1657 lines
54 KiB
C
1657 lines
54 KiB
C
/* A Media Gateway Control Protocol Media Gateway: RFC 3435 */
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/* The protocol implementation */
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/*
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* (C) 2009-2012 by Holger Hans Peter Freyther <zecke@selfish.org>
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* (C) 2009-2012 by On-Waves
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* All Rights Reserved
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU Affero General Public License as published by
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* the Free Software Foundation; either version 3 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Affero General Public License for more details.
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*
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* You should have received a copy of the GNU Affero General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*
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*/
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#include <string.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <errno.h>
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#include <time.h>
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#include <limits.h>
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#include <arpa/inet.h>
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#include <osmocom/core/msgb.h>
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#include <osmocom/core/select.h>
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#include <osmocom/core/socket.h>
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#include <osmocom/core/byteswap.h>
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#include <osmocom/netif/rtp.h>
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#include <osmocom/netif/amr.h>
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#include <osmocom/mgcp/mgcp.h>
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#include <osmocom/mgcp/mgcp_common.h>
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#include <osmocom/mgcp/mgcp_network.h>
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#include <osmocom/mgcp/mgcp_protocol.h>
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#include <osmocom/mgcp/mgcp_stat.h>
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#include <osmocom/mgcp/osmux.h>
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#include <osmocom/mgcp/mgcp_conn.h>
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#include <osmocom/mgcp/mgcp_endp.h>
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#include <osmocom/mgcp/mgcp_trunk.h>
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#include <osmocom/mgcp/mgcp_codec.h>
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#include <osmocom/mgcp/debug.h>
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#include <osmocom/codec/codec.h>
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#include <osmocom/mgcp/mgcp_e1.h>
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#define RTP_SEQ_MOD (1 << 16)
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#define RTP_MAX_DROPOUT 3000
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#define RTP_MAX_MISORDER 100
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enum rtp_proto {
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MGCP_PROTO_RTP,
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MGCP_PROTO_RTCP,
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};
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static void rtpconn_rate_ctr_add(struct mgcp_conn_rtp *conn_rtp, struct mgcp_endpoint *endp,
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int id, int inc)
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{
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struct rate_ctr_group *conn_stats = conn_rtp->rate_ctr_group;
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struct rate_ctr_group *mgw_stats = endp->trunk->ratectr.all_rtp_conn_stats;
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/* add to both the per-connection and the global stats */
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rate_ctr_add(&conn_stats->ctr[id], inc);
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rate_ctr_add(&mgw_stats->ctr[id], inc);
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}
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static void rtpconn_rate_ctr_inc(struct mgcp_conn_rtp *conn_rtp, struct mgcp_endpoint *endp, int id)
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{
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rtpconn_rate_ctr_add(conn_rtp, endp, id, 1);
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}
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static int rx_rtp(struct msgb *msg);
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static bool addr_is_any(struct osmo_sockaddr *osa) {
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if (osa->u.sa.sa_family == AF_INET6) {
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struct in6_addr ip6_any = IN6ADDR_ANY_INIT;
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return memcmp(&osa->u.sin6.sin6_addr,
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&ip6_any, sizeof(ip6_any)) == 0;
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} else {
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return osa->u.sin.sin_addr.s_addr == 0;
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}
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}
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/*! Determine the local rtp bind IP-address.
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* \param[out] addr caller provided memory to store the resulting IP-Address.
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* \param[in] endp mgcp endpoint, that holds a copy of the VTY parameters.
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*
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* The local bind IP-address is automatically selected by probing the
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* IP-Address of the interface that is pointing towards the remote IP-Address,
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* if no remote IP-Address is known yet, the statically configured
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* IP-Addresses are used as fallback. */
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void mgcp_get_local_addr(char *addr, struct mgcp_conn_rtp *conn)
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{
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struct mgcp_endpoint *endp;
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char ipbuf[INET6_ADDRSTRLEN];
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int rc;
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endp = conn->conn->endp;
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bool rem_addr_set = !addr_is_any(&conn->end.addr);
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char *bind_addr;
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/* Try probing the local IP-Address */
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if (endp->cfg->net_ports.bind_addr_probe && rem_addr_set) {
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rc = osmo_sock_local_ip(addr, osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf));
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if (rc < 0)
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LOGPCONN(conn->conn, DRTP, LOGL_ERROR,
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"local interface auto detection failed, using configured addresses...\n");
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else {
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LOGPCONN(conn->conn, DRTP, LOGL_DEBUG,
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"selected local rtp bind ip %s by probing using remote ip %s\n",
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addr, osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf));
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return;
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}
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}
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/* Select from preconfigured IP-Addresses. We don't have bind_addr for Osmux (yet?). */
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if (rem_addr_set) {
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/* Check there is a bind IP for the RTP traffic configured,
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* if so, use that IP-Address */
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bind_addr = conn->end.addr.u.sa.sa_family == AF_INET6 ?
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endp->cfg->net_ports.bind_addr_v6 :
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endp->cfg->net_ports.bind_addr_v4;
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} else {
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/* Choose any of the bind addresses, preferring v6 over v4 */
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bind_addr = endp->cfg->net_ports.bind_addr_v6;
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if (!bind_addr)
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bind_addr = endp->cfg->net_ports.bind_addr_v4;
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}
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if (bind_addr) {
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LOGPCONN(conn->conn, DRTP, LOGL_DEBUG,
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"using configured rtp bind ip as local bind ip %s\n",
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bind_addr);
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} else {
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/* No specific bind IP is configured for the RTP traffic, so
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* assume the IP where we listen for incoming MGCP messages
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* as bind IP */
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bind_addr = endp->cfg->source_addr;
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LOGPCONN(conn->conn, DRTP, LOGL_DEBUG,
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"using mgcp bind ip as local rtp bind ip: %s\n", bind_addr);
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}
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osmo_strlcpy(addr, bind_addr, INET6_ADDRSTRLEN);
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}
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/* This does not need to be a precision timestamp and
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* is allowed to wrap quite fast. The returned value is
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* 1/codec_rate seconds. */
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static uint32_t get_current_ts(unsigned codec_rate)
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{
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struct timespec tp;
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uint64_t ret;
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if (!codec_rate)
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return 0;
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memset(&tp, 0, sizeof(tp));
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if (clock_gettime(CLOCK_MONOTONIC, &tp) != 0)
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LOGP(DRTP, LOGL_NOTICE, "Getting the clock failed.\n");
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/* convert it to 1/unit seconds */
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ret = tp.tv_sec;
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ret *= codec_rate;
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ret += (int64_t) tp.tv_nsec * codec_rate / 1000 / 1000 / 1000;
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return ret;
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}
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/*! send udp packet.
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* \param[in] fd associated file descriptor.
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* \param[in] addr destination ip-address.
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* \param[in] port destination UDP port (network byte order).
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* \param[in] buf buffer that holds the data to be send.
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* \param[in] len length of the data to be sent.
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* \returns bytes sent, -1 on error. */
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int mgcp_udp_send(int fd, struct osmo_sockaddr *addr, int port, char *buf, int len)
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{
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char ipbuf[INET6_ADDRSTRLEN];
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size_t addr_len;
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bool is_ipv6 = addr->u.sa.sa_family == AF_INET6;
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LOGP(DRTP, LOGL_DEBUG,
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"sending %i bytes length packet to %s:%u ...\n", len,
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osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
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ntohs(port));
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if (is_ipv6) {
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addr->u.sin6.sin6_port = port;
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addr_len = sizeof(addr->u.sin6);
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} else {
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addr->u.sin.sin_port = port;
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addr_len = sizeof(addr->u.sin);
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}
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return sendto(fd, buf, len, 0, &addr->u.sa, addr_len);
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}
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/*! send RTP dummy packet (to keep NAT connection open).
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* \param[in] endp mcgp endpoint that holds the RTP connection.
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* \param[in] conn associated RTP connection.
