osmo-msc/src/libmsc/call_leg.c

399 lines
13 KiB
C

/* Implementation to manage two RTP streams that make up an MO or MT call leg's RTP forwarding. */
/*
* (C) 2019 by sysmocom - s.f.m.c. GmbH <info@sysmocom.de>
* All Rights Reserved
*
* Author: Neels Hofmeyr
*
* SPDX-License-Identifier: AGPL-3.0+
*
* 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.
*/
#include <osmocom/core/fsm.h>
#include <osmocom/mgcp_client/mgcp_client_endpoint_fsm.h>
#include <osmocom/msc/debug.h>
#include <osmocom/msc/gsm_data.h>
#include <osmocom/msc/msc_a.h>
#include <osmocom/msc/call_leg.h>
#include <osmocom/msc/rtp_stream.h>
#define LOG_CALL_LEG(cl, level, fmt, args...) \
LOGPFSML(cl ? cl->fi : NULL, level, fmt, ##args)
static struct gsm_network *gsmnet = NULL;
enum call_leg_state {
CALL_LEG_ST_ESTABLISHING,
CALL_LEG_ST_ESTABLISHED,
CALL_LEG_ST_RELEASING,
};
struct osmo_tdef g_mgw_tdefs[] = {
{ .T=-2427, .default_val=4, .desc="MGCP response timeout" },
{ .T=-2, .default_val=30, .desc="RTP stream establishing timeout" },
{}
};
static const struct osmo_tdef_state_timeout call_leg_fsm_timeouts[32] = {
[CALL_LEG_ST_ESTABLISHING] = { .T = -2 },
[CALL_LEG_ST_RELEASING] = { .T = -2 },
};
#define call_leg_state_chg(cl, state) \
osmo_tdef_fsm_inst_state_chg((cl)->fi, state, call_leg_fsm_timeouts, g_mgw_tdefs, 10)
static struct osmo_fsm call_leg_fsm;
void call_leg_init(struct gsm_network *net)
{
gsmnet = net;
OSMO_ASSERT( osmo_fsm_register(&call_leg_fsm) == 0 );
}
/* Allocate a call leg FSM instance as child of an arbitrary other FSM instance.
* The call leg FSM dispatches events to its parent FSM instance on specific events:
* - parent_event_term: dispatch this to the parent FI when the call leg terminates (call ended, either planned or by
* failure).
* - parent_event_rtp_addr_available: one of the rtp_stream instances managed by the call leg has received an RTP
* address from the MGW. The struct rtp_stream instance is passed as data argument for the event dispatch.
* - parent_event_rtp_complete: one of the rtp_stream instances entered the RTP_STREAM_ST_ESTABLISHED state.
*/
struct call_leg *call_leg_alloc(struct osmo_fsm_inst *parent_fi,
uint32_t parent_event_term,
uint32_t parent_event_rtp_addr_available,
uint32_t parent_event_rtp_complete)
{
struct call_leg *cl;
struct osmo_fsm_inst *fi = osmo_fsm_inst_alloc_child(&call_leg_fsm, parent_fi, parent_event_term);
OSMO_ASSERT(fi);
cl = talloc(fi, struct call_leg);
OSMO_ASSERT(cl);
fi->priv = cl;
*cl = (struct call_leg){
.fi = fi,
.parent_event_rtp_addr_available = parent_event_rtp_addr_available,
.parent_event_rtp_complete = parent_event_rtp_complete,
};
return cl;
}
void call_leg_reparent(struct call_leg *cl,
struct osmo_fsm_inst *new_parent_fi,
uint32_t parent_event_term,
uint32_t parent_event_rtp_addr_available,
uint32_t parent_event_rtp_complete)
{
LOG_CALL_LEG(cl, LOGL_DEBUG, "Reparenting from parent %s to parent %s\n",
cl->fi->proc.parent->name, new_parent_fi->name);
