1283 lines
40 KiB
C
1283 lines
40 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/netif/rtp.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_internal.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/debug.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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#define RTP_BUF_SIZE 4096
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enum {
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MGCP_PROTO_RTP,
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MGCP_PROTO_RTCP,
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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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int rc;
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endp = conn->conn->endp;
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/* Try probing the local IP-Address */
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if (endp->cfg->net_ports.bind_addr_probe && conn->end.addr.s_addr != 0) {
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rc = osmo_sock_local_ip(addr, inet_ntoa(conn->end.addr));
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if (rc < 0)
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LOGP(DRTP, LOGL_ERROR,
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"endpoint:0x%x CI:%s local interface auto detection failed, using configured addresses...\n",
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ENDPOINT_NUMBER(endp), conn->conn->id);
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else {
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LOGP(DRTP, LOGL_DEBUG,
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"endpoint:0x%x CI:%s selected local rtp bind ip %s by probing using remote ip %s\n",
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ENDPOINT_NUMBER(endp), conn->conn->id, addr,
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inet_ntoa(conn->end.addr));
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return;
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}
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}
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/* Select from preconfigured IP-Addresses */
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if (endp->cfg->net_ports.bind_addr) {
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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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osmo_strlcpy(addr, endp->cfg->net_ports.bind_addr, INET_ADDRSTRLEN);
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LOGP(DRTP, LOGL_DEBUG,
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"endpoint:0x%x CI:%s using configured rtp bind ip as local bind ip %s\n",
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ENDPOINT_NUMBER(endp), conn->conn->id, 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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osmo_strlcpy(addr, endp->cfg->source_addr, INET_ADDRSTRLEN);
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LOGP(DRTP, LOGL_DEBUG,
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"endpoint:0x%x CI:%s using mgcp bind ip as local rtp bind ip: %s\n",
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ENDPOINT_NUMBER(endp), conn->conn->id, addr);
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}
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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
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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 in_addr *addr, int port, char *buf, int len)
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{
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struct sockaddr_in out;
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LOGP(DRTP, LOGL_DEBUG,
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"sending %i bytes length packet to %s:%u ...\n",
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len, inet_ntoa(*addr), ntohs(port));
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out.sin_family = AF_INET;
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out.sin_port = port;
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memcpy(&out.sin_addr, addr, sizeof(*addr));
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return sendto(fd, buf, len, 0, (struct sockaddr *)&out, sizeof(out));
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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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LOGP(DRTP, LOGL_DEBUG,
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"endpoint:0x%x sending dummy packet...\n", ENDPOINT_NUMBER(endp));
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LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x conn:%s\n",
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ENDPOINT_NUMBER(endp), 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->tcfg->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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LOGP(DRTP, LOGL_ERROR,
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"endpoint:0x%x Failed to send dummy %s packet.\n",
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ENDPOINT_NUMBER(endp), 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 sockaddr_in *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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/* 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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LOGP(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 0x%x 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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ENDPOINT_NUMBER(endp), sstate->ssrc, timestamp,
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inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
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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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LOGP(DRTP, LOGL_NOTICE,