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* \returns bytes sent, -1 on error. */
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int mgcp_send_dummy(struct mgcp_endpoint *endp, struct mgcp_conn_rtp *conn)
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{
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static char buf[] = { MGCP_DUMMY_LOAD };
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int rc;
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int was_rtcp = 0;
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OSMO_ASSERT(endp);
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OSMO_ASSERT(conn);
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LOGPCONN(conn->conn, DRTP, LOGL_DEBUG,"sending dummy packet... %s\n",
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mgcp_conn_dump(conn->conn));
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rc = mgcp_udp_send(conn->end.rtp.fd, &conn->end.addr,
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conn->end.rtp_port, buf, 1);
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if (rc == -1)
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goto failed;
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if (endp->trunk->omit_rtcp)
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return rc;
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was_rtcp = 1;
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rc = mgcp_udp_send(conn->end.rtcp.fd, &conn->end.addr,
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conn->end.rtcp_port, buf, 1);
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if (rc >= 0)
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return rc;
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failed:
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LOGPCONN(conn->conn, DRTP, LOGL_ERROR,
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"Failed to send dummy %s packet.\n",
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was_rtcp ? "RTCP" : "RTP");
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return -1;
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}
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/* Compute timestamp alignment error */
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static int32_t ts_alignment_error(struct mgcp_rtp_stream_state *sstate,
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int ptime, uint32_t timestamp)
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{
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int32_t timestamp_delta;
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if (ptime == 0)
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return 0;
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/* Align according to: T - Tlast = k * Tptime */
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timestamp_delta = timestamp - sstate->last_timestamp;
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return timestamp_delta % ptime;
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}
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/* Check timestamp and sequence number for plausibility */
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static int check_rtp_timestamp(struct mgcp_endpoint *endp,
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struct mgcp_rtp_state *state,
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struct mgcp_rtp_stream_state *sstate,
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struct mgcp_rtp_end *rtp_end,
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struct osmo_sockaddr *addr,
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uint16_t seq, uint32_t timestamp,
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const char *text, int32_t * tsdelta_out)
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{
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int32_t tsdelta;
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int32_t timestamp_error;
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char ipbuf[INET6_ADDRSTRLEN];
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/* Not fully intialized, skip */
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if (sstate->last_tsdelta == 0 && timestamp == sstate->last_timestamp)
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return 0;
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if (seq == sstate->last_seq) {
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if (timestamp != sstate->last_timestamp) {
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rate_ctr_inc(sstate->err_ts_ctr);
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LOGPENDP(endp, DRTP, LOGL_ERROR,
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"The %s timestamp delta is != 0 but the sequence "
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"number %d is the same, "
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"TS offset: %d, SeqNo offset: %d "
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"on SSRC: %u timestamp: %u "
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"from %s:%d\n",
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text, seq,
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state->patch.timestamp_offset, state->patch.seq_offset,
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sstate->ssrc, timestamp,
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osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
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osmo_sockaddr_port(&addr->u.sa));
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}
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return 0;
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}
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tsdelta =
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(int32_t)(timestamp - sstate->last_timestamp) /
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(int16_t)(seq - sstate->last_seq);
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if (tsdelta == 0) {
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/* Don't update *tsdelta_out */
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LOGPENDP(endp, DRTP, LOGL_NOTICE,
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"The %s timestamp delta is %d "
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"on SSRC: %u timestamp: %u "
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"from %s:%d\n",
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text, tsdelta, sstate->ssrc, timestamp,
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osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
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osmo_sockaddr_port(&addr->u.sa));
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return 0;
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}
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if (sstate->last_tsdelta != tsdelta) {
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if (sstate->last_tsdelta) {
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LOGPENDP(endp, DRTP, LOGL_INFO,
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"The %s timestamp delta changes from %d to %d "
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"on SSRC: %u timestamp: %u from %s:%d\n",
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text, sstate->last_tsdelta, tsdelta,
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sstate->ssrc, timestamp,
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osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
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osmo_sockaddr_port(&addr->u.sa));
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}
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}
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if (tsdelta_out)
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*tsdelta_out = tsdelta;
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timestamp_error =
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ts_alignment_error(sstate, state->packet_duration, timestamp);
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if (timestamp_error) {
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rate_ctr_inc(sstate->err_ts_ctr);
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LOGPENDP(endp, DRTP, LOGL_NOTICE,
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"The %s timestamp has an alignment error of %d "
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"on SSRC: %u "
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"SeqNo delta: %d, TS delta: %d, dTS/dSeq: %d "
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"from %s:%d. ptime: %d\n",
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text, timestamp_error,
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sstate->ssrc,
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(int16_t)(seq - sstate->last_seq),
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(int32_t)(timestamp - sstate->last_timestamp),
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tsdelta,
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osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
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osmo_sockaddr_port(&addr->u.sa),
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state->packet_duration);
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}
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return 1;
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}
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/* Set the timestamp offset according to the packet duration. */
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static int adjust_rtp_timestamp_offset(struct mgcp_endpoint *endp,
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struct mgcp_rtp_state *state,
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struct mgcp_rtp_end *rtp_end,
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struct osmo_sockaddr *addr,
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int16_t delta_seq, uint32_t in_timestamp)
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{
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int32_t tsdelta = state->packet_duration;
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int timestamp_offset;
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uint32_t out_timestamp;
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char ipbuf[INET6_ADDRSTRLEN];
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if (tsdelta == 0) {
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tsdelta = state->out_stream.last_tsdelta;
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if (tsdelta != 0) {
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LOGPENDP(endp, DRTP, LOGL_NOTICE,
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"A fixed packet duration is not available, "
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"using last output timestamp delta instead: %d "
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"from %s:%d\n", tsdelta,
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osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
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osmo_sockaddr_port(&addr->u.sa));
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} else {
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tsdelta = rtp_end->codec->rate * 20 / 1000;
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LOGPENDP(endp, DRTP, LOGL_NOTICE,
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"Fixed packet duration and last timestamp delta "
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"are not available, "
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"using fixed 20ms instead: %d "
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"from %s:%d\n", tsdelta,
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osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
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osmo_sockaddr_port(&addr->u.sa));
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}
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}
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out_timestamp = state->out_stream.last_timestamp + delta_seq * tsdelta;
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timestamp_offset = out_timestamp - in_timestamp;
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if (state->patch.timestamp_offset != timestamp_offset) {
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state->patch.timestamp_offset = timestamp_offset;
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LOGPENDP(endp, DRTP, LOGL_NOTICE,
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"Timestamp offset change on SSRC: %u "
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"SeqNo delta: %d, TS offset: %d, "
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"from %s:%d\n", state->in_stream.ssrc,
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delta_seq, state->patch.timestamp_offset,
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osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
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osmo_sockaddr_port(&addr->u.sa));
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}
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return timestamp_offset;
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}
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/* Set the timestamp offset according to the packet duration. */
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static int align_rtp_timestamp_offset(struct mgcp_endpoint *endp,
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struct mgcp_rtp_state *state,
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struct mgcp_rtp_end *rtp_end,
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struct osmo_sockaddr *addr,
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uint32_t timestamp)
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{
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char ipbuf[INET6_ADDRSTRLEN];
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int ts_error = 0;
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int ts_check = 0;
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int ptime = state->packet_duration;
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/* Align according to: T + Toffs - Tlast = k * Tptime */
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ts_error = ts_alignment_error(&state->out_stream, ptime,
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timestamp + state->patch.timestamp_offset);
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/* If there is an alignment error, we have to compensate it */
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if (ts_error) {
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state->patch.timestamp_offset += ptime - ts_error;
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LOGPENDP(endp, DRTP, LOGL_NOTICE,
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"Corrected timestamp alignment error of %d on SSRC: %u "
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"new TS offset: %d, "
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"from %s:%d\n",
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ts_error, state->in_stream.ssrc,
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state->patch.timestamp_offset,
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osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
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osmo_sockaddr_port(&addr->u.sa));
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}
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/* Check we really managed to compensate the timestamp
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* offset. There should not be any remaining error, failing
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* here would point to a serous problem with the alignment
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* error computation function */
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ts_check = ts_alignment_error(&state->out_stream, ptime,
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timestamp + state->patch.timestamp_offset);
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OSMO_ASSERT(ts_check == 0);
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/* Return alignment error before compensation */
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return ts_error;
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}
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/*! dummy callback to disable transcoding (see also cfg->rtp_processing_cb).
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* \param[in] associated endpoint.
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* \param[in] destination RTP end.
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* \param[in,out] pointer to buffer with voice data.
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* \param[in] voice data length.
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* \param[in] maximum size of caller provided voice data buffer.
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* \returns ignores input parameters, return always 0. */
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int mgcp_rtp_processing_default(struct mgcp_endpoint *endp,
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struct mgcp_rtp_end *dst_end,
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char *data, int *len, int buf_size)
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{
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LOGPENDP(endp, DRTP, LOGL_DEBUG, "transcoding disabled\n");
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return 0;
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}
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|
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/*! dummy callback to disable transcoding (see also cfg->setup_rtp_processing_cb).
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* \param[in] associated endpoint.
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* \param[in] destination RTP connnection.
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* \param[in] source RTP connection.
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* \returns ignores input parameters, return always 0. */
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int mgcp_setup_rtp_processing_default(struct mgcp_endpoint *endp,
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struct mgcp_conn_rtp *conn_dst,
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|
struct mgcp_conn_rtp *conn_src)
|
|
{
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG, "transcoding disabled\n");
|
|
return 0;
|
|
}
|
|
|
|
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)
|
|
{
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG, "conn:%s using format defaults\n",
|
|
mgcp_conn_dump(conn->conn));
|
|
|
|
*codec = conn->end.codec;
|
|
*fmtp_extra = conn->end.fmtp_extra;
|
|
}
|
|
|
|
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)
|
|
{
|
|
int32_t d;
|
|
|
|
/* initialize or re-initialize */
|
|
if (!state->stats.initialized || state->stats.ssrc != ssrc) {
|
|
state->stats.initialized = 1;
|
|
state->stats.base_seq = seq;
|
|
state->stats.max_seq = seq - 1;
|
|
state->stats.ssrc = ssrc;
|
|
state->stats.jitter = 0;
|
|
state->stats.transit = transit;
|
|
state->stats.cycles = 0;
|
|
} else {
|
|
uint16_t udelta;
|
|
|
|
/* The below takes the shape of the validation of
|
|
* Appendix A. Check if there is something weird with
|
|
* the sequence number, otherwise check for a wrap
|
|
* around in the sequence number.
