diff options
Diffstat (limited to 'src/libmsc/rtp_stream.c')
-rw-r--r-- | src/libmsc/rtp_stream.c | 389 |
1 files changed, 389 insertions, 0 deletions
diff --git a/src/libmsc/rtp_stream.c b/src/libmsc/rtp_stream.c new file mode 100644 index 000000000..fa31ee7e6 --- /dev/null +++ b/src/libmsc/rtp_stream.c @@ -0,0 +1,389 @@ +/* + * (C) 2019 by sysmocom - s.m.f.c. GmbH <info@sysmocom.de> + * All Rights Reserved + * + * SPDX-License-Identifier: AGPL-3.0+ + * + * Author: Neels Hofmeyr + * + * 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/>. + */ + +#include <osmocom/core/fsm.h> + +#include <osmocom/mgcp_client/mgcp_client_endpoint_fsm.h> + +#include <osmocom/msc/debug.h> +#include <osmocom/msc/transaction.h> +#include <osmocom/msc/call_leg.h> +#include <osmocom/msc/rtp_stream.h> + +#define LOG_RTPS(rtps, level, fmt, args...) \ + LOGPFSML(rtps->fi, level, fmt, ##args) + +enum rtp_stream_event { + RTP_STREAM_EV_CRCX_OK, + RTP_STREAM_EV_CRCX_FAIL, + RTP_STREAM_EV_MDCX_OK, + RTP_STREAM_EV_MDCX_FAIL, +}; + +enum rtp_stream_state { + RTP_STREAM_ST_UNINITIALIZED, + RTP_STREAM_ST_ESTABLISHING, + RTP_STREAM_ST_ESTABLISHED, + RTP_STREAM_ST_DISCARDING, +}; + +static struct osmo_fsm rtp_stream_fsm; + +static struct osmo_tdef_state_timeout rtp_stream_fsm_timeouts[32] = { + [RTP_STREAM_ST_ESTABLISHING] = { .T = -2 }, +}; + +#define rtp_stream_state_chg(rtps, state) \ + osmo_tdef_fsm_inst_state_chg((rtps)->fi, state, rtp_stream_fsm_timeouts, g_mgw_tdefs, 5) + +static __attribute__((constructor)) void rtp_stream_init() +{ + OSMO_ASSERT(osmo_fsm_register(&rtp_stream_fsm) == 0); +} + +void rtp_stream_update_id(struct rtp_stream *rtps) +{ + char buf[256]; + char *p; + struct osmo_strbuf sb = { .buf = buf, .len = sizeof(buf) }; + OSMO_STRBUF_PRINTF(sb, "%s", rtps->fi->proc.parent->id); + if (rtps->for_trans) + OSMO_STRBUF_PRINTF(sb, ":trans-%u", rtps->for_trans->transaction_id); + OSMO_STRBUF_PRINTF(sb, ":call-%u", rtps->call_id); + OSMO_STRBUF_PRINTF(sb, ":%s", rtp_direction_name(rtps->dir)); + if (!osmo_mgcpc_ep_ci_id(rtps->ci)) { + OSMO_STRBUF_PRINTF(sb, ":no-CI"); + } else { + OSMO_STRBUF_PRINTF(sb, ":CI-%s", osmo_mgcpc_ep_ci_id(rtps->ci)); + if (!osmo_sockaddr_str_is_set(&rtps->remote)) + OSMO_STRBUF_PRINTF(sb, ":no-remote-port"); + else if (!rtps->remote_sent_to_mgw) + OSMO_STRBUF_PRINTF(sb, ":remote-port-not-sent"); + if (!rtps->codec_known) + OSMO_STRBUF_PRINTF(sb, ":no-codec"); + else if (!rtps->codec_sent_to_mgw) + OSMO_STRBUF_PRINTF(sb, ":codec-not-sent"); + } + if (osmo_sockaddr_str_is_set(&rtps->local)) + OSMO_STRBUF_PRINTF(sb, ":local-%s-%u", rtps->local.ip, rtps->local.port); + if (osmo_sockaddr_str_is_set(&rtps->remote)) + OSMO_STRBUF_PRINTF(sb, ":remote-%s-%u", rtps->remote.ip, rtps->remote.port); + + /* Replace any dots in the IP address, dots not allowed as FSM instance name */ + for (p = buf; *p; p++) + if (*p == '.') + *p = '-'; + + osmo_fsm_inst_update_id_f(rtps->fi, "%s", buf); +} + +/* Allocate RTP stream under a call leg. This