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"The %s timestamp delta is %d "
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"on 0x%x SSRC: %u timestamp: %u "
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"from %s:%d\n",
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text, tsdelta,
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ENDPOINT_NUMBER(endp), sstate->ssrc, timestamp,
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inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
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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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LOGP(DRTP, LOGL_INFO,
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"The %s timestamp delta changes from %d to %d "
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"on 0x%x SSRC: %u timestamp: %u from %s:%d\n",
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text, sstate->last_tsdelta, tsdelta,
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ENDPOINT_NUMBER(endp), sstate->ssrc, timestamp,
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inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
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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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LOGP(DRTP, LOGL_NOTICE,
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"The %s timestamp has an alignment error of %d "
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"on 0x%x 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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ENDPOINT_NUMBER(endp), 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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inet_ntoa(addr->sin_addr), ntohs(addr->sin_port),
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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 sockaddr_in *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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if (tsdelta == 0) {
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tsdelta = state->out_stream.last_tsdelta;
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if (tsdelta != 0) {
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LOGP(DRTP, LOGL_NOTICE,
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"A fixed packet duration is not available on 0x%x, "
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"using last output timestamp delta instead: %d "
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"from %s:%d\n",
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ENDPOINT_NUMBER(endp), tsdelta,
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inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
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} else {
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tsdelta = rtp_end->codec->rate * 20 / 1000;
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LOGP(DRTP, LOGL_NOTICE,
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"Fixed packet duration and last timestamp delta "
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"are not available on 0x%x, "
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"using fixed 20ms instead: %d "
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"from %s:%d\n",
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ENDPOINT_NUMBER(endp), tsdelta,
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inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
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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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LOGP(DRTP, LOGL_NOTICE,
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"Timestamp offset change on 0x%x SSRC: %u "
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"SeqNo delta: %d, TS offset: %d, "
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"from %s:%d\n",
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ENDPOINT_NUMBER(endp), state->in_stream.ssrc,
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delta_seq, state->patch.timestamp_offset,
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inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
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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 sockaddr_in *addr,
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uint32_t timestamp)
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{
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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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LOGP(DRTP, LOGL_NOTICE,
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"Corrected timestamp alignment error of %d on 0x%x SSRC: %u "
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"new TS offset: %d, "
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"from %s:%d\n",
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ts_error,
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ENDPOINT_NUMBER(endp), state->in_stream.ssrc,
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state->patch.timestamp_offset, inet_ntoa(addr->sin_addr),
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ntohs(addr->sin_port));
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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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LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x transcoding disabled\n",
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ENDPOINT_NUMBER(endp));
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return 0;
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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)
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{