|
|
* It can't wrap during the initialization so let's
|
|
* skip it here. The Appendix A probably doesn't have
|
|
* this issue because of the probation. */
|
|
udelta = seq - state->stats.max_seq;
|
|
if (udelta < RTP_MAX_DROPOUT) {
|
|
if (seq < state->stats.max_seq)
|
|
state->stats.cycles += RTP_SEQ_MOD;
|
|
} else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
|
|
LOGPENDP(endp, DRTP, LOGL_NOTICE,
|
|
"RTP seqno made a very large jump on delta: %u\n",
|
|
udelta);
|
|
}
|
|
}
|
|
|
|
/* Calculate the jitter between the two packages. The TS should be
|
|
* taken closer to the read function. This was taken from the
|
|
* Appendix A of RFC 3550. Timestamp and arrival_time have a 1/rate
|
|
* resolution. */
|
|
d = transit - state->stats.transit;
|
|
state->stats.transit = transit;
|
|
if (d < 0)
|
|
d = -d;
|
|
state->stats.jitter += d - ((state->stats.jitter + 8) >> 4);
|
|
state->stats.max_seq = seq;
|
|
}
|
|
|
|
/* There may be different payload type numbers negotiated for two connections.
|
|
* Patch the payload type of an RTP packet so that it uses the payload type
|
|
* that is valid for the destination connection (conn_dst) */
|
|
static int mgcp_patch_pt(struct mgcp_conn_rtp *conn_src,
|
|
struct mgcp_conn_rtp *conn_dst, struct msgb *msg)
|
|
{
|
|
struct rtp_hdr *rtp_hdr;
|
|
uint8_t pt_in;
|
|
int pt_out;
|
|
|
|
if (msgb_length(msg) < sizeof(struct rtp_hdr)) {
|
|
LOG_CONN_RTP(conn_src, LOGL_ERROR, "RTP packet too short (%u < %zu)\n",
|
|
msgb_length(msg), sizeof(struct rtp_hdr));
|
|
return -EINVAL;
|
|
}
|
|
|
|
rtp_hdr = (struct rtp_hdr *)msgb_data(msg);
|
|
|
|
pt_in = rtp_hdr->payload_type;
|
|
pt_out = mgcp_codec_pt_translate(conn_src, conn_dst, pt_in);
|
|
if (pt_out < 0)
|
|
return -EINVAL;
|
|
|
|
rtp_hdr->payload_type = (uint8_t) pt_out;
|
|
return 0;
|
|
}
|
|
|
|
/* The RFC 3550 Appendix A assumes there are multiple sources but
|
|
* some of the supported endpoints (e.g. the nanoBTS) can only handle
|
|
* one source and this code will patch RTP header to appear as if there
|
|
* is only one source.
|
|
* There is also no probation period for new sources. Every RTP header
|
|
* we receive will be seen as a switch in streams. */
|
|
void mgcp_patch_and_count(struct mgcp_endpoint *endp,
|
|
struct mgcp_rtp_state *state,
|
|
struct mgcp_rtp_end *rtp_end,
|
|
struct osmo_sockaddr *addr, struct msgb *msg)
|
|
{
|
|
char ipbuf[INET6_ADDRSTRLEN];
|
|
uint32_t arrival_time;
|
|
int32_t transit;
|
|
uint16_t seq;
|
|
uint32_t timestamp, ssrc;
|
|
struct rtp_hdr *rtp_hdr;
|
|
int payload = rtp_end->codec->payload_type;
|
|
unsigned int len = msgb_length(msg);
|
|
|
|
if (len < sizeof(*rtp_hdr))
|
|
return;
|
|
|
|
rtp_hdr = (struct rtp_hdr *)msgb_data(msg);
|
|
seq = ntohs(rtp_hdr->sequence);
|
|
timestamp = ntohl(rtp_hdr->timestamp);
|
|
arrival_time = get_current_ts(rtp_end->codec->rate);
|
|
ssrc = ntohl(rtp_hdr->ssrc);
|
|
transit = arrival_time - timestamp;
|
|
|
|
mgcp_rtp_annex_count(endp, state, seq, transit, ssrc);
|
|
|
|
if (!state->initialized) {
|
|
state->initialized = 1;
|
|
state->in_stream.last_seq = seq - 1;
|
|
state->in_stream.ssrc = state->patch.orig_ssrc = ssrc;
|
|
state->in_stream.last_tsdelta = 0;
|
|
state->packet_duration =
|
|
mgcp_rtp_packet_duration(endp, rtp_end);
|
|
state->out_stream.last_seq = seq - 1;
|
|
state->out_stream.ssrc = state->patch.orig_ssrc = ssrc;
|
|
state->out_stream.last_tsdelta = 0;
|
|
state->out_stream.last_timestamp = timestamp;
|
|
state->out_stream.ssrc = ssrc - 1; /* force output SSRC change */
|
|
LOGPENDP(endp, DRTP, LOGL_INFO,
|
|
"initializing stream, SSRC: %u timestamp: %u "
|
|
"pkt-duration: %d, from %s:%d\n",
|
|
state->in_stream.ssrc,
|
|
state->patch.seq_offset, state->packet_duration,
|
|
osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
|
|
osmo_sockaddr_port(&addr->u.sa));
|
|
if (state->packet_duration == 0) {
|
|
state->packet_duration =
|
|
rtp_end->codec->rate * 20 / 1000;
|
|
LOGPENDP(endp, DRTP, LOGL_NOTICE,
|
|
"fixed packet duration is not available, "
|
|
"using fixed 20ms instead: %d from %s:%d\n",
|
|
state->packet_duration,
|
|
osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
|
|
osmo_sockaddr_port(&addr->u.sa));
|
|
}
|
|
} else if (state->in_stream.ssrc != ssrc) {
|
|
LOGPENDP(endp, DRTP, LOGL_NOTICE,
|
|
"SSRC changed: %u -> %u "
|
|
"from %s:%d\n",
|
|
state->in_stream.ssrc, rtp_hdr->ssrc,
|
|
osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
|
|
osmo_sockaddr_port(&addr->u.sa));
|
|
|
|
state->in_stream.ssrc = ssrc;
|
|
if (rtp_end->force_constant_ssrc) {
|
|
int16_t delta_seq;
|
|
|
|
/* Always increment seqno by 1 */
|
|
state->patch.seq_offset =
|
|
(state->out_stream.last_seq + 1) - seq;
|
|
|
|
/* Estimate number of packets that would have been sent */
|
|
delta_seq =
|
|
(arrival_time - state->in_stream.last_arrival_time
|
|
+ state->packet_duration / 2) /
|
|
state->packet_duration;
|
|
|
|
adjust_rtp_timestamp_offset(endp, state, rtp_end, addr,
|
|
delta_seq, timestamp);
|
|
|
|
state->patch.patch_ssrc = 1;
|
|
ssrc = state->patch.orig_ssrc;
|
|
if (rtp_end->force_constant_ssrc != -1)
|
|
rtp_end->force_constant_ssrc -= 1;
|
|
|
|
LOGPENDP(endp, DRTP, LOGL_NOTICE,
|
|
"SSRC patching enabled, SSRC: %u "
|
|
"SeqNo offset: %d, TS offset: %d "
|
|
"from %s:%d\n", state->in_stream.ssrc,
|
|
state->patch.seq_offset, state->patch.timestamp_offset,
|
|
osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
|
|
osmo_sockaddr_port(&addr->u.sa));
|
|
}
|
|
|
|
state->in_stream.last_tsdelta = 0;
|
|
} else {
|
|
/* Compute current per-packet timestamp delta */
|
|
check_rtp_timestamp(endp, state, &state->in_stream, rtp_end,
|
|
addr, seq, timestamp, "input",
|
|
&state->in_stream.last_tsdelta);
|
|
|
|
if (state->patch.patch_ssrc)
|
|
ssrc = state->patch.orig_ssrc;
|
|
}
|
|
|
|
/* Save before patching */
|
|
state->in_stream.last_timestamp = timestamp;
|
|
state->in_stream.last_seq = seq;
|
|
state->in_stream.last_arrival_time = arrival_time;
|
|
|
|
if (rtp_end->force_aligned_timing &&
|
|
state->out_stream.ssrc == ssrc && state->packet_duration)
|
|
/* Align the timestamp offset */
|
|
align_rtp_timestamp_offset(endp, state, rtp_end, addr,
|
|
timestamp);
|
|
|
|
/* Store the updated SSRC back to the packet */
|
|
if (state->patch.patch_ssrc)
|
|
rtp_hdr->ssrc = htonl(ssrc);
|
|
|
|
/* Apply the offset and store it back to the packet.
|
|
* This won't change anything if the offset is 0, so the conditional is
|
|
* omitted. */
|
|
seq += state->patch.seq_offset;
|
|
rtp_hdr->sequence = htons(seq);
|
|
timestamp += state->patch.timestamp_offset;
|
|
rtp_hdr->timestamp = htonl(timestamp);
|
|
|
|
/* Check again, whether the timestamps are still valid */
|
|
if (state->out_stream.ssrc == ssrc)
|
|
check_rtp_timestamp(endp, state, &state->out_stream, rtp_end,
|
|
addr, seq, timestamp, "output",
|
|
&state->out_stream.last_tsdelta);
|
|
|
|
/* Save output values */
|
|
state->out_stream.last_seq = seq;
|
|
state->out_stream.last_timestamp = timestamp;
|
|
state->out_stream.ssrc = ssrc;
|
|
|
|
if (payload < 0)
|
|
return;
|
|
|
|
#if 0
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG, "payload hdr payload %u -> endp payload %u\n",
|
|
rtp_hdr->payload_type, payload);
|
|
rtp_hdr->payload_type = payload;
|
|
#endif
|
|
}
|
|
|
|
/* There are different dialects used to format and transfer voice data. When
|
|
* the receiving end expects GSM-HR data to be formated after RFC 5993, this
|
|
* function is used to convert between RFC 5993 and TS 101318, which we normally
|
|
* use.