osmo_fsm_inst_change_parent(cl->fi, new_parent_fi, parent_event_term);
talloc_steal(new_parent_fi, cl->fi);
cl->parent_event_rtp_addr_available = parent_event_rtp_addr_available;
cl->parent_event_rtp_complete = parent_event_rtp_complete;
}
static int call_leg_fsm_timer_cb(struct osmo_fsm_inst *fi)
{
struct call_leg *cl = fi->priv;
call_leg_release(cl);
return 0;
}
void call_leg_release(struct call_leg *cl)
{
if (!cl)
return;
if (cl->fi->state == CALL_LEG_ST_RELEASING)
return;
call_leg_state_chg(cl, CALL_LEG_ST_RELEASING);
}
static void call_leg_mgw_endpoint_gone(struct call_leg *cl)
{
struct mgcp_client *mgcp_client;
int i;
/* Put MGCP client back into MGW pool */
mgcp_client = osmo_mgcpc_ep_client(cl->mgw_endpoint);
mgcp_client_pool_put(mgcp_client);
cl->mgw_endpoint = NULL;
for (i = 0; i < ARRAY_SIZE(cl->rtp); i++) {
if (!cl->rtp[i])
continue;
cl->rtp[i]->ci = NULL;
}
}
static void call_leg_fsm_establishing_established(struct osmo_fsm_inst *fi, uint32_t event, void *data)
{
struct call_leg *cl = fi->priv;
struct rtp_stream *rtps;
int i;
bool established;
switch (event) {
case CALL_LEG_EV_RTP_STREAM_ESTABLISHED:
/* An rtp_stream says it is established. If all are now established, change to state
* CALL_LEG_ST_ESTABLISHED. */
established = true;
for (i = 0; i < ARRAY_SIZE(cl->rtp); i++) {
if (!rtp_stream_is_established(cl->rtp[i])) {
established = false;
break;
}
}
if (!established)
break;
if (cl->fi->state != CALL_LEG_ST_ESTABLISHED)
call_leg_state_chg(cl, CALL_LEG_ST_ESTABLISHED);
break;
case CALL_LEG_EV_RTP_STREAM_ADDR_AVAILABLE:
rtps = data;
osmo_fsm_inst_dispatch(fi->proc.parent, cl->parent_event_rtp_addr_available, rtps);
break;
case CALL_LEG_EV_RTP_STREAM_GONE:
call_leg_release(cl);
break;
case CALL_LEG_EV_MGW_ENDPOINT_GONE:
call_leg_mgw_endpoint_gone(cl);
call_leg_release(cl);
break;
default:
OSMO_ASSERT(false);
}
}
void call_leg_fsm_established_onenter(struct osmo_fsm_inst *fi, uint32_t prev_state)
{
struct call_leg *cl = fi->priv;
osmo_fsm_inst_dispatch(fi->proc.parent, cl->parent_event_rtp_complete, cl);
}
void call_leg_fsm_releasing_onenter(struct osmo_fsm_inst *fi, uint32_t prev_state)
{
/* Trigger termination of children FSMs (rtp_stream(s)) before
* terminating ourselves, otherwise we are not able to receive
* CALL_LEG_EV_MGW_ENDPOINT_GONE from cl->mgw_endpoint (call_leg =>
* rtp_stream => mgw_endpoint), because osmo_fsm disabled dispatching
* events to an FSM in process of terminating. */
osmo_fsm_inst_term_children(fi, OSMO_FSM_TERM_PARENT, NULL);
osmo_fsm_inst_term(fi, OSMO_FSM_TERM_REGULAR, NULL);
}
static void call_leg_fsm_releasing(struct osmo_fsm_inst *fi, uint32_t event, void *data)
{
struct call_leg *cl = fi->priv;
switch (event) {
case CALL_LEG_EV_RTP_STREAM_GONE:
/* We're already terminating, child RTP streams will also terminate, there is nothing left to do. */
break;
case CALL_LEG_EV_MGW_ENDPOINT_GONE:
call_leg_mgw_endpoint_gone(cl);
break;
default:
OSMO_ASSERT(false);
}
}