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LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x transcoding disabled\n",
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ENDPOINT_NUMBER(endp));
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return 0;
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}
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void mgcp_get_net_downlink_format_default(struct mgcp_endpoint *endp,
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int *payload_type,
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const char **audio_name,
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const char **fmtp_extra,
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struct mgcp_conn_rtp *conn)
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{
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LOGP(DRTP, LOGL_DEBUG,
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"endpoint:0x%x conn:%s using format defaults\n",
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ENDPOINT_NUMBER(endp), mgcp_conn_dump(conn->conn));
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*payload_type = conn->end.codec->payload_type;
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*audio_name = conn->end.codec->audio_name;
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*fmtp_extra = conn->end.fmtp_extra;
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}
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|
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void mgcp_rtp_annex_count(struct mgcp_endpoint *endp,
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struct mgcp_rtp_state *state, const uint16_t seq,
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const int32_t transit, const uint32_t ssrc)
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{
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int32_t d;
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|
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/* initialize or re-initialize */
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if (!state->stats.initialized || state->stats.ssrc != ssrc) {
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state->stats.initialized = 1;
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state->stats.base_seq = seq;
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state->stats.max_seq = seq - 1;
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state->stats.ssrc = ssrc;
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state->stats.jitter = 0;
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state->stats.transit = transit;
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state->stats.cycles = 0;
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} else {
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uint16_t udelta;
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|
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/* The below takes the shape of the validation of
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* Appendix A. Check if there is something weird with
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* the sequence number, otherwise check for a wrap
|
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* around in the sequence number.
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* It can't wrap during the initialization so let's
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* skip it here. The Appendix A probably doesn't have
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* this issue because of the probation. */
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udelta = seq - state->stats.max_seq;
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if (udelta < RTP_MAX_DROPOUT) {
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if (seq < state->stats.max_seq)
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state->stats.cycles += RTP_SEQ_MOD;
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} else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
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LOGP(DRTP, LOGL_NOTICE,
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"RTP seqno made a very large jump on 0x%x delta: %u\n",
|
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ENDPOINT_NUMBER(endp), udelta);
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}
|
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}
|
|
|
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/* Calculate the jitter between the two packages. The TS should be
|
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* 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;
|
|
}
|
|
|
|
/* 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 sockaddr_in *addr, char *data, int len)
|
|
{
|
|
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;
|
|
|
|
if (len < sizeof(*rtp_hdr))
|
|
return;
|
|
|
|
rtp_hdr = (struct rtp_hdr *)data;
|
|
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 */
|
|
LOGP(DRTP, LOGL_INFO,
|
|
"endpoint:0x%x initializing stream, SSRC: %u timestamp: %u "
|
|
"pkt-duration: %d, from %s:%d\n",
|
|
ENDPOINT_NUMBER(endp), state->in_stream.ssrc,
|
|
state->patch.seq_offset, state->packet_duration,
|
|
inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
|
|
if (state->packet_duration == 0) {
|
|
state->packet_duration =
|
|
rtp_end->codec->rate * 20 / 1000;
|
|
LOGP(DRTP, LOGL_NOTICE,
|
|
"endpoint:0x%x fixed packet duration is not available, "
|
|
"using fixed 20ms instead: %d from %s:%d\n",
|
|
ENDPOINT_NUMBER(endp), state->packet_duration,
|
|
inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
|
|
}
|
|
} else if (state->in_stream.ssrc != ssrc) {
|
|
LOGP(DRTP, LOGL_NOTICE,
|
|
"endpoint:0x%x SSRC changed: %u -> %u "
|
|
"from %s:%d\n",
|
|
ENDPOINT_NUMBER(endp),
|
|
state->in_stream.ssrc, rtp_hdr->ssrc,
|
|
inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