|
|
* Return 0 on sucess, negative on errors like invalid data length. */
|
|
static int rfc5993_hr_convert(struct mgcp_endpoint *endp, struct msgb *msg)
|
|
{
|
|
struct rtp_hdr *rtp_hdr;
|
|
if (msgb_length(msg) < sizeof(struct rtp_hdr)) {
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR, "AMR RTP packet too short (%d < %zu)\n",
|
|
msgb_length(msg), sizeof(struct rtp_hdr));
|
|
return -EINVAL;
|
|
}
|
|
|
|
rtp_hdr = (struct rtp_hdr *)msgb_data(msg);
|
|
|
|
if (msgb_length(msg) == GSM_HR_BYTES + sizeof(struct rtp_hdr)) {
|
|
/* TS 101318 encoding => RFC 5993 encoding */
|
|
msgb_put(msg, 1);
|
|
memmove(rtp_hdr->data + 1, rtp_hdr->data, GSM_HR_BYTES);
|
|
rtp_hdr->data[0] = 0x00;
|
|
} else if (msgb_length(msg) == GSM_HR_BYTES + sizeof(struct rtp_hdr) + 1) {
|
|
/* RFC 5993 encoding => TS 101318 encoding */
|
|
memmove(rtp_hdr->data, rtp_hdr->data + 1, GSM_HR_BYTES);
|
|
msgb_trim(msg, msgb_length(msg) - 1);
|
|
} else {
|
|
/* It is possible that multiple payloads occur in one RTP
|
|
* packet. This is not supported yet. */
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR,
|
|
"cannot figure out how to convert RTP packet\n");
|
|
return -ENOTSUP;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
/* For AMR RTP two framing modes are defined RFC3267. There is a bandwith
|
|
* efficient encoding scheme where all fields are packed together one after
|
|
* another and an octet aligned mode where all fields are aligned to octet
|
|
* boundaries. This function is used to convert between the two modes */
|
|
static int amr_oa_bwe_convert(struct mgcp_endpoint *endp, struct msgb *msg,
|
|
bool target_is_oa)
|
|
{
|
|
/* NOTE: the msgb has an allocated length of RTP_BUF_SIZE, so there is
|
|
* plenty of space available to store the slightly larger, converted
|
|
* data */
|
|
struct rtp_hdr *rtp_hdr;
|
|
unsigned int payload_len;
|
|
int rc;
|
|
|
|
if (msgb_length(msg) < sizeof(struct rtp_hdr)) {
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR, "AMR RTP packet too short (%d < %zu)\n", msgb_length(msg), sizeof(struct rtp_hdr));
|
|
return -EINVAL;
|
|
}
|
|
|
|
rtp_hdr = (struct rtp_hdr *)msgb_data(msg);
|
|
|
|
payload_len = msgb_length(msg) - sizeof(struct rtp_hdr);
|
|
|
|
if (osmo_amr_is_oa(rtp_hdr->data, payload_len)) {
|
|
if (!target_is_oa)
|
|
/* Input data is oa an target format is bwe
|
|
* ==> convert */
|
|
rc = osmo_amr_oa_to_bwe(rtp_hdr->data, payload_len);
|
|
else
|
|
/* Input data is already bew, but we accept it anyway
|
|
* ==> no conversion needed */
|
|
rc = payload_len;
|
|
} else {
|
|
if (target_is_oa)
|
|
/* Input data is bwe an target format is oa
|
|
* ==> convert */
|
|
rc = osmo_amr_bwe_to_oa(rtp_hdr->data, payload_len,
|
|
RTP_BUF_SIZE);
|
|
else
|
|
/* Input data is already oa, but we accept it anyway
|
|
* ==> no conversion needed */
|
|
rc = payload_len;
|
|
}
|
|
if (rc < 0) {
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR,
|
|
"AMR RTP packet conversion failed\n");
|
|
return -EINVAL;
|
|
}
|
|
|
|
return msgb_trim(msg, rc + sizeof(struct rtp_hdr));
|
|
}
|
|
|
|
/* Check if a conversion between octet-aligned and bandwith-efficient mode is
|
|
* indicated. */
|
|
static bool amr_oa_bwe_convert_indicated(struct mgcp_rtp_codec *codec)
|
|
{
|
|
if (codec->param_present == false)
|
|
return false;
|
|
if (!codec->param.amr_octet_aligned_present)
|
|
return false;
|
|
if (strcmp(codec->subtype_name, "AMR") != 0)
|
|
return false;
|
|
return true;
|
|
}
|
|
|
|
|
|
/* Return whether an RTP packet with AMR payload is in octet-aligned mode.
|
|
* Return 0 if in bandwidth-efficient mode, 1 for octet-aligned mode, and negative if the RTP data is invalid. */
|
|
static int amr_oa_check(char *data, int len)
|
|
{
|
|
struct rtp_hdr *rtp_hdr;
|
|
unsigned int payload_len;
|
|
|
|
if (len < sizeof(struct rtp_hdr))
|
|
return -EINVAL;
|
|
|
|
rtp_hdr = (struct rtp_hdr *)data;
|
|
|
|
payload_len = len - sizeof(struct rtp_hdr);
|
|
if (payload_len < sizeof(struct amr_hdr))
|
|
return -EINVAL;
|
|
|
|
return osmo_amr_is_oa(rtp_hdr->data, payload_len) ? 1 : 0;
|
|
}
|
|
|
|
/* Forward data to a debug tap. This is debug function that is intended for
|
|
* debugging the voice traffic with tools like gstreamer */
|
|
static void forward_data(int fd, struct mgcp_rtp_tap *tap, struct msgb *msg)
|
|
{
|
|
int rc;
|
|
|
|
if (!tap->enabled)
|
|
return;
|
|
|
|
rc = sendto(fd, msgb_data(msg), msgb_length(msg), 0, (struct sockaddr *)&tap->forward,
|
|
sizeof(tap->forward));
|
|
|
|
if (rc < 0)
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"Forwarding tapped (debug) voice data failed.\n");
|
|
}
|
|
|
|
/* Generate an RTP header if it is missing */
|
|
static void gen_rtp_header(struct msgb *msg, struct mgcp_rtp_end *rtp_end,
|
|
struct mgcp_rtp_state *state)
|
|
{
|
|
struct rtp_hdr *hdr = (struct rtp_hdr *)msgb_data(msg);
|
|
|
|
if (hdr->version > 0)
|
|
return;
|
|
|
|
hdr->version = 2;
|
|
hdr->payload_type = rtp_end->codec->payload_type;
|
|
hdr->timestamp = osmo_htonl(get_current_ts(rtp_end->codec->rate));
|
|
hdr->sequence = osmo_htons(state->alt_rtp_tx_sequence);
|
|
hdr->ssrc = state->alt_rtp_tx_ssrc;
|
|
}
|
|
|
|
|
|
/*! Send RTP/RTCP data to a specified destination connection.
|
|
* \param[in] endp associated endpoint (for configuration, logging).
|
|
* \param[in] is_rtp flag to specify if the packet is of type RTP or RTCP.
|
|
* \param[in] spoofed source address (set to NULL to disable).
|
|
* \param[in] buf buffer that contains the RTP/RTCP data.
|
|
* \param[in] len length of the buffer that contains the RTP/RTCP data.
|
|
* \param[in] conn_src associated source connection.
|
|
* \param[in] conn_dst associated destination connection.