static const struct value_string call_leg_fsm_event_names[] = {
OSMO_VALUE_STRING(CALL_LEG_EV_RTP_STREAM_ADDR_AVAILABLE),
OSMO_VALUE_STRING(CALL_LEG_EV_RTP_STREAM_ESTABLISHED),
OSMO_VALUE_STRING(CALL_LEG_EV_RTP_STREAM_GONE),
OSMO_VALUE_STRING(CALL_LEG_EV_MGW_ENDPOINT_GONE),
{}
};
#define S(x) (1 << (x))
static const struct osmo_fsm_state call_leg_fsm_states[] = {
[CALL_LEG_ST_ESTABLISHING] = {
.name = "ESTABLISHING",
.in_event_mask = 0
| S(CALL_LEG_EV_RTP_STREAM_ADDR_AVAILABLE)
| S(CALL_LEG_EV_RTP_STREAM_ESTABLISHED)
| S(CALL_LEG_EV_RTP_STREAM_GONE)
| S(CALL_LEG_EV_MGW_ENDPOINT_GONE)
,
.out_state_mask = 0
| S(CALL_LEG_ST_ESTABLISHED)
| S(CALL_LEG_ST_RELEASING)
,
.action = call_leg_fsm_establishing_established,
},
[CALL_LEG_ST_ESTABLISHED] = {
.name = "ESTABLISHED",
.in_event_mask = 0
| S(CALL_LEG_EV_RTP_STREAM_ADDR_AVAILABLE)
| S(CALL_LEG_EV_RTP_STREAM_ESTABLISHED)
| S(CALL_LEG_EV_RTP_STREAM_GONE)
| S(CALL_LEG_EV_MGW_ENDPOINT_GONE)
,
.out_state_mask = 0
| S(CALL_LEG_ST_ESTABLISHING)
| S(CALL_LEG_ST_RELEASING)
,
.onenter = call_leg_fsm_established_onenter,
.action = call_leg_fsm_establishing_established, /* same action function as above */
},
[CALL_LEG_ST_RELEASING] = {
.name = "RELEASING",
.in_event_mask = 0
| S(CALL_LEG_EV_RTP_STREAM_GONE)
| S(CALL_LEG_EV_MGW_ENDPOINT_GONE)
,
.onenter = call_leg_fsm_releasing_onenter,
.action = call_leg_fsm_releasing,
},
};
static struct osmo_fsm call_leg_fsm = {
.name = "call_leg",
.states = call_leg_fsm_states,
.num_states = ARRAY_SIZE(call_leg_fsm_states),
.log_subsys = DCC,
.event_names = call_leg_fsm_event_names,
.timer_cb = call_leg_fsm_timer_cb,
};
const struct value_string rtp_direction_names[] = {
OSMO_VALUE_STRING(RTP_TO_RAN),
OSMO_VALUE_STRING(RTP_TO_CN),
{}
};
int call_leg_ensure_rtp_alloc(struct call_leg *cl, enum rtp_direction dir, uint32_t call_id, struct gsm_trans *for_trans)
{
if (cl->rtp[dir])
return 0;
if (!cl->mgw_endpoint) {
struct mgcp_client *mgcp_client = mgcp_client_pool_get(gsmnet->mgw.mgw_pool);
if (!mgcp_client) {
LOG_CALL_LEG(cl, LOGL_ERROR,
"cannot ensure MGW endpoint -- no MGW configured, check configuration!\n");
return -ENODEV;
}
cl->mgw_endpoint = osmo_mgcpc_ep_alloc(cl->fi, CALL_LEG_EV_MGW_ENDPOINT_GONE,
mgcp_client, gsmnet->mgw.tdefs, cl->fi->id,
"%s", mgcp_client_rtpbridge_wildcard(mgcp_client));
}
if (!cl->mgw_endpoint) {
LOG_CALL_LEG(cl, LOGL_ERROR, "failed to setup MGW endpoint\n");
return -EIO;
}
cl->rtp[dir] = rtp_stream_alloc(cl->fi, CALL_LEG_EV_RTP_STREAM_GONE, CALL_LEG_EV_RTP_STREAM_ADDR_AVAILABLE,
CALL_LEG_EV_RTP_STREAM_ESTABLISHED, dir, call_id, for_trans);
OSMO_ASSERT(cl->rtp[dir]);
return 0;
}
struct osmo_sockaddr_str *call_leg_local_ip(struct call_leg *cl, enum rtp_direction dir)
{
struct rtp_stream *rtps;
if (!cl)
return NULL;
rtps = cl->rtp[dir];
if (!rtps)
return NULL;
if (!osmo_sockaddr_str_is_nonzero(&rtps->local))
return NULL;
return &rtps->local;
}
/* Make sure an MGW endpoint CI is set up for an RTP connection.