|
|
|
|
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;
|
|
|
|
LOGP(DRTP, LOGL_NOTICE,
|
|
"endpoint:0x%x SSRC patching enabled, SSRC: %u "
|
|
"SeqNo offset: %d, TS offset: %d "
|
|
"from %s:%d\n",
|
|
ENDPOINT_NUMBER(endp), state->in_stream.ssrc,
|
|
state->patch.seq_offset, state->patch.timestamp_offset,
|
|
inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
|
|
}
|
|
|
|
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
|
|
DEBUGP(DRTP,
|
|
"endpoint:0x%x payload hdr payload %u -> endp payload %u\n",
|
|
ENDPOINT_NUMBER(endp), rtp_hdr->payload_type, payload);
|
|
rtp_hdr->payload_type = payload;
|
|
#endif
|
|
}
|
|
|
|
/* 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, const char *buf,
|
|
int len)
|
|
{
|
|
int rc;
|
|
|
|
if (!tap->enabled)
|
|
return;
|
|
|
|
rc = sendto(fd, buf, len, 0, (struct sockaddr *)&tap->forward,
|
|
sizeof(tap->forward));
|
|
|
|
if (rc < 0)
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"Forwarding tapped (debug) voice data failed.\n");
|
|
}
|
|
|
|
/*! 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 sockaddr_in *addr,
|
|
char *buf, int len, 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_config *tcfg = endp->tcfg;
|
|
struct mgcp_rtp_end *rtp_end;
|
|
struct mgcp_rtp_state *rtp_state;
|
|
char *dest_name;
|
|
|
|
OSMO_ASSERT(conn_src);
|
|
OSMO_ASSERT(conn_dst);
|
|
|
|
if (is_rtp) {
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"endpoint:0x%x delivering RTP packet...\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
} else {
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"endpoint:0x%x delivering RTCP packet...\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
}
|
|
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"endpoint:0x%x loop:%d, mode:%d%s\n",
|
|
ENDPOINT_NUMBER(endp), tcfg->audio_loop, conn_src->conn->mode,
|
|
conn_src->conn->mode == MGCP_CONN_LOOPBACK ? " (loopback)" : "");
|
|
|
|
/* 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;
|
|
|
|
if (!rtp_end->output_enabled) {
|
|
rate_ctr_inc(&conn_dst->rate_ctr_group->ctr[RTP_DROPPED_PACKETS_CTR]);
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"endpoint:0x%x output disabled, drop to %s %s "
|
|
"rtp_port:%u rtcp_port:%u\n",
|
|
ENDPOINT_NUMBER(endp),
|
|
dest_name,
|
|
inet_ntoa(rtp_end->addr),
|
|
ntohs(rtp_end->rtp_port), ntohs(rtp_end->rtcp_port)
|
|
);
|
|
} else if (is_rtp) {
|
|
int cont;
|
|
int nbytes = 0;
|
|
int buflen = len;
|
|
do {
|
|
/* Run transcoder */
|
|
cont = endp->cfg->rtp_processing_cb(endp, rtp_end,
|
|
buf, &buflen,
|
|
RTP_BUF_SIZE);
|
|
if (cont < 0)
|
|
break;
|
|
|
|
if (addr)
|
|
mgcp_patch_and_count(endp, rtp_state, rtp_end,
|
|
addr, buf, buflen);
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"endpoint:0x%x process/send to %s %s "
|
|
"rtp_port:%u rtcp_port:%u\n",
|
|
ENDPOINT_NUMBER(endp), dest_name,
|
|
inet_ntoa(rtp_end->addr), 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,
|
|
buf, buflen);
|
|
|
|
/* 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 = (uint8_t *) & buf[12];
|
|
if (data[0] == 0xe0) {
|
|
data[0] = 0xe4;
|
|
data[1] = 0x00;
|
|
rtp_state->patched_first_rtp_payload = true;
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"endpoint:0x%x Patching over first two bytes"
|
|
" to fake an IuUP Initialization Ack\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
}
|
|
}
|
|
|
|
len = mgcp_udp_send(rtp_end->rtp.fd,
|
|
&rtp_end->addr,
|
|
rtp_end->rtp_port, buf, buflen);
|
|
|
|
if (len <= 0)
|
|
return len;
|
|
|
|
rate_ctr_inc(&conn_dst->rate_ctr_group->ctr[RTP_PACKETS_TX_CTR]);
|
|
rate_ctr_add(&conn_dst->rate_ctr_group->ctr[RTP_OCTETS_TX_CTR], len);
|
|
|
|
nbytes += len;
|
|
buflen = cont;
|
|
} while (buflen > 0);
|
|
return nbytes;
|
|
} else if (!tcfg->omit_rtcp) {
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"endpoint:0x%x send to %s %s rtp_port:%u rtcp_port:%u\n",
|
|
ENDPOINT_NUMBER(endp),
|
|
dest_name,
|
|
inet_ntoa(rtp_end->addr),
|
|
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, buf, len);
|
|
|
|
rate_ctr_inc(&conn_dst->rate_ctr_group->ctr[RTP_PACKETS_TX_CTR]);
|
|
rate_ctr_add(&conn_dst->rate_ctr_group->ctr[RTP_OCTETS_TX_CTR], len);
|
|
|
|
return len;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* Helper function for mgcp_recv(),
|
|
Receive one RTP Packet + Originating address from file descriptor */
|
|
static int receive_from(struct mgcp_endpoint *endp, int fd,
|
|
struct sockaddr_in *addr, char *buf, int bufsize)
|
|
{
|
|
int rc;
|
|
socklen_t slen = sizeof(*addr);
|
|
struct sockaddr_in addr_sink;
|
|
char buf_sink[RTP_BUF_SIZE];
|
|
bool tossed = false;
|
|
|
|
if (!addr)
|
|
addr = &addr_sink;
|
|
if (!buf) {
|
|
tossed = true;
|
|
buf = buf_sink;
|
|
bufsize = sizeof(buf_sink);
|
|
}
|
|
|
|
rc = recvfrom(fd, buf, bufsize, 0, (struct sockaddr *)addr, &slen);
|
|
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"receiving %u bytes length packet from %s:%u ...\n",
|
|
rc, inet_ntoa(addr->sin_addr), ntohs(addr->sin_port));
|
|
|
|
if (rc < 0) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x failed to receive packet, errno: %d/%s\n",
|
|
ENDPOINT_NUMBER(endp), errno, strerror(errno));
|
|
return -1;
|
|
}
|
|
|
|
if (tossed) {
|
|
LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x packet tossed\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
}
|
|
|
|
return rc;
|
|
}
|
|
|
|
/* 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 sockaddr_in *addr)
|
|
{
|
|
struct mgcp_endpoint *endp;
|
|
endp = conn->conn->endp;
|
|
|
|
/* Note: Check if the inbound RTP data comes from the same host to
|
|
* which we send our outgoing RTP traffic. */
|
|
if (memcmp(&addr->sin_addr, &conn->end.addr, sizeof(addr->sin_addr))
|
|
!= 0) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x data from wrong address: %s, ",
|
|
ENDPOINT_NUMBER(endp), inet_ntoa(addr->sin_addr));
|
|
LOGPC(DRTP, LOGL_ERROR, "expected: %s\n",
|
|
inet_ntoa(conn->end.addr));