|
|
* \returns 0 on success, -1 on ERROR. */
|
|
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)
|
|
{
|
|
/*! When no destination connection is available (e.g. when only one
|
|
* connection in loopback mode exists), then the source connection
|
|
* shall be specified as destination connection */
|
|
|
|
struct mgcp_trunk *trunk = endp->trunk;
|
|
struct mgcp_rtp_end *rtp_end;
|
|
struct mgcp_rtp_state *rtp_state;
|
|
char ipbuf[INET6_ADDRSTRLEN];
|
|
char *dest_name;
|
|
int rc;
|
|
int len;
|
|
|
|
OSMO_ASSERT(conn_src);
|
|
OSMO_ASSERT(conn_dst);
|
|
|
|
if (is_rtp) {
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG, "delivering RTP packet...\n");
|
|
} else {
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG, "delivering RTCP packet...\n");
|
|
}
|
|
|
|
/* FIXME: It is legal that the payload type on the egress connection is
|
|
* different from the payload type that has been negotiated on the
|
|
* ingress connection. Essentially the codecs are the same so we can
|
|
* match them and patch the payload type. However, if we can not find
|
|
* the codec pendant (everything ist equal except the PT), we are of
|
|
* course unable to patch the payload type. A situation like this
|
|
* should not occur if transcoding is consequently avoided. Until
|
|
* we have transcoding support in osmo-mgw we can not resolve this. */
|
|
if (is_rtp) {
|
|
rc = mgcp_patch_pt(conn_src, conn_dst, msg);
|
|
if (rc < 0) {
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG,
|
|
"can not patch PT because no suitable egress codec was found.\n");
|
|
}
|
|
}
|
|
|
|
/* Note: In case of loopback configuration, both, the source and the
|
|
* destination will point to the same connection. */
|
|
rtp_end = &conn_dst->end;
|
|
rtp_state = &conn_src->state;
|
|
dest_name = conn_dst->conn->name;
|
|
|
|
/* Ensure we have an alternative SSRC in case we need it, see also
|
|
* gen_rtp_header() */
|
|
if (rtp_state->alt_rtp_tx_ssrc == 0)
|
|
rtp_state->alt_rtp_tx_ssrc = rand();
|
|
|
|
if (!rtp_end->output_enabled) {
|
|
rtpconn_rate_ctr_inc(conn_dst, endp, RTP_DROPPED_PACKETS_CTR);
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG,
|
|
"output disabled, drop to %s %s "
|
|
"rtp_port:%u rtcp_port:%u\n",
|
|
dest_name,
|
|
osmo_sockaddr_ntop(&rtp_end->addr.u.sa, ipbuf),
|
|
ntohs(rtp_end->rtp_port), ntohs(rtp_end->rtcp_port)
|
|
);
|
|
} else if (is_rtp) {
|
|
int cont;
|
|
int nbytes = 0;
|
|
int buflen = msgb_length(msg);
|
|
|
|
/* Make sure we have a valid RTP header, in cases where no RTP
|
|
* header is present, we will generate one. */
|
|
gen_rtp_header(msg, rtp_end, rtp_state);
|
|
|
|
do {
|
|
/* Run transcoder */
|
|
cont = endp->cfg->rtp_processing_cb(endp, rtp_end,
|
|
(char*)msgb_data(msg), &buflen,
|
|
RTP_BUF_SIZE);
|
|
if (cont < 0)
|
|
break;
|
|
|
|
if (addr)
|
|
mgcp_patch_and_count(endp, rtp_state, rtp_end,
|
|
addr, msg);
|
|
|
|
if (amr_oa_bwe_convert_indicated(conn_dst->end.codec)) {
|
|
rc = amr_oa_bwe_convert(endp, msg,
|
|
conn_dst->end.codec->param.amr_octet_aligned);
|
|
if (rc < 0) {
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR,
|
|
"Error in AMR octet-aligned <-> bandwidth-efficient mode conversion\n");
|
|
break;
|
|
}
|
|
}
|
|
else if (rtp_end->rfc5993_hr_convert
|
|
&& strcmp(conn_src->end.codec->subtype_name,
|
|
"GSM-HR-08") == 0) {
|
|
rc = rfc5993_hr_convert(endp, msg);
|
|
if (rc < 0) {
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR, "Error while converting to GSM-HR-08\n");
|
|
break;
|
|
}
|
|
}
|
|
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG,
|
|
"process/send to %s %s "
|
|
"rtp_port:%u rtcp_port:%u\n",
|
|
dest_name,
|
|
osmo_sockaddr_ntop(&rtp_end->addr.u.sa, ipbuf),
|
|
ntohs(rtp_end->rtp_port), ntohs(rtp_end->rtcp_port)
|
|
);
|
|
|
|
/* Forward a copy of the RTP data to a debug ip/port */
|
|
forward_data(rtp_end->rtp.fd, &conn_src->tap_out,
|
|
msg);
|
|
|
|
/* FIXME: HACK HACK HACK. See OS#2459.
|
|
* The ip.access nano3G needs the first RTP payload's first two bytes to read hex
|
|
* 'e400', or it will reject the RAB assignment. It seems to not harm other femto
|
|
* cells (as long as we patch only the first RTP payload in each stream).
|
|
*/
|
|
if (!rtp_state->patched_first_rtp_payload
|
|
&& conn_src->conn->mode == MGCP_CONN_LOOPBACK) {
|
|
uint8_t *data = msgb_data(msg) + 12;
|
|
if (data[0] == 0xe0) {
|
|
data[0] = 0xe4;
|
|
data[1] = 0x00;
|
|
rtp_state->patched_first_rtp_payload = true;
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG,
|
|
"Patching over first two bytes"
|
|
" to fake an IuUP Initialization Ack\n");
|
|
}
|
|
}
|
|
|
|
len = mgcp_udp_send(rtp_end->rtp.fd, &rtp_end->addr, rtp_end->rtp_port,
|
|
(char*)msgb_data(msg), msgb_length(msg));
|
|
|
|
if (len <= 0)
|
|
return len;
|
|
|
|
rtpconn_rate_ctr_inc(conn_dst, endp, RTP_PACKETS_TX_CTR);
|
|
rtpconn_rate_ctr_add(conn_dst, endp, RTP_OCTETS_TX_CTR, len);
|
|
rtp_state->alt_rtp_tx_sequence++;
|
|
|
|
nbytes += len;
|
|
buflen = cont;
|
|
} while (buflen > 0);
|
|
return nbytes;
|
|
} else if (!trunk->omit_rtcp) {
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG,
|
|
"send to %s %s rtp_port:%u rtcp_port:%u\n",
|
|
dest_name, osmo_sockaddr_ntop(&rtp_end->addr.u.sa, ipbuf),
|
|
ntohs(rtp_end->rtp_port), ntohs(rtp_end->rtcp_port)
|
|
);
|
|
|
|
len = mgcp_udp_send(rtp_end->rtcp.fd,
|
|
&rtp_end->addr,
|
|
rtp_end->rtcp_port, (char*)msgb_data(msg), msgb_length(msg));
|
|
|
|
rtpconn_rate_ctr_inc(conn_dst, endp, RTP_PACKETS_TX_CTR);
|
|
rtpconn_rate_ctr_add(conn_dst, endp, RTP_OCTETS_TX_CTR, len);
|
|
rtp_state->alt_rtp_tx_sequence++;
|
|
|
|
return len;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* Check if the origin (addr) matches the address/port data of the RTP
|
|
* connections. */
|
|
static int check_rtp_origin(struct mgcp_conn_rtp *conn, struct osmo_sockaddr *addr)
|
|
{
|
|
char ipbuf[INET6_ADDRSTRLEN];
|
|
|
|
if (addr_is_any(&conn->end.addr)) {
|
|
switch (conn->conn->mode) {
|
|
case MGCP_CONN_LOOPBACK:
|
|
/* HACK: for IuUP, we want to reply with an IuUP Initialization ACK upon the first RTP
|
|
* message received. We currently hackishly accomplish that by putting the endpoint in
|
|
* loopback mode and patching over the looped back RTP message to make it look like an
|
|
* ack. We don't know the femto cell's IP address and port until the RAB Assignment
|
|
* Response is received, but the nano3G expects an IuUP Initialization Ack before it even
|
|
* sends the RAB Assignment Response. Hence, if the remote address is 0.0.0.0 and the
|
|
* MGCP port is in loopback mode, allow looping back the packet to any source. */
|
|
LOGPCONN(conn->conn, DRTP, LOGL_ERROR,
|
|
"In loopback mode and remote address not set:"
|
|
" allowing data from address: %s\n",
|
|
osmo_sockaddr_ntop(&addr->u.sa, ipbuf));
|
|
return 0;
|
|
|
|
default:
|
|
/* Receiving early media before the endpoint is configured. Instead of logging
|
|
* this as an error that occurs on every call, keep it more low profile to not
|
|
* confuse humans with expected errors. */
|
|
LOGPCONN(conn->conn, DRTP, LOGL_INFO,
|
|
"Rx RTP from %s, but remote address not set:"
|
|
" dropping early media\n",
|
|
osmo_sockaddr_ntop(&addr->u.sa, ipbuf));
|
|
return -1;
|
|
}
|
|
}
|
|
|
|
/* Note: Check if the inbound RTP data comes from the same host to
|
|
* which we send our outgoing RTP traffic. */
|
|
if (conn->end.addr.u.sa.sa_family != addr->u.sa.sa_family ||
|
|
(conn->end.addr.u.sa.sa_family == AF_INET &&
|
|
conn->end.addr.u.sin.sin_addr.s_addr != addr->u.sin.sin_addr.s_addr) ||
|
|
(conn->end.addr.u.sa.sa_family == AF_INET6 &&
|
|
memcmp(&conn->end.addr.u.sin6.sin6_addr, &addr->u.sin6.sin6_addr,
|
|
sizeof(struct in6_addr)))) {
|
|
LOGPCONN(conn->conn, DRTP, LOGL_ERROR,
|
|
"data from wrong address: %s, ",
|
|
osmo_sockaddr_ntop(&addr->u.sa, ipbuf));
|
|
LOGPC(DRTP, LOGL_ERROR, "expected: %s\n",
|
|
osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf));
|
|
LOGPCONN(conn->conn, DRTP, LOGL_ERROR, "packet tossed\n");
|
|
return -1;
|
|
}
|
|
|
|
/* Note: Usually the remote remote port of the data we receive will be
