Files
osmo-mgw/include/osmocom/mgcp/mgcp_internal.h
Philipp Maier bc0346e080 mgw: clean up codec negotiation (sdp)
The codec negotiation via SDP is currently in a neglected state. Also
osmo-mgw does some kind of codec decision wile the SDP is parsed, the
result is information for one codec, even when there are multiple codecs
negotiated. This is problematic because we loose all information about
alternate codecs while we parse. This should be untangled and the
information should be presevered. Also we are not really capable
picking a default. Wehen we do not supply any codec information (not
even LCO), then we should pick a sane default codec.

- separate the codec decision from the sdp parser and concentrate
  codec related code in a separate c file
- add support for multiple codecs in one SDP negotiation
- do not initalize "magic" codec defaults during conn allocation
- do not allow invalid payload types, especially not 255. When
  someone tries to select an invalid payload type, do not fail
  hard, just pick a sane default.
- handle the codec decision in protocol.c, pick a sane default
  codec when no (valid) codec has been negotiated (no LCO, no SDP)

Change-Id: If730d022ba6bdb217ad4e20b3fbbd1114dbb4b8f
Closes: OS#2658
Related: OS#3114
Related: OS#2728
2018-06-23 11:39:44 +00:00

342 lines
8.7 KiB
C

/* MGCP Private Data */
/*
* (C) 2009-2012 by Holger Hans Peter Freyther <zecke@selfish.org>
* (C) 2009-2012 by On-Waves
* All Rights Reserved
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU Affero General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Affero General Public License for more details.
*
* You should have received a copy of the GNU Affero General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
*/
#pragma once
#include <string.h>
#include <inttypes.h>
#include <osmocom/core/select.h>
#include <osmocom/mgcp/mgcp.h>
#include <osmocom/core/linuxlist.h>
#include <osmocom/core/counter.h>
#include <osmocom/core/rate_ctr.h>
#define CI_UNUSED 0
/* FIXME: This this is only needed to compile the currently
* broken OSMUX support. Remove when fixed */
#define CONN_ID_BTS "0"
#define CONN_ID_NET "1"
enum mgcp_trunk_type {
MGCP_TRUNK_VIRTUAL,
MGCP_TRUNK_E1,
};
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? */
int 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;
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;
bool patched_first_rtp_payload; /* FIXME: drop this, see OS#2459 */
};
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;
char *subtype_name;
};
/* 'mgcp_rtp_end': basically a wrapper around the RTP+RTCP ports */
struct mgcp_rtp_end {
/* statistics */
struct {
unsigned int packets_rx;
unsigned int octets_rx;
unsigned int packets_tx;
unsigned int octets_tx;
unsigned int dropped_packets;
} stats;
/* local IP address of the RTP socket */
struct in_addr addr;
/* in network byte order */
int rtp_port, 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 (1) or dropping (0) outbound packets */
int 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;
/* FIXME: not used anymore, used to be [external] transcoding related */
void *rtp_process_data;
/* 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;
};
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 sockaddr_in forward;
};
struct mgcp_lco {
char *string;
char *codec;
int pkt_period_min; /* time in ms */
int pkt_period_max; /* time in ms */
};
/* Specific rtp connection type (see struct mgcp_conn_rtp) */
enum mgcp_conn_rtp_type {
MGCP_RTP_DEFAULT = 0,
MGCP_OSMUX_BSC,
MGCP_OSMUX_BSC_NAT,
};
#include <osmocom/mgcp/osmux.h>
struct mgcp_conn;
/* MGCP connection (RTP) */
struct mgcp_conn_rtp {
/* Backpointer to conn struct */
struct mgcp_conn *conn;