|
|
LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x packet tossed\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
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 (conn->end.rtp_port != addr->sin_port &&
|
|
conn->end.rtcp_port != addr->sin_port) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x data from wrong source port: %d, ",
|
|
ENDPOINT_NUMBER(endp), ntohs(addr->sin_port));
|
|
LOGPC(DRTP, LOGL_ERROR,
|
|
"expected: %d for RTP or %d for RTCP\n",
|
|
ntohs(conn->end.rtp_port), ntohs(conn->end.rtcp_port));
|
|
LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x packet tossed\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
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)
|
|
{
|
|
struct mgcp_endpoint *endp;
|
|
endp = conn->conn->endp;
|
|
|
|
/* Note: it is legal to create a connection but never setting a port
|
|
* and IP-address for outgoing data. */
|
|
if (strcmp(inet_ntoa(conn->end.addr), "0.0.0.0") == 0 && conn->end.rtp_port == 0) {
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"endpoint:0x%x destination IP-address and rtp port is (not yet) known\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
return -1;
|
|
}
|
|
|
|
if (strcmp(inet_ntoa(conn->end.addr), "0.0.0.0") == 0) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x destination IP-address is invalid\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
return -1;
|
|
}
|
|
|
|
if (conn->end.rtp_port == 0) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x destination rtp port is invalid\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
return -1;
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* Receive RTP data from a specified source connection and dispatch it to a
|
|
* destination connection. */
|
|
static int mgcp_recv(int *proto, struct sockaddr_in *addr, char *buf,
|
|
unsigned int buf_size, struct osmo_fd *fd)
|
|
{
|
|
struct mgcp_endpoint *endp;
|
|
struct mgcp_conn_rtp *conn;
|
|
struct mgcp_trunk_config *tcfg;
|
|
int rc;
|
|
|
|
conn = (struct mgcp_conn_rtp*) fd->data;
|
|
endp = conn->conn->endp;
|
|
tcfg = endp->tcfg;
|
|
|
|
LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x receiving RTP/RTCP packet...\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
|
|
rc = receive_from(endp, fd->fd, addr, buf, buf_size);
|
|
if (rc <= 0)
|
|
return -1;
|
|
*proto = fd == &conn->end.rtp ? MGCP_PROTO_RTP : MGCP_PROTO_RTCP;
|
|
|
|
LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x ", ENDPOINT_NUMBER(endp));
|
|
LOGPC(DRTP, LOGL_DEBUG, "receiving from %s %s %d\n",
|
|
conn->conn->name, inet_ntoa(addr->sin_addr),
|
|
ntohs(addr->sin_port));
|
|
LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x conn:%s\n", ENDPOINT_NUMBER(endp),
|
|
mgcp_conn_dump(conn->conn));
|
|
|
|
/* Check if the origin of the RTP packet seems plausible */
|
|
if (tcfg->rtp_accept_all == 0) {
|
|
if (check_rtp_origin(conn, addr) != 0)
|
|
return -1;
|
|
}
|
|
|
|
/* Filter out dummy message */
|
|
if (rc == 1 && buf[0] == MGCP_DUMMY_LOAD) {
|
|
LOGP(DRTP, LOGL_NOTICE,
|
|
"endpoint:0x%x dummy message received\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x packet tossed\n", ENDPOINT_NUMBER(endp));
|
|
return 0;
|
|
}
|
|
|
|
/* Increment RX statistics */
|
|
rate_ctr_inc(&conn->rate_ctr_group->ctr[RTP_PACKETS_RX_CTR]);
|
|
rate_ctr_add(&conn->rate_ctr_group->ctr[RTP_OCTETS_RX_CTR], rc);
|
|
|
|
/* Forward a copy of the RTP data to a debug ip/port */
|
|
forward_data(fd->fd, &conn->tap_in, buf, rc);
|
|
|
|
return rc;
|
|
}
|
|
|
|
/* Send RTP data. Possible options are standard RTP packet
|
|
* transmission or trsmission via an osmux connection */
|
|
static int mgcp_send_rtp(int proto, struct sockaddr_in *addr, char *buf,
|
|
unsigned int buf_size,
|
|
struct mgcp_conn_rtp *conn_src,
|
|
struct mgcp_conn_rtp *conn_dst)
|
|
{
|
|
struct mgcp_endpoint *endp;
|
|
endp = conn_src->conn->endp;
|
|
|
|
LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x destin conn:%s\n",
|
|
ENDPOINT_NUMBER(endp), 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:
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"endpoint:0x%x endpoint type is MGCP_RTP_DEFAULT, "
|
|
"using mgcp_send() to forward data directly\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
return mgcp_send(endp, proto == MGCP_PROTO_RTP,
|
|
addr, buf, buf_size, conn_src, conn_dst);
|
|
case MGCP_OSMUX_BSC_NAT:
|
|
case MGCP_OSMUX_BSC:
|
|
LOGP(DRTP, LOGL_DEBUG,
|
|
"endpoint:0x%x endpoint type is MGCP_OSMUX_BSC_NAT, "
|
|
"using osmux_xfrm_to_osmux() to forward data through OSMUX\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
return osmux_xfrm_to_osmux(buf, buf_size, 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 */
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x bad MGCP type -- data discarded!\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
|
|
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(int proto, struct sockaddr_in *addr, char *buf,
|
|
unsigned int buf_size, struct mgcp_conn *conn)
|
|
{
|
|
struct mgcp_conn *conn_dst;
|
|
struct mgcp_endpoint *endp;
|
|
endp = conn->endp;
|
|
|
|
/*! NOTE: This callback function implements the endpoint specific
|
|
* dispatch bahviour 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. */
|
|
|
|
/* 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) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x unable to find destination conn\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
return -1;
|
|
}
|
|
|
|
/* The destination conn is not an RTP connection */
|
|
if (conn_dst->type != MGCP_CONN_TYPE_RTP) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x unable to find suitable destination conn\n",
|
|
ENDPOINT_NUMBER(endp));
|
|
return -1;
|
|
}
|
|
|
|
/* Dispatch RTP packet to destination RTP connection */
|
|
return mgcp_send_rtp(proto, addr, buf,
|
|
buf_size, &conn->u.rtp, &conn_dst->u.rtp);
|
|
|
|
}
|
|
|
|
/*! 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).