* This is the one-stop for all to either completely set up a new endpoint connection, or to modify an existing one.
* If not yet present, allocate the rtp_stream for the given direction.
* Then, call rtp_stream_set_codecs() if codecs_if_known is non-NULL, and/or rtp_stream_set_remote_addr() if
* remote_addr_if_known is non-NULL.
* Finally make sure that a CRCX is sent out for this direction, if this has not already happened.
* If the CRCX has already happened but new codec / remote_addr data was passed, call rtp_stream_commit() to trigger an
* MDCX.
*/
int call_leg_ensure_ci(struct call_leg *cl, enum rtp_direction dir, uint32_t call_id, struct gsm_trans *for_trans,
const struct sdp_audio_codecs *codecs_if_known,
const struct osmo_sockaddr_str *remote_addr_if_known)
{
if (call_leg_ensure_rtp_alloc(cl, dir, call_id, for_trans))
return -EIO;
rtp_stream_set_mode(cl->rtp[dir], cl->crcx_conn_mode[dir]);
if (dir == RTP_TO_RAN && cl->ran_peer_supports_osmux) {
cl->rtp[dir]->use_osmux = true;
cl->rtp[dir]->remote_osmux_cid = -1; /* wildcard */
}
if (codecs_if_known)
rtp_stream_set_codecs(cl->rtp[dir], codecs_if_known);
if (remote_addr_if_known && osmo_sockaddr_str_is_nonzero(remote_addr_if_known))
rtp_stream_set_remote_addr(cl->rtp[dir], remote_addr_if_known);
return rtp_stream_ensure_ci(cl->rtp[dir], cl->mgw_endpoint);
}
int call_leg_local_bridge(struct call_leg *cl1, uint32_t call_id1, struct gsm_trans *trans1,
struct call_leg *cl2, uint32_t call_id2, struct gsm_trans *trans2)
{
struct sdp_audio_codecs *cn_codecs = NULL;
cl1->local_bridge = cl2;
cl2->local_bridge = cl1;
/* Marry the two CN sides of the call legs. Call establishment should have made all efforts for these to be
* compatible. However, for local bridging, the codecs and payload type numbers must be exactly identical on
* both sides. Both sides may so far have different payload type numbers or slightly differing codecs, but it
* will only work when the SDP on the RTP_TO_CN sides of the call legs talk the same payload type numbers.
* So, simply take the SDP from one RTP_TO_CN side, and overwrite the other RTP_TO_CN side's SDP with it.
* If all goes to plan, the codecs will be identical, or possibly the MGW will do a conversion like AMR-BE to
* AMR-OA. In the worst case, the other call leg cannot transcode, and the call fails -- because codec
* negotiation did not do a good enough job.
*
* Copy one call leg's CN config to the other:
*
* call leg 1 call leg 2
* ---MGW-ep------- ---MGW-ep-------
* RAN CN CN RAN
* AMR:112 AMR:112 AMR:96 AMR:96
* |
* +-------+
* |
* V
* AMR:112 AMR:112 AMR:112 AMR:96
* ^MGW-endpoint converts payload type numbers between 112 and 96.
*/
if (cl1->rtp[RTP_TO_CN] && cl1->rtp[RTP_TO_CN]->codecs_known)
cn_codecs = &cl1->rtp[RTP_TO_CN]->codecs;
else if (cl2->rtp[RTP_TO_CN] && cl2->rtp[RTP_TO_CN]->codecs_known)
cn_codecs = &cl2->rtp[RTP_TO_CN]->codecs;
if (!cn_codecs) {
LOG_CALL_LEG(cl1, LOGL_ERROR, "RAN-side CN stream codec is not known, not ready for bridging\n");
LOG_CALL_LEG(cl2, LOGL_ERROR, "RAN-side CN stream codec is not known, not ready for bridging\n");
return -EINVAL;
}
call_leg_ensure_ci(cl1, RTP_TO_CN, call_id1, trans1,
cn_codecs, &cl2->rtp[RTP_TO_CN]->local);
call_leg_ensure_ci(cl2, RTP_TO_CN, call_id2, trans2,
cn_codecs, &cl1->rtp[RTP_TO_CN]->local);
return 0;
}