is one RTP connection from some remote entity with address and port to a + * local RTP address and port. call_id is stored for sending in MGCP transactions and as logging context. for_trans is + * optional, merely stored for reference by callers, and appears as log context if not NULL. */ +struct rtp_stream *rtp_stream_alloc(struct call_leg *parent_call_leg, enum rtp_direction dir, + uint32_t call_id, struct gsm_trans *for_trans) +{ + struct osmo_fsm_inst *fi; + struct rtp_stream *rtps; + + fi = osmo_fsm_inst_alloc_child(&rtp_stream_fsm, parent_call_leg->fi, CALL_LEG_EV_RTP_STREAM_GONE); + OSMO_ASSERT(fi); + + rtps = talloc_zero(fi, struct rtp_stream); + OSMO_ASSERT(rtps); + fi->priv = rtps; + *rtps = (struct rtp_stream){ + .fi = fi, + .parent_call_leg = parent_call_leg, + .call_id = call_id, + .for_trans = for_trans, + .dir = dir, + }; + + rtp_stream_update_id(rtps); + + return rtps; +} + +static void check_established(struct rtp_stream *rtps) +{ + if (rtps->fi->state != RTP_STREAM_ST_ESTABLISHED + && osmo_sockaddr_str_is_set(&rtps->local) + && osmo_sockaddr_str_is_set(&rtps->remote) + && rtps->remote_sent_to_mgw + && rtps->codec_known) + rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHED); +} + +static void rtp_stream_fsm_establishing_established(struct osmo_fsm_inst *fi, uint32_t event, void *data) +{ + struct rtp_stream *rtps = fi->priv; + const struct mgcp_conn_peer *crcx_info; + switch (event) { + case RTP_STREAM_EV_CRCX_OK: + crcx_info = osmo_mgcpc_ep_ci_get_rtp_info(rtps->ci); + osmo_sockaddr_str_from_str(&rtps->local, crcx_info->addr, crcx_info->port); + rtp_stream_update_id(rtps); + osmo_fsm_inst_dispatch(fi->proc.parent, CALL_LEG_EV_RTP_STREAM_ADDR_AVAILABLE, rtps); + check_established(rtps); + + if ((!rtps->remote_sent_to_mgw || !rtps->codec_sent_to_mgw) + && osmo_sockaddr_str_is_set(&rtps->remote) + && rtps->codec_known) { + LOG_RTPS(rtps, LOGL_DEBUG, + "local ip:port set;%s%s triggering MDCX to send the new settings\n", + (!rtps->remote_sent_to_mgw)? " remote ip:port not yet sent," : "", + (!rtps->codec_sent_to_mgw)? " codec not yet sent," : ""); + rtp_stream_do_mdcx(rtps); + } + return; + + case RTP_STREAM_EV_MDCX_OK: + rtp_stream_update_id(rtps); + check_established(rtps); + return; + + case RTP_STREAM_EV_CRCX_FAIL: + case RTP_STREAM_EV_MDCX_FAIL: + rtps->remote_sent_to_mgw = false; + rtps->codec_sent_to_mgw = false; + rtp_stream_update_id(rtps); + rtp_stream_state_chg(rtps, RTP_STREAM_ST_DISCARDING); + return; + + default: + OSMO_ASSERT(false); + }; +} + +void rtp_stream_fsm_established_onenter(struct osmo_fsm_inst *fi, uint32_t prev_state) +{ + struct rtp_stream *rtps = fi->priv; + osmo_fsm_inst_dispatch(fi->proc.parent, CALL_LEG_EV_RTP_STREAM_ESTABLISHED, rtps); +} + +static int rtp_stream_fsm_timer_cb(struct osmo_fsm_inst *fi) +{ + struct rtp_stream *rtps = fi->priv; + rtp_stream_state_chg(rtps, RTP_STREAM_ST_DISCARDING); + return 0; +} + +static void rtp_stream_fsm_cleanup(struct osmo_fsm_inst *fi, enum