|
|
* the same as the remote port where we transmit outgoing RTP traffic
|
|
* to (set by MDCX). We use this to check the origin of the data for
|
|
* plausibility. */
|
|
if (ntohs(conn->end.rtp_port) != osmo_sockaddr_port(&addr->u.sa) &&
|
|
ntohs(conn->end.rtcp_port) != osmo_sockaddr_port(&addr->u.sa)) {
|
|
LOGPCONN(conn->conn, DRTP, LOGL_ERROR,
|
|
"data from wrong source port: %d, ",
|
|
osmo_sockaddr_port(&addr->u.sa));
|
|
LOGPC(DRTP, LOGL_ERROR,
|
|
"expected: %d for RTP or %d for RTCP\n",
|
|
ntohs(conn->end.rtp_port), ntohs(conn->end.rtcp_port));
|
|
LOGPCONN(conn->conn, DRTP, LOGL_ERROR, "packet tossed\n");
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* Check the if the destination address configuration of an RTP connection
|
|
* makes sense */
|
|
static int check_rtp_destin(struct mgcp_conn_rtp *conn)
|
|
{
|
|
char ipbuf[INET6_ADDRSTRLEN];
|
|
bool ip_is_any = addr_is_any(&conn->end.addr);
|
|
|
|
/* Note: it is legal to create a connection but never setting a port
|
|
* and IP-address for outgoing data. */
|
|
if (ip_is_any && conn->end.rtp_port == 0) {
|
|
LOGPCONN(conn->conn, DRTP, LOGL_DEBUG,
|
|
"destination IP-address and rtp port is (not yet) known (%s:%u)\n",
|
|
osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf), conn->end.rtp_port);
|
|
return -1;
|
|
}
|
|
|
|
if (ip_is_any) {
|
|
LOGPCONN(conn->conn, DRTP, LOGL_ERROR,
|
|
"destination IP-address is invalid (%s:%u)\n",
|
|
osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf), conn->end.rtp_port);
|
|
return -1;
|
|
}
|
|
|
|
if (conn->end.rtp_port == 0) {
|
|
LOGPCONN(conn->conn, DRTP, LOGL_ERROR,
|
|
"destination rtp port is invalid (%s:%u)\n",
|
|
osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf), conn->end.rtp_port);
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* Do some basic checks to make sure that the RTCP packets we are going to
|
|
* process are not complete garbage */
|
|
static int check_rtcp(struct mgcp_conn_rtp *conn_src, struct msgb *msg)
|
|
{
|
|
struct rtcp_hdr *hdr;
|
|
unsigned int len;
|
|
uint8_t type;
|
|
|
|
/* RTPC packets that are just a header without data do not make
|
|
* any sense. */
|
|
if (msgb_length(msg) < sizeof(struct rtcp_hdr)) {
|
|
LOG_CONN_RTP(conn_src, LOGL_ERROR, "RTCP packet too short (%u < %zu)\n",
|
|
msgb_length(msg), sizeof(struct rtcp_hdr));
|
|
return -EINVAL;
|
|
}
|
|
|
|
/* Make sure that the length of the received packet does not exceed
|
|
* the available buffer size */
|
|
hdr = (struct rtcp_hdr *)msgb_data(msg);
|
|
len = (osmo_ntohs(hdr->length) + 1) * 4;
|
|
if (len > msgb_length(msg)) {
|
|
LOG_CONN_RTP(conn_src, LOGL_ERROR, "RTCP header length exceeds packet size (%u > %u)\n",
|
|
len, msgb_length(msg));
|
|
return -EINVAL;
|
|
}
|
|
|
|
/* Make sure we accept only packets that have a proper packet type set
|
|
* See also: http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml */
|
|
type = hdr->type;
|
|
if ((type < 192 || type > 195) && (type < 200 || type > 213)) {
|
|
LOG_CONN_RTP(conn_src, LOGL_ERROR, "RTCP header: invalid type: %u\n", type);
|
|
return -EINVAL;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* Do some basic checks to make sure that the RTP packets we are going to
|
|
* process are not complete garbage */
|
|
static int check_rtp(struct mgcp_conn_rtp *conn_src, struct msgb *msg)
|
|
{
|
|
size_t min_size = sizeof(struct rtp_hdr);
|
|
if (msgb_length(msg) < min_size) {
|
|
LOG_CONN_RTP(conn_src, LOGL_ERROR, "RTP packet too short (%u < %zu)\n",
|
|
msgb_length(msg), min_size);
|
|
return -1;
|
|
}
|
|
|
|
/* FIXME: Add more checks, the reason why we do not check more than
|
|
* the length is because we currently handle IUUP packets as RTP
|
|
* packets, so they must pass this check, if we weould be more
|
|
* strict here, we would possibly break 3G. (see also FIXME note
|
|
* below */
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* Send RTP data. Possible options are standard RTP packet
|
|
* transmission or trsmission via an osmux connection */
|
|
static int mgcp_send_rtp(struct mgcp_conn_rtp *conn_dst, struct msgb *msg)
|
|
{
|
|
struct osmo_rtp_msg_ctx *mc = OSMO_RTP_MSG_CTX(msg);
|
|
enum rtp_proto proto = mc->proto;
|
|
struct mgcp_conn_rtp *conn_src = mc->conn_src;
|
|
struct mgcp_endpoint *endp = conn_src->conn->endp;
|
|
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG, "destin conn:%s\n",
|
|
mgcp_conn_dump(conn_dst->conn));
|
|
|
|
/* Before we try to deliver the packet, we check if the destination
|
|
* port and IP-Address make sense at all. If not, we will be unable
|
|
* to deliver the packet. */
|
|
if (check_rtp_destin(conn_dst) != 0)
|
|
return -1;
|
|
|
|
/* Depending on the RTP connection type, deliver the RTP packet to the
|
|
* destination connection. */
|
|
switch (conn_dst->type) {
|
|
case MGCP_RTP_DEFAULT:
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG,
|
|
"endpoint type is MGCP_RTP_DEFAULT, "
|
|
"using mgcp_send() to forward data directly\n");
|
|
return mgcp_send(endp, proto == MGCP_PROTO_RTP,
|
|
mc->from_addr, msg, conn_src, conn_dst);
|
|
case MGCP_OSMUX_BSC_NAT:
|
|
case MGCP_OSMUX_BSC:
|
|
LOGPENDP(endp, DRTP, LOGL_DEBUG,
|
|
"endpoint type is MGCP_OSMUX_BSC_NAT, "
|
|
"using osmux_xfrm_to_osmux() to forward data through OSMUX\n");
|
|
return osmux_xfrm_to_osmux((char*)msgb_data(msg), msgb_length(msg), conn_dst);
|
|
}
|
|
|
|
/* If the data has not been handled/forwarded until here, it will
|
|
* be discarded, this should not happen, normally the MGCP type
|
|
* should be properly set */
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR, "bad MGCP type -- data discarded!\n");
|
|
|
|
return -1;
|
|
}
|
|
|
|
/*! dispatch incoming RTP packet to opposite RTP connection.
|
|
* \param[in] proto protocol (MGCP_CONN_TYPE_RTP or MGCP_CONN_TYPE_RTCP).
|
|
* \param[in] addr socket address where the RTP packet has been received from.
|
|
* \param[in] buf buffer that hold the RTP payload.
|
|
* \param[in] buf_size size data length of buf.
|
|
* \param[in] conn originating connection.
|
|
* \returns 0 on success, -1 on ERROR. */
|
|
int mgcp_dispatch_rtp_bridge_cb(struct msgb *msg)
|
|
{
|
|
struct osmo_rtp_msg_ctx *mc = OSMO_RTP_MSG_CTX(msg);
|
|
struct mgcp_conn_rtp *conn_src = mc->conn_src;
|
|
struct mgcp_conn *conn = conn_src->conn;
|
|
struct mgcp_conn *conn_dst;
|
|
struct osmo_sockaddr *from_addr = mc->from_addr;
|
|
|
|
/*! NOTE: This callback function implements the endpoint specific
|
|
* dispatch behaviour of an rtp bridge/proxy endpoint. It is assumed
|
|
* that the endpoint will hold only two connections. This premise
|
|
* is used to determine the opposite connection (it is always the
|
|
* connection that is not the originating connection). Once the
|
|
* destination connection is known the RTP packet is sent via
|
|
* the destination connection. */
|
|
|
|
|
|
/* Check if the connection is in loopback mode, if yes, just send the
|
|
* incoming data back to the origin */
|
|
if (conn->mode == MGCP_CONN_LOOPBACK) {
|
|
/* When we are in loopback mode, we loop back all incoming
|
|
* packets back to their origin. We will use the originating
|
|
* address data from the UDP packet header to patch the
|
|
* outgoing address in connection on the fly */
|
|
if (conn->u.rtp.end.rtp_port == 0) {
|
|
OSMO_ASSERT(conn->u.rtp.end.addr.u.sa.sa_family == from_addr->u.sa.sa_family);
|
|
switch (from_addr->u.sa.sa_family) {
|
|
case AF_INET:
|
|
conn->u.rtp.end.addr.u.sin.sin_addr = from_addr->u.sin.sin_addr;
|
|
conn->u.rtp.end.rtp_port = from_addr->u.sin.sin_port;
|
|
break;
|
|
case AF_INET6:
|
|
conn->u.rtp.end.addr.u.sin6.sin6_addr = from_addr->u.sin6.sin6_addr;
|
|
conn->u.rtp.end.rtp_port = from_addr->u.sin6.sin6_port;
|
|
break;
|
|
default:
|
|
OSMO_ASSERT(false);
|
|
}
|
|
}
|
|
return mgcp_send_rtp(conn_src, msg);
|
|
}
|
|
|
|
/* Find a destination connection. */
|
|
/* NOTE: This code path runs every time an RTP packet is received. The
|
|
* function mgcp_find_dst_conn() we use to determine the detination
|
|
* connection will iterate the connection list inside the endpoint.