|
|
* \returns 0 on success, -1 on ERROR. */
|
|
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) {
|
|
conn_cleanup->priv = NULL;
|
|
}
|
|
}
|
|
|
|
/* 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 sockaddr_in addr;
|
|
|
|
char buf[RTP_BUF_SIZE];
|
|
int proto;
|
|
int len;
|
|
|
|
conn_src = (struct mgcp_conn_rtp *)fd->data;
|
|
OSMO_ASSERT(conn_src);
|
|
endp = conn_src->conn->endp;
|
|
OSMO_ASSERT(endp);
|
|
|
|
LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x source conn:%s\n",
|
|
ENDPOINT_NUMBER(endp), mgcp_conn_dump(conn_src->conn));
|
|
|
|
/* Receive packet */
|
|
len = mgcp_recv(&proto, &addr, buf, sizeof(buf), fd);
|
|
if (len < 0)
|
|
return -1;
|
|
|
|
/* Check if the connection is in loopback mode, if yes, just send the
|
|
* incoming data back to the origin */
|
|
|
|
if (conn_src->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_src->end.rtp_port == 0) {
|
|
conn_src->end.addr = addr.sin_addr;
|
|
conn_src->end.rtp_port = addr.sin_port;
|
|
}
|
|
return mgcp_send_rtp(proto, &addr, buf,
|
|
len, conn_src, conn_src);
|
|
}
|
|
|
|
/* Execute endpoint specific implementation that handles the
|
|
* dispatching of the RTP data */
|
|
return endp->type->dispatch_rtp_cb(proto, &addr, buf, len,
|
|
conn_src->conn);
|
|
}
|
|
|
|
/*! 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_INET, 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, int endpno)
|
|
{
|
|
/* 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) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x failed to create RTP port: %s:%d\n", endpno,
|
|
source_addr, rtp_end->local_port);
|
|
goto cleanup0;
|
|
}
|
|
|
|
if (mgcp_create_bind(source_addr, &rtp_end->rtcp,
|
|
rtp_end->local_port + 1) != 0) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x failed to create RTCP port: %s:%d\n", endpno,
|
|
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);
|
|
|
|
rtp_end->rtp.when = BSC_FD_READ;
|
|
if (osmo_fd_register(&rtp_end->rtp) != 0) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x failed to register RTP port %d\n", endpno,
|
|
rtp_end->local_port);
|
|
goto cleanup2;
|
|
}
|
|
|
|
rtp_end->rtcp.when = BSC_FD_READ;
|
|
if (osmo_fd_register(&rtp_end->rtcp) != 0) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x failed to register RTCP port %d\n", endpno,
|
|
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;
|
|
char local_ip_addr[INET_ADDRSTRLEN];
|
|
|
|
snprintf(name, sizeof(name), "%s-%s", conn->conn->name, conn->conn->id);
|
|
end = &conn->end;
|
|
|
|
if (end->rtp.fd != -1 || end->rtcp.fd != -1) {
|
|
LOGP(DRTP, LOGL_ERROR,
|
|
"endpoint:0x%x %u was already bound on conn:%s\n",
|
|
ENDPOINT_NUMBER(endp), 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;
|
|
end->rtp.cb = rtp_data_net;
|
|
end->rtp.data = conn;
|
|
end->rtcp.data = conn;
|
|
end->rtcp.cb = rtp_data_net;
|
|
|
|
mgcp_get_local_addr(local_ip_addr, conn);
|
|
|
|
return bind_rtp(endp->cfg, local_ip_addr, end,
|
|
ENDPOINT_NUMBER(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);
|
|
}
|
|
}
|