id, transport_id, type, local_pref,
 &foundation, addr, base_addr, rel_addr, addr_len, NULL);
 ao2_ref(ice, -1);
 ao2_lock(instance);
 if (!rtp->ice || status != PJ_SUCCESS) {
 ast_debug_ice(2, "(%p) ICE unable to add candidate: %s, %d\n", instance, ast_sockaddr_stringify(
 &candidate->address), candidate->priority);
 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->real_ice->lcand[rtp->ice->real_ice->lcand_cnt - 1].prio;

 ast_debug_ice(2, "(%p) ICE add candidate: %s, %d\n", instance, ast_sockaddr_stringify(
 &candidate->address), candidate->priority);

 ao2_link(rtp->ice_local_candidates, candidate);
 ao2_ref(candidate, -1);
}

ast_rtp_ice_add_remote_candidate()

static void ast_rtp_ice_add_remote_candidate ( struct ast_rtp_instance *  instance, const struct ast_rtp_engine_ice_candidate *  candidate  )

static

Precondition

instance is locked

Definition at line 821 of file res_rtp_asterisk.c.

References ast_rtp_engine_ice_candidate::address, ao2_alloc, AO2_ALLOC_OPT_LOCK_MUTEX, ao2_container_alloc_list, ao2_container_count(), ao2_link, ao2_ref, ast_debug_ice, ast_rtp_ice_candidate_destroy(), ast_rtp_instance_get_data(), ast_sockaddr_copy(), ast_strdup, ast_rtp_engine_ice_candidate::foundation, ice_candidate_cmp(), ast_rtp_engine_ice_candidate::id, NULL, ast_rtp_engine_ice_candidate::priority, ast_rtp_engine_ice_candidate::relay_address, ast_rtp_engine_ice_candidate::transport, and ast_rtp_engine_ice_candidate::type.

{
 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_list(
 AO2_ALLOC_OPT_LOCK_MUTEX, 0, NULL, ice_candidate_cmp);
 if (!rtp->ice_proposed_remote_candidates) {
 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;

 ast_debug_ice(2, "(%p) ICE add remote candidate\n", instance);

 ao2_link(rtp->ice_proposed_remote_candidates, remote_candidate);
 ao2_ref(remote_candidate, -1);
}

ast_rtp_ice_candidate_destroy()

static void ast_rtp_ice_candidate_destroy ( void *  obj )

static

Destructor for locally created ICE candidates.

Definition at line 766 of file res_rtp_asterisk.c.

References ast_free, ast_rtp_engine_ice_candidate::foundation, and ast_rtp_engine_ice_candidate::transport.

Referenced by ast_rtp_ice_add_cand(), and ast_rtp_ice_add_remote_candidate().

{
 struct ast_rtp_engine_ice_candidate *candidate = obj;

 if (candidate->foundation) {
 ast_free(candidate->foundation);
 }

 if (candidate->transport) {
 ast_free(candidate->transport);
 }
}

ast_rtp_ice_change_components()

static void ast_rtp_ice_change_components ( struct ast_rtp_instance *  instance, int  num_components  )

static

Precondition

instance is locked

Definition at line 1734 of file res_rtp_asterisk.c.

References ast_debug_ice, ast_rtp_instance_get_data(), ice_reset_session(), and icesupport.

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);

 /* Don't do anything if ICE is unsupported or if we're not changing the
 * number of components
 */
 if (!icesupport || !rtp->ice || rtp->ice_num_components == num_components) {
 return;
 }

 ast_debug_ice(2, "(%p) ICE change number of components %u -> %u\n", instance,
 rtp->ice_num_components, num_components);

 rtp->ice_num_components = num_components;
 ice_reset_session(instance);
}

ast_rtp_ice_get_local_candidates()

static struct ao2_container* ast_rtp_ice_get_local_candidates ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 1189 of file res_rtp_asterisk.c.

References ao2_ref, and ast_rtp_instance_get_data().

{
 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;
}

ast_rtp_ice_get_password()

static const char* ast_rtp_ice_get_password ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 1181 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_data().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);

 return rtp->local_passwd;
}

ast_rtp_ice_get_ufrag()

static const char* ast_rtp_ice_get_ufrag ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 1173 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_data().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);

 return rtp->local_ufrag;
}

ast_rtp_ice_lite()

static void ast_rtp_ice_lite ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 1201 of file res_rtp_asterisk.c.

