osmo-mgw/include/osmocom/mgcp/mgcp_network.h

186 lines
5.8 KiB
C

#pragma once
#include <inttypes.h>
#include <stdbool.h>
#include <osmocom/core/socket.h>
#include <osmocom/mgcp/mgcp.h>
/* The following constant defines an RTP dummy payload that is used for
* "UDP Hole Punching" (NAT) */
#define MGCP_DUMMY_LOAD 0x23
static const char rtp_dummy_payload[] = { MGCP_DUMMY_LOAD };
/* Check if the data in a given message buffer matches the rtp dummy payload
* defined above */
#define mgcp_is_rtp_dummy_payload(msg) \
(msgb_length(msg) == sizeof(rtp_dummy_payload) && \
memcmp(msgb_data(msg), rtp_dummy_payload, sizeof(rtp_dummy_payload)) == 0)
#define RTP_BUF_SIZE 4096
struct mgcp_rtp_stream_state {
uint32_t ssrc;
uint16_t last_seq;
uint32_t last_timestamp;
struct rate_ctr *err_ts_ctr;
int32_t last_tsdelta;
uint32_t last_arrival_time;
};
struct mgcp_rtp_state {
/* has this state structure been initialized? */
int initialized;
struct {
/* are we patching the SSRC value? */
bool patch_ssrc;
/* original SSRC (to which we shall patch any different SSRC) */
uint32_t orig_ssrc;
/* offset to apply on the sequence number */
int seq_offset;
/* offset to apply on the timestamp number */
int32_t timestamp_offset;
} patch;
/* duration of a packet (FIXME: in which unit?) */
uint32_t packet_duration;
/* Note: These states are not continuously updated, they serve as an
* information source to patch certain values in the RTP header. Do
* not use this state if constantly updated data about the RTP stream
* is needed. (see also mgcp_patch_and_count() */
struct mgcp_rtp_stream_state in_stream;
struct mgcp_rtp_stream_state out_stream;
/* jitter and packet loss calculation */
struct {
int initialized;
uint16_t base_seq;
uint16_t max_seq;
uint32_t ssrc;
uint32_t jitter;
int32_t transit;
int cycles;
} stats;
/* Alternative values for RTP tx, in case no sufficient header
* information is available so the header needs to be generated
* locally (when just forwarding packets, the header of incoming
* data is just re-used) */
uint16_t alt_rtp_tx_sequence;
uint32_t alt_rtp_tx_ssrc;
};
struct mgcp_rtp_codec {
uint32_t rate;
int channels;
uint32_t frame_duration_num;
uint32_t frame_duration_den;
int payload_type;
char audio_name[64];
char subtype_name[64];
bool param_present;
struct mgcp_codec_param param;
};
/* 'mgcp_rtp_end': basically a wrapper around the RTP+RTCP ports */
struct mgcp_rtp_end {
/* remote IP address of the RTP socket */
struct osmo_sockaddr addr;
/* in network byte order */
int rtcp_port;
/* currently selected audio codec */
struct mgcp_rtp_codec *codec;
/* array with assigned audio codecs to choose from (SDP) */
struct mgcp_rtp_codec codecs[MGCP_MAX_CODECS];
/* number of assigned audio codecs (SDP) */
unsigned int codecs_assigned;
/* per endpoint data */
int frames_per_packet;
uint32_t packet_duration_ms;
int maximum_packet_time; /* -1: not set */
char *fmtp_extra;
/* are we transmitting packets (true) or dropping (false) outbound packets */
bool output_enabled;
/* FIXME: This parameter can be set + printed, but is nowhere used! */
int force_output_ptime;
/* RTP patching */
int force_constant_ssrc; /* -1: always, 0: don't, 1: once */
/* should we perform align_rtp_timestamp_offset() (1) or not (0) */
int force_aligned_timing;
bool rfc5993_hr_convert;
/* Each end has a separate socket for RTP and RTCP */
struct osmo_fd rtp;
struct osmo_fd rtcp;
/* local UDP port number of the RTP socket; RTCP is +1 */
int local_port;
/* where the endpoint RTP connection binds to, set during CRCX and
* possibly updated during MDCX */
char local_addr[INET6_ADDRSTRLEN];
};
bool mgcp_rtp_end_remote_addr_available(const struct mgcp_rtp_end *rtp_end);
struct mgcp_rtp_tap {
/* is this tap active (1) or not (0) */
int enabled;
/* IP/port to which we're forwarding the tapped data */
struct osmo_sockaddr forward;
};
struct mgcp_conn;
int mgcp_send(struct mgcp_endpoint *endp, int is_rtp, struct osmo_sockaddr *addr,
struct msgb *msg, struct mgcp_conn_rtp *conn_src,
struct mgcp_conn_rtp *conn_dst);
int mgcp_send_dummy(struct mgcp_endpoint *endp, struct mgcp_conn_rtp *conn);
int mgcp_dispatch_rtp_bridge_cb(struct msgb *msg);
void mgcp_cleanup_rtp_bridge_cb(struct mgcp_endpoint *endp, struct mgcp_conn *conn);
int mgcp_dispatch_e1_bridge_cb(struct msgb *msg);
void mgcp_cleanup_e1_bridge_cb(struct mgcp_endpoint *endp, struct mgcp_conn *conn);
int mgcp_bind_net_rtp_port(struct mgcp_endpoint *endp, int rtp_port,
struct mgcp_conn_rtp *conn);
void mgcp_free_rtp_port(struct mgcp_rtp_end *end);
void mgcp_patch_and_count(const struct mgcp_endpoint *endp,
struct mgcp_rtp_state *state,
struct mgcp_rtp_end *rtp_end,
struct osmo_sockaddr *addr, struct msgb *msg);
int mgcp_get_local_addr(char *addr, struct mgcp_conn_rtp *conn);
/* payload processing default functions */
int mgcp_rtp_processing_default(struct mgcp_endpoint *endp, struct mgcp_rtp_end *dst_end,
char *data, int *len, int buf_size);
int mgcp_setup_rtp_processing_default(struct mgcp_endpoint *endp,
struct mgcp_conn_rtp *conn_dst,
struct mgcp_conn_rtp *conn_src);
void mgcp_get_net_downlink_format_default(struct mgcp_endpoint *endp,
const struct mgcp_rtp_codec **codec,
const char **fmtp_extra,
struct mgcp_conn_rtp *conn);
/* internal RTP Annex A counting */
void mgcp_rtp_annex_count(const struct mgcp_endpoint *endp, struct mgcp_rtp_state *state,
const uint16_t seq, const int32_t transit,
const uint32_t ssrc, const bool marker_bit);
void rtpconn_rate_ctr_add(struct mgcp_conn_rtp *conn_rtp, struct mgcp_endpoint *endp,
int id, int inc);
void rtpconn_rate_ctr_inc(struct mgcp_conn_rtp *conn_rtp, struct mgcp_endpoint *endp,
int id);
void forward_data_tap(int fd, struct mgcp_rtp_tap *tap, struct msgb *msg);
uint32_t mgcp_get_current_ts(unsigned codec_rate);