|
|
* Since list iterations are quite costly, we will figure out the
|
|
* destination only once and use the optional private data pointer of
|
|
* the connection to cache the destination connection pointer. */
|
|
if (!conn->priv) {
|
|
conn_dst = mgcp_find_dst_conn(conn);
|
|
conn->priv = conn_dst;
|
|
} else {
|
|
conn_dst = (struct mgcp_conn *)conn->priv;
|
|
}
|
|
|
|
/* There is no destination conn, stop here */
|
|
if (!conn_dst) {
|
|
LOGPCONN(conn, DRTP, LOGL_DEBUG,
|
|
"no connection to forward an incoming RTP packet to\n");
|
|
return -1;
|
|
}
|
|
|
|
/* The destination conn is not an RTP connection */
|
|
if (conn_dst->type != MGCP_CONN_TYPE_RTP) {
|
|
LOGPCONN(conn, DRTP, LOGL_ERROR,
|
|
"unable to find suitable destination conn\n");
|
|
return -1;
|
|
}
|
|
|
|
/* Dispatch RTP packet to destination RTP connection */
|
|
return mgcp_send_rtp(&conn_dst->u.rtp, msg);
|
|
}
|
|
|
|
/*! dispatch incoming RTP packet to E1 subslot, handle RTCP packets locally.
|
|
* \param[in] proto protocol (MGCP_CONN_TYPE_RTP or MGCP_CONN_TYPE_RTCP).
|
|
* \param[in] addr socket address where the RTP packet has been received from.
|
|
* \param[in] buf buffer that hold the RTP payload.
|
|
* \param[in] buf_size size data length of buf.
|
|
* \param[in] conn originating connection.
|
|
* \returns 0 on success, -1 on ERROR. */
|
|
int mgcp_dispatch_e1_bridge_cb(struct msgb *msg)
|
|
{
|
|
struct osmo_rtp_msg_ctx *mc = OSMO_RTP_MSG_CTX(msg);
|
|
struct mgcp_conn_rtp *conn_src = mc->conn_src;
|
|
struct mgcp_conn *conn = conn_src->conn;
|
|
struct osmo_sockaddr *from_addr = mc->from_addr;
|
|
|
|
/* Check if the connection is in loopback mode, if yes, just send the
|
|
* incoming data back to the origin */
|
|
if (conn->mode == MGCP_CONN_LOOPBACK) {
|
|
/* When we are in loopback mode, we loop back all incoming
|
|
* packets back to their origin. We will use the originating
|
|
* address data from the UDP packet header to patch the
|
|
* outgoing address in connection on the fly */
|
|
if (conn->u.rtp.end.rtp_port == 0) {
|
|
OSMO_ASSERT(conn->u.rtp.end.addr.u.sa.sa_family == from_addr->u.sa.sa_family);
|
|
switch (from_addr->u.sa.sa_family) {
|
|
case AF_INET:
|
|
conn->u.rtp.end.addr.u.sin.sin_addr = from_addr->u.sin.sin_addr;
|
|
conn->u.rtp.end.rtp_port = from_addr->u.sin.sin_port;
|
|
break;
|
|
case AF_INET6:
|
|
conn->u.rtp.end.addr.u.sin6.sin6_addr = from_addr->u.sin6.sin6_addr;
|
|
conn->u.rtp.end.rtp_port = from_addr->u.sin6.sin6_port;
|
|
break;
|
|
default:
|
|
OSMO_ASSERT(false);
|
|
}
|
|
}
|
|
return mgcp_send_rtp(conn_src, msg);
|
|
}
|
|
|
|
/* Forward to E1 */
|
|
return mgcp_e1_send_rtp(conn->endp, conn->u.rtp.end.codec, msg);
|
|
}
|
|
|
|
/*! cleanup an endpoint when a connection on an RTP bridge endpoint is removed.
|
|
* \param[in] endp Endpoint on which the connection resides.
|
|
* \param[in] conn Connection that is about to be removed (ignored). */
|
|
void mgcp_cleanup_rtp_bridge_cb(struct mgcp_endpoint *endp, struct mgcp_conn *conn)
|
|
{
|
|
struct mgcp_conn *conn_cleanup;
|
|
|
|
/* In mgcp_dispatch_rtp_bridge_cb() we use conn->priv to cache the
|
|
* pointer to the destination connection, so that we do not have
|
|
* to go through the list every time an RTP packet arrives. To prevent
|
|
* a use-after-free situation we invalidate this information for all
|
|
* connections present when one connection is removed from the
|
|
* endpoint. */
|
|
llist_for_each_entry(conn_cleanup, &endp->conns, entry) {
|
|
if (conn_cleanup->priv == conn)
|
|
conn_cleanup->priv = NULL;
|
|
}
|
|
}
|
|
|
|
/*! cleanup an endpoint when a connection on an E1 endpoint is removed.
|
|
* \param[in] endp Endpoint on which the connection resides.
|
|
* \param[in] conn Connection that is about to be removed (ignored). */
|
|
void mgcp_cleanup_e1_bridge_cb(struct mgcp_endpoint *endp, struct mgcp_conn *conn)
|
|
{
|
|
/* Cleanup tasks for E1 are the same as for regular endpoint. The
|
|
* shut down of the E1 part is handled separately. */
|
|
mgcp_cleanup_rtp_bridge_cb(endp, conn);
|
|
}
|
|
|
|
static bool is_dummy_msg(enum rtp_proto proto, struct msgb *msg)
|
|
{
|
|
return msgb_length(msg) == 1 && msgb_data(msg)[0] == MGCP_DUMMY_LOAD;
|
|
}
|
|
|
|
/* Handle incoming RTP data from NET */
|
|
static int rtp_data_net(struct osmo_fd *fd, unsigned int what)
|
|
{
|
|
/* NOTE: This is a generic implementation. RTP data is received. In
|
|
* case of loopback the data is just sent back to its origin. All
|
|
* other cases implement endpoint specific behaviour (e.g. how is the
|
|
* destination connection determined?). That specific behaviour is
|
|
* implemented by the callback function that is called at the end of
|
|
* the function */
|
|
|
|
struct mgcp_conn_rtp *conn_src;
|
|
struct mgcp_endpoint *endp;
|
|
struct osmo_sockaddr addr;
|
|
socklen_t slen = sizeof(addr);
|
|
char ipbuf[INET6_ADDRSTRLEN];
|
|
int ret;
|
|
enum rtp_proto proto;
|
|
struct osmo_rtp_msg_ctx *mc;
|
|
struct msgb *msg = msgb_alloc(RTP_BUF_SIZE, "RTP-rx");
|
|
int rc;
|
|
|
|
conn_src = (struct mgcp_conn_rtp *)fd->data;
|
|
OSMO_ASSERT(conn_src);
|
|
endp = conn_src->conn->endp;
|
|
OSMO_ASSERT(endp);
|
|
|
|
proto = (fd == &conn_src->end.rtp)? MGCP_PROTO_RTP : MGCP_PROTO_RTCP;
|
|
|
|
ret = recvfrom(fd->fd, msgb_data(msg), msg->data_len, 0, (struct sockaddr *)&addr.u.sa, &slen);
|
|
|
|
if (ret <= 0) {
|
|
LOG_CONN_RTP(conn_src, LOGL_ERROR, "recvfrom error: %s\n", strerror(errno));
|
|
rc = -1;
|
|
goto out;
|
|
}
|
|
|
|
msgb_put(msg, ret);
|
|
|
|
LOG_CONN_RTP(conn_src, LOGL_DEBUG, "%s: rx %u bytes from %s:%u\n",
|
|
proto == MGCP_PROTO_RTP ? "RTP" : "RTPC",
|
|
msgb_length(msg), osmo_sockaddr_ntop(&addr.u.sa, ipbuf),
|
|
osmo_sockaddr_port(&addr.u.sa));
|
|
|
|
if ((proto == MGCP_PROTO_RTP && check_rtp(conn_src, msg))
|
|
|| (proto == MGCP_PROTO_RTCP && check_rtcp(conn_src, msg))) {
|
|
/* Logging happened in the two check_ functions */
|
|
rc = -1;
|
|
goto out;
|
|
}
|
|
|
|
if (is_dummy_msg(proto, msg)) {
|
|
LOG_CONN_RTP(conn_src, LOGL_DEBUG, "rx dummy packet (dropped)\n");
|
|
rc = 0;
|
|
goto out;
|
|
}
|
|
|
|
/* Since the msgb remains owned and freed by this function, the msg ctx data struct can just be on the stack and
|
|
* needs not be allocated with the msgb. */
|
|
mc = OSMO_RTP_MSG_CTX(msg);
|
|
*mc = (struct osmo_rtp_msg_ctx){
|
|
.proto = proto,
|
|
.conn_src = conn_src,
|
|
.from_addr = &addr,
|
|
};
|
|
LOG_CONN_RTP(conn_src, LOGL_DEBUG, "msg ctx: %d %p %s\n",
|
|
mc->proto, mc->conn_src,
|
|
osmo_hexdump((void*)mc->from_addr,
|
|
mc->from_addr->u.sa.sa_family == AF_INET6 ?