osmo_fsm_term_cause cause) +{ + struct rtp_stream *rtps = fi->priv; + if (rtps->ci) { + osmo_mgcpc_ep_ci_dlcx(rtps->ci); + rtps->ci = NULL; + } +} + +void rtp_stream_fsm_discarding_onenter(struct osmo_fsm_inst *fi, uint32_t prev_state) +{ + osmo_fsm_inst_term(fi, OSMO_FSM_TERM_REGULAR, NULL); +} + +static const struct value_string rtp_stream_fsm_event_names[] = { + OSMO_VALUE_STRING(RTP_STREAM_EV_CRCX_OK), + OSMO_VALUE_STRING(RTP_STREAM_EV_CRCX_FAIL), + OSMO_VALUE_STRING(RTP_STREAM_EV_MDCX_OK), + OSMO_VALUE_STRING(RTP_STREAM_EV_MDCX_FAIL), + {} +}; + +#define S(x) (1 << (x)) + +static const struct osmo_fsm_state rtp_stream_fsm_states[] = { + [RTP_STREAM_ST_UNINITIALIZED] = { + .name = "UNINITIALIZED", + .out_state_mask = 0 + | S(RTP_STREAM_ST_ESTABLISHING) + | S(RTP_STREAM_ST_DISCARDING) + , + }, + [RTP_STREAM_ST_ESTABLISHING] = { + .name = "ESTABLISHING", + .in_event_mask = 0 + | S(RTP_STREAM_EV_CRCX_OK) + | S(RTP_STREAM_EV_CRCX_FAIL) + | S(RTP_STREAM_EV_MDCX_OK) + | S(RTP_STREAM_EV_MDCX_FAIL) + , + .out_state_mask = 0 + | S(RTP_STREAM_ST_ESTABLISHED) + | S(RTP_STREAM_ST_DISCARDING) + , + .action = rtp_stream_fsm_establishing_established, + }, + [RTP_STREAM_ST_ESTABLISHED] = { + .name = "ESTABLISHED", + .out_state_mask = 0 + | S(RTP_STREAM_ST_ESTABLISHING) + | S(RTP_STREAM_ST_DISCARDING) + , + .onenter = rtp_stream_fsm_established_onenter, + .action = rtp_stream_fsm_establishing_established, + }, + [RTP_STREAM_ST_DISCARDING] = { + .name = "DISCARDING", + .onenter = rtp_stream_fsm_discarding_onenter, + .out_state_mask = 0 + | S(RTP_STREAM_ST_DISCARDING) + , + }, +}; + +static struct osmo_fsm rtp_stream_fsm = { + .name = "rtp_stream", + .states = rtp_stream_fsm_states, + .num_states = ARRAY_SIZE(rtp_stream_fsm_states), + .log_subsys = DCC, + .event_names = rtp_stream_fsm_event_names, + .timer_cb = rtp_stream_fsm_timer_cb, + .cleanup = rtp_stream_fsm_cleanup, +}; + +static int rtp_stream_do_mgcp_verb(struct rtp_stream *rtps, enum mgcp_verb verb, uint32_t ok_event, uint32_t fail_event) +{ + struct mgcp_conn_peer verb_info; + + if (!rtps->ci) { + LOG_RTPS(rtps, LOGL_ERROR, "Cannot send %s, no endpoint CI allocated\n", osmo_mgcp_verb_name(verb)); + return -EINVAL; + } + + verb_info = (struct mgcp_conn_peer){ + .call_id = rtps->call_id, + .ptime = 20, + }; + + if (verb == MGCP_VERB_CRCX) + verb_info.conn_mode = rtps->crcx_conn_mode; + + if (rtps->codec_known) { + verb_info.codecs[0] = rtps->codec; + verb_info.codecs_len = 1; + rtps->codec_sent_to_mgw = true; + } + if (osmo_sockaddr_str_is_set(&rtps->remote)) { + int rc = osmo_strlcpy(verb_info.addr, rtps->remote.ip, sizeof(verb_info.addr)); + if (rc <= 0 || rc >= sizeof(verb_info.addr)) { + LOG_RTPS(rtps, LOGL_ERROR, "Failure to write IP address to MGCP message (rc=%d)\n", rc); + return -ENOSPC; + } + verb_info.port = rtps->remote.port; + rtps->remote_sent_to_mgw = true; + } + + osmo_mgcpc_ep_ci_request(rtps->ci, verb, &verb_info, rtps->fi, ok_event, fail_event, NULL); + return 0; +} + +int