References ast_rtp_instance_get_data(), and pj_thread_register_check().

{
 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->real_ice, PJ_ICE_SESS_ROLE_CONTROLLING);
}

ast_rtp_ice_set_authentication()

static void ast_rtp_ice_set_authentication ( struct ast_rtp_instance *  instance, const char *  ufrag, const char *  password  )

static

Precondition

instance is locked

Definition at line 780 of file res_rtp_asterisk.c.

References ast_copy_string(), ast_rtp_instance_get_data(), ast_strlen_zero, and generate_random_string().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);
 int ice_attrb_reset = 0;

 if (!ast_strlen_zero(ufrag)) {
 if (!ast_strlen_zero(rtp->remote_ufrag) && strcmp(ufrag, rtp->remote_ufrag)) {
 ice_attrb_reset = 1;
 }
 ast_copy_string(rtp->remote_ufrag, ufrag, sizeof(rtp->remote_ufrag));
 }

 if (!ast_strlen_zero(password)) {
 if (!ast_strlen_zero(rtp->remote_passwd) && strcmp(password, rtp->remote_passwd)) {
 ice_attrb_reset = 1;
 }
 ast_copy_string(rtp->remote_passwd, password, sizeof(rtp->remote_passwd));
 }

 /* If the remote ufrag or passwd changed, local ufrag and passwd need to regenerate */
 if (ice_attrb_reset) {
 generate_random_string(rtp->local_ufrag, sizeof(rtp->local_ufrag));
 generate_random_string(rtp->local_passwd, sizeof(rtp->local_passwd));
 }
}

ast_rtp_ice_set_role()

static void ast_rtp_ice_set_role ( struct ast_rtp_instance *  instance, enum ast_rtp_ice_role  role  )

static

Precondition

instance is locked

Definition at line 1215 of file res_rtp_asterisk.c.

References ast_debug_ice, AST_RTP_ICE_ROLE_CONTROLLED, ast_rtp_instance_get_data(), and pj_thread_register_check().

{
 struct ast_rtp *rtp = ast_rtp_instance_get_data(instance);

 if (!rtp->ice) {
 ast_debug_ice(3, "(%p) ICE set role failed; no ice instance\n", instance);
 return;
 }

 rtp->role = role;

 if (!rtp->ice->real_ice->is_nominating && !rtp->ice->real_ice->is_complete) {
 pj_thread_register_check();
 ast_debug_ice(2, "(%p) ICE set role to %s\n",
 instance, role == AST_RTP_ICE_ROLE_CONTROLLED ? "CONTROLLED" : "CONTROLLING");
 pj_ice_sess_change_role(rtp->ice->real_ice, role == AST_RTP_ICE_ROLE_CONTROLLED ?
 PJ_ICE_SESS_ROLE_CONTROLLED : PJ_ICE_SESS_ROLE_CONTROLLING);
 } else {
 ast_debug_ice(2, "(%p) ICE not setting role because state is %s\n",
 instance, rtp->ice->real_ice->is_nominating ? "nominating" : "complete");
 }
}

ast_rtp_ice_start()

static void ast_rtp_ice_start ( struct ast_rtp_instance *  instance )

static

Precondition

instance is locked

Definition at line 1031 of file res_rtp_asterisk.c.

References ast_rtp_engine_ice_candidate::address, ao2_cleanup, ao2_container_count(), ao2_iterator_destroy(), ao2_iterator_init(), ao2_iterator_next, ao2_lock, ao2_ref, ao2_unlock, ast_debug_ice, ast_log, AST_RTP_ICE_CANDIDATE_TYPE_HOST, AST_RTP_ICE_CANDIDATE_TYPE_RELAYED, AST_RTP_ICE_CANDIDATE_TYPE_SRFLX, AST_RTP_ICE_COMPONENT_RTCP, AST_RTP_ICE_COMPONENT_RTP, ast_rtp_instance_get_data(), ast_sockaddr_isnull(), ast_sockaddr_stringify(), ast_test_suite_event_notify, ast_rtp_engine_ice_candidate::foundation, ice_candidates_compare(), ice_reset_session(), ast_rtp_engine_ice_candidate::id, LOG_NOTICE, LOG_WARNING, NULL, pj2ast_rtp_ice_role(), pj_thread_register_check(), ast_rtp_engine_ice_candidate::priority, ice_wrap::real_ice, ast_rtp_engine_ice_candidate::relay_address, STRICT_RTP_OPEN, timer_heap, and ast_rtp_engine_ice_candidate::type.