|
|
sizeof(struct sockaddr_in6) :
|
|
sizeof(struct sockaddr_in)));
|
|
|
|
/* Increment RX statistics */
|
|
rate_ctr_inc(&conn_src->rate_ctr_group->ctr[RTP_PACKETS_RX_CTR]);
|
|
rate_ctr_add(&conn_src->rate_ctr_group->ctr[RTP_OCTETS_RX_CTR], msgb_length(msg));
|
|
/* FIXME: count RTP and RTCP separately, also count IuUP payload-less separately */
|
|
|
|
/* Forward a copy of the RTP data to a debug ip/port */
|
|
forward_data(fd->fd, &conn_src->tap_in, msg);
|
|
|
|
rc = rx_rtp(msg);
|
|
|
|
out:
|
|
msgb_free(msg);
|
|
return rc;
|
|
}
|
|
|
|
static int rx_rtp(struct msgb *msg)
|
|
{
|
|
struct osmo_rtp_msg_ctx *mc = OSMO_RTP_MSG_CTX(msg);
|
|
struct mgcp_conn_rtp *conn_src = mc->conn_src;
|
|
struct osmo_sockaddr *from_addr = mc->from_addr;
|
|
struct mgcp_conn *conn = conn_src->conn;
|
|
struct mgcp_trunk *trunk = conn->endp->trunk;
|
|
|
|
LOG_CONN_RTP(conn_src, LOGL_DEBUG, "rx_rtp(%u bytes)\n", msgb_length(msg));
|
|
|
|
mgcp_conn_watchdog_kick(conn_src->conn);
|
|
|
|
/* If AMR is configured for the ingress connection a conversion of the
|
|
* framing mode (octet-aligned vs. bandwith-efficient is explicitly
|
|
* define, then we check if the incoming payload matches that
|
|
* expectation. */
|
|
if (amr_oa_bwe_convert_indicated(conn_src->end.codec)) {
|
|
int oa = amr_oa_check((char*)msgb_data(msg), msgb_length(msg));
|
|
if (oa < 0)
|
|
return -1;
|
|
if (((bool)oa) != conn_src->end.codec->param.amr_octet_aligned)
|
|
return -1;
|
|
}
|
|
|
|
/* Check if the origin of the RTP packet seems plausible */
|
|
if (!trunk->rtp_accept_all && check_rtp_origin(conn_src, from_addr))
|
|
return -1;
|
|
|
|
/* Execute endpoint specific implementation that handles the
|
|
* dispatching of the RTP data */
|
|
return conn->endp->type->dispatch_rtp_cb(msg);
|
|
}
|
|
|
|
/*! set IP Type of Service parameter.
|
|
* \param[in] fd associated file descriptor.
|
|
* \param[in] tos dscp value.
|
|
* \returns 0 on success, -1 on ERROR. */
|
|
int mgcp_set_ip_tos(int fd, int tos)
|
|
{
|
|
int ret;
|
|
ret = setsockopt(fd, IPPROTO_IP, IP_TOS, &tos, sizeof(tos));
|
|
|
|
if (ret < 0)
|
|
return -1;
|
|
return 0;
|
|
}
|
|
|
|
/*! bind RTP port to osmo_fd.
|
|
* \param[in] source_addr source (local) address to bind on.
|
|
* \param[in] fd associated file descriptor.
|
|
* \param[in] port to bind on.
|
|
* \returns 0 on success, -1 on ERROR. */
|
|
int mgcp_create_bind(const char *source_addr, struct osmo_fd *fd, int port)
|
|
{
|
|
int rc;
|
|
|
|
rc = osmo_sock_init2(AF_UNSPEC, SOCK_DGRAM, IPPROTO_UDP, source_addr, port,
|
|
NULL, 0, OSMO_SOCK_F_BIND);
|
|
if (rc < 0) {
|
|
LOGP(DRTP, LOGL_ERROR, "failed to bind UDP port (%s:%i).\n",
|
|
source_addr, port);
|
|
return -1;
|
|
}
|
|
fd->fd = rc;
|
|
LOGP(DRTP, LOGL_DEBUG, "created socket + bound UDP port (%s:%i).\n", source_addr, port);
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* Bind RTP and RTCP port (helper function for mgcp_bind_net_rtp_port()) */
|
|
static int bind_rtp(struct mgcp_config *cfg, const char *source_addr,
|
|
struct mgcp_rtp_end *rtp_end, struct mgcp_endpoint *endp)
|
|
{
|
|
/* NOTE: The port that is used for RTCP is the RTP port incremented by one
|
|
* (e.g. RTP-Port = 16000 ==> RTCP-Port = 16001) */
|
|
|
|
if (mgcp_create_bind(source_addr, &rtp_end->rtp,
|
|
rtp_end->local_port) != 0) {
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR,
|
|
"failed to create RTP port: %s:%d\n",
|
|
source_addr, rtp_end->local_port);
|
|
goto cleanup0;
|
|
}
|
|
|
|
if (mgcp_create_bind(source_addr, &rtp_end->rtcp,
|
|
rtp_end->local_port + 1) != 0) {
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR,
|
|
"failed to create RTCP port: %s:%d\n",
|
|
source_addr, rtp_end->local_port + 1);
|
|
goto cleanup1;
|
|
}
|
|
|
|
/* Set Type of Service (DSCP-Value) as configured via VTY */
|
|
mgcp_set_ip_tos(rtp_end->rtp.fd, cfg->endp_dscp);
|
|
mgcp_set_ip_tos(rtp_end->rtcp.fd, cfg->endp_dscp);
|
|
|
|
if (osmo_fd_register(&rtp_end->rtp) != 0) {
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR,
|
|
"failed to register RTP port %d\n",
|
|
rtp_end->local_port);
|
|
goto cleanup2;
|
|
}
|
|
|
|
if (osmo_fd_register(&rtp_end->rtcp) != 0) {
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR,
|
|
"failed to register RTCP port %d\n",
|
|
rtp_end->local_port + 1);
|
|
goto cleanup3;
|
|
}
|
|
|
|
return 0;
|
|
|
|
cleanup3:
|
|
osmo_fd_unregister(&rtp_end->rtp);
|
|
cleanup2:
|
|
close(rtp_end->rtcp.fd);
|
|
rtp_end->rtcp.fd = -1;
|
|
cleanup1:
|
|
close(rtp_end->rtp.fd);
|
|
rtp_end->rtp.fd = -1;
|
|
cleanup0:
|
|
return -1;
|
|
}
|
|
|
|
/*! bind RTP port to endpoint/connection.
|
|
* \param[in] endp endpoint that holds the RTP connection.
|
|
* \param[in] rtp_port port number to bind on.
|
|
* \param[in] conn associated RTP connection.
|
|
* \returns 0 on success, -1 on ERROR. */
|
|
int mgcp_bind_net_rtp_port(struct mgcp_endpoint *endp, int rtp_port,
|
|
struct mgcp_conn_rtp *conn)
|
|
{
|
|
char name[512];
|
|
struct mgcp_rtp_end *end;
|
|
|
|
snprintf(name, sizeof(name), "%s-%s", conn->conn->name, conn->conn->id);
|
|
end = &conn->end;
|
|
|
|
if (end->rtp.fd != -1 || end->rtcp.fd != -1) {
|
|
LOGPENDP(endp, DRTP, LOGL_ERROR, "%u was already bound on conn:%s\n",
|
|
rtp_port, mgcp_conn_dump(conn->conn));
|
|
|
|
/* Double bindings should never occour! Since we always allocate
|
|
* connections dynamically and free them when they are not
|
|
* needed anymore, there must be no previous binding leftover.
|
|
* Should there be a connection bound twice, we have a serious
|
|
* problem and must exit immediately! */
|
|
OSMO_ASSERT(false);
|
|
}
|
|
|
|
end->local_port = rtp_port;
|
|
osmo_fd_setup(&end->rtp, -1, OSMO_FD_READ, rtp_data_net, conn, 0);
|
|
osmo_fd_setup(&end->rtcp, -1, OSMO_FD_READ, rtp_data_net, conn, 0);
|
|
|
|
return bind_rtp(endp->cfg, conn->end.local_addr, end, endp);
|
|
}
|
|
|
|
/*! free allocated RTP and RTCP ports.
|
|
* \param[in] end RTP end */
|
|
void mgcp_free_rtp_port(struct mgcp_rtp_end *end)
|
|
{
|
|
if (end->rtp.fd != -1) {
|
|
close(end->rtp.fd);
|
|
end->rtp.fd = -1;
|
|
osmo_fd_unregister(&end->rtp);
|
|
}
|
|
|
|
if (end->rtcp.fd != -1) {
|
|
close(end->rtcp.fd);
|
|
end->rtcp.fd = -1;
|
|
osmo_fd_unregister(&end->rtcp);
|
|
}
|
|
}
|