/* Specific connection type */
enum mgcp_conn_rtp_type type;
/* Port status */
struct mgcp_rtp_end end;
/* Sequence bits */
struct mgcp_rtp_state state;
/* taps for the rtp connection; one per direction */
struct mgcp_rtp_tap tap_in;
struct mgcp_rtp_tap tap_out;
/* Osmux states (optional) */
struct {
/* Osmux state: disabled, activating, active */
enum osmux_state state;
/* Allocated Osmux circuit ID for this endpoint */
int allocated_cid;
/* Used Osmux circuit ID for this endpoint */
uint8_t cid;
/* handle to batch messages */
struct osmux_in_handle *in;
/* handle to unbatch messages */
struct osmux_out_handle out;
/* statistics */
struct {
uint32_t chunks;
uint32_t octets;
} stats;
} osmux;
struct rate_ctr_group *rate_ctr_group;
};
/*! Connection type, specifies which member of the union "u" in mgcp_conn
* contains a useful connection description (currently only RTP) */
enum mgcp_conn_type {
MGCP_CONN_TYPE_RTP,
};
/*! MGCP connection (untyped) */
struct mgcp_conn {
/*!< list head */
struct llist_head entry;
/*!< Backpointer to the endpoint where the conn belongs to */
struct mgcp_endpoint *endp;
/*!< type of the connection (union) */
enum mgcp_conn_type type;
/*!< mode of the connection */
enum mgcp_connection_mode mode;
/*!< copy of the mode to restore the original setting (VTY) */
enum mgcp_connection_mode mode_orig;
/*!< connection id to identify the connection */
char id[MGCP_CONN_ID_LENGTH];
/*!< human readable name (vty, logging) */
char name[256];
/*!< union with connection description */
union {
struct mgcp_conn_rtp rtp;
} u;
/*!< pointer to optional private data */
void *priv;
};
#include <osmocom/mgcp/mgcp_conn.h>
struct mgcp_endpoint_type;
/**
* Internal structure while parsing a request
*/
struct mgcp_parse_data {
struct mgcp_config *cfg;
struct mgcp_endpoint *endp;
char *trans;
char *save;
};
int mgcp_send(struct mgcp_endpoint *endp, int is_rtp, struct sockaddr_in *addr,
char *buf, int rc, 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(int proto, struct sockaddr_in *addr, char *buf,
unsigned int buf_size, struct mgcp_conn *conn);
void mgcp_cleanup_rtp_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);
/* For transcoding we need to manage an in and an output that are connected */
static inline int endp_back_channel(int endpoint)
{
return endpoint + 60;
}
struct mgcp_trunk_config *mgcp_trunk_alloc(struct mgcp_config *cfg, int index);
struct mgcp_trunk_config *mgcp_trunk_num(struct mgcp_config *cfg, int index);
char *get_lco_identifier(const char *options);
int check_local_cx_options(void *ctx, const char *options);
void mgcp_rtp_end_config(struct mgcp_endpoint *endp, int expect_ssrc_change,
struct mgcp_rtp_end *rtp);
uint32_t mgcp_rtp_packet_duration(struct mgcp_endpoint *endp,
struct mgcp_rtp_end *rtp);
/* 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_rtp_end *dst_end,
struct mgcp_rtp_end *src_end);
void mgcp_get_net_downlink_format_default(struct mgcp_endpoint *endp,
int *payload_type,
const char**audio_name,
const char**fmtp_extra,
struct mgcp_conn_rtp *conn);
/* internal RTP Annex A counting */
void mgcp_rtp_annex_count(struct mgcp_endpoint *endp, struct mgcp_rtp_state *state,
const uint16_t seq, const int32_t transit,
const uint32_t ssrc);
int mgcp_set_ip_tos(int fd, int tos);
enum {
MGCP_DEST_NET = 0,
MGCP_DEST_BTS,
};
#define MGCP_DUMMY_LOAD 0x23
/**
* SDP related information
*/
/* Assume audio frame length of 20ms */
#define DEFAULT_RTP_AUDIO_FRAME_DUR_NUM 20
#define DEFAULT_RTP_AUDIO_FRAME_DUR_DEN 1000
#define DEFAULT_RTP_AUDIO_PACKET_DURATION_MS 20
#define DEFAULT_RTP_AUDIO_DEFAULT_RATE 8000
#define DEFAULT_RTP_AUDIO_DEFAULT_CHANNELS 1
#define PTYPE_UNDEFINED (-1)
void mgcp_get_local_addr(char *addr, struct mgcp_conn_rtp *conn);