{
 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)) {
 ast_debug_ice(2, "(%p) ICE proposed equals active candidates\n", instance);
 ao2_cleanup(rtp->ice_proposed_remote_candidates);
 rtp->ice_proposed_remote_candidates = NULL;
 /* If this ICE session is being preserved then go back to the role it currently is */
 pj2ast_rtp_ice_role(rtp->ice->real_ice->role, &rtp->role);
 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;

 ast_debug_ice(2, "(%p) ICE start\n", instance);

 /* Reset the ICE session. Is this going to work? */
 if (ice_reset_session(instance)) {
 ast_log(LOG_NOTICE, "(%p) ICE failed to create replacement session\n", instance);
 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->real_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) {
 ast_debug_ice(2, "(%p) ICE RTP candidate %s\n", instance, ast_sockaddr_stringify(&candidate->address));
 /* Release the instance lock to avoid deadlock with PJPROJECT group lock */
 ao2_unlock(instance);
 pj_turn_sock_set_perm(rtp->turn_rtp, 1, &candidates[cand_cnt].addr, 1);
 ao2_lock(instance);
 } else if (candidate->id == AST_RTP_ICE_COMPONENT_RTCP && rtp->turn_rtcp) {
 ast_debug_ice(2, "(%p) ICE RTCP candidate %s\n", instance, ast_sockaddr_stringify(&candidate->address));
 /* Release the instance lock to avoid deadlock with PJPROJECT group lock */
 ao2_unlock(instance);
 pj_turn_sock_set_perm(rtp->turn_rtcp, 1, &candidates[cand_cnt].addr, 1);
 ao2_lock(instance);
 }

 cand_cnt++;
 ao2_ref(candidate, -1);
 }

 ao2_iterator_destroy(&i);

 if (cand_cnt < ao2_container_count(rtp->ice_active_remote_candidates)) {
 ast_log(LOG_WARNING, "(%p) ICE lost %d candidates. Consider increasing PJ_ICE_MAX_CAND in PJSIP\n",
 instance, ao2_container_count(rtp->ice_active_remote_candidates) - cand_cnt);
 }

 if (!has_rtp) {
 ast_log(LOG_WARNING, "(%p) ICE no RTP candidates; skipping checklist\n", instance);
 }

 /* If we're only dealing with one ICE component, then we don't care about the lack of RTCP candidates */
 if (!has_rtcp && rtp->ice_num_components > 1) {
 ast_log(LOG_WARNING, "(%p) ICE no RTCP candidates; skipping checklist\n", instance);
 }

 if (rtp->ice && has_rtp && (has_rtcp || rtp->ice_num_components == 1)) {
 pj_status_t res;
 char reason[80];
 struct ice_wrap *ice;

 /* Release the instance lock to avoid deadlock with PJPROJECT group lock */
 ice = rtp->ice;
 ao2_ref(ice, +1);
 ao2_unlock(instance);
 res = pj_ice_sess_create_check_list(ice->real_ice, &ufrag, &passwd, cand_cnt, &candidates[0]);
 if (res == PJ_SUCCESS) {
 ast_debug_ice(2, "(%p) ICE successfully created checklist\n", instance);
 ast_test_suite_event_notify("ICECHECKLISTCREATE", "Result: SUCCESS");
 pj_ice_sess_start_check(ice->real_ice);
 pj_timer_heap_poll(timer_heap, NULL);
 ao2_ref(ice, -1);
 ao2_lock(instance);
 rtp->strict_rtp_state = STRICT_RTP_OPEN;
 return;
 }
 ao2_ref(ice, -1);
 ao2_lock(instance);

 pj_strerror(res, reason, sizeof(reason));
 ast_log(LOG_WARNING, "(%p) ICE failed to create session check list: %s\n", instance, reason);
 }

 ast_test_suite_event_notify("ICECHECKLISTCREATE", "Result: FAILURE");

 /* even though create check list failed don't stop ice as
 it might still work */
 /* 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;
 if (rtp->ice) {
 rtp->ice->real_ice->rcand_cnt = rtp->ice->real_ice->clist.count = 0;
 }
}