rtp_stream_ensure_ci(struct rtp_stream *rtps, struct osmo_mgcpc_ep *at_endpoint) +{ + if (rtps->ci) + return rtp_stream_commit(rtps); + + rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHING); + + rtps->ci = osmo_mgcpc_ep_ci_add(at_endpoint, "%s", rtp_direction_name(rtps->dir)); + if (!rtps->ci) + return -ENODEV; + + return rtp_stream_do_mgcp_verb(rtps, MGCP_VERB_CRCX, RTP_STREAM_EV_CRCX_OK, RTP_STREAM_EV_CRCX_FAIL); +} + +int rtp_stream_do_mdcx(struct rtp_stream *rtps) +{ + return rtp_stream_do_mgcp_verb(rtps, MGCP_VERB_MDCX, RTP_STREAM_EV_MDCX_OK, RTP_STREAM_EV_MDCX_FAIL); +} + +void rtp_stream_release(struct rtp_stream *rtps) +{ + if (!rtps) + return; + + rtp_stream_state_chg(rtps, RTP_STREAM_ST_DISCARDING); +} + +/* After setting up a remote RTP address or a new codec, call this to trigger an MDCX. + * The MDCX will only trigger if all data needed by an endpoint is available (both RTP address and codec) and if at + * least one of them has not yet been sent to the MGW in a previous CRCX or MDCX. */ +int rtp_stream_commit(struct rtp_stream *rtps) +{ + if (!rtps->ci) { + LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no MGW endpoint CI set up\n"); + return -1; + } + if (!osmo_sockaddr_str_is_set(&rtps->remote)) { + LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no remote RTP address known\n"); + return -1; + } + if (!rtps->codec_known) { + LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: no codec known\n"); + return -1; + } + if (rtps->remote_sent_to_mgw && rtps->codec_sent_to_mgw) { + LOG_RTPS(rtps, LOGL_DEBUG, "Not committing: both remote RTP address and codec already set up at MGW\n"); + return 0; + } + + LOG_RTPS(rtps, LOGL_DEBUG, "Committing: Tx MDCX to update the MGW: updating%s%s\n", + rtps->remote_sent_to_mgw ? "" : " remote-RTP-IP-port", + rtps->codec_sent_to_mgw ? "" : " codec"); + return rtp_stream_do_mdcx(rtps); +} + +void rtp_stream_set_codec(struct rtp_stream *rtps, enum mgcp_codecs codec) +{ + if (rtps->fi->state == RTP_STREAM_ST_ESTABLISHED) + rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHING); + LOG_RTPS(rtps, LOGL_DEBUG, "setting codec to %s\n", osmo_mgcpc_codec_name(codec)); + rtps->codec = codec; + rtps->codec_known = true; + rtps->codec_sent_to_mgw = false; + rtp_stream_update_id(rtps); +} + +void rtp_stream_set_remote_addr(struct rtp_stream *rtps, const struct osmo_sockaddr_str *r) +{ + if (rtps->fi->state == RTP_STREAM_ST_ESTABLISHED) + rtp_stream_state_chg(rtps, RTP_STREAM_ST_ESTABLISHING); + LOG_RTPS(rtps, LOGL_DEBUG, "setting remote addr to " OSMO_SOCKADDR_STR_FMT "\n", OSMO_SOCKADDR_STR_FMT_ARGS(r)); + rtps->remote = *r; + rtps->remote_sent_to_mgw = false; + rtp_stream_update_id(rtps); +} + +bool rtp_stream_is_established(struct rtp_stream *rtps) +{ + if (!rtps) + return false; + if (!rtps->fi) + return false; + if (rtps->fi->state != RTP_STREAM_ST_ESTABLISHED) + return false; + if (!rtps->remote_sent_to_mgw + || !rtps->codec_sent_to_mgw) + return false; + return true; +} |