| /* A Media Gateway Control Protocol Media Gateway: RFC 3435 */ |
| /* The protocol implementation */ |
| |
| /* |
| * (C) 2009-2012 by Holger Hans Peter Freyther <zecke@selfish.org> |
| * (C) 2009-2012 by On-Waves |
| * All Rights Reserved |
| * |
| * This program is free software; you can redistribute it and/or modify |
| * it under the terms of the GNU Affero General Public License as published by |
| * the Free Software Foundation; either version 3 of the License, or |
| * (at your option) any later version. |
| * |
| * This program is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
| * GNU Affero General Public License for more details. |
| * |
| * You should have received a copy of the GNU Affero General Public License |
| * along with this program. If not, see <http://www.gnu.org/licenses/>. |
| * |
| */ |
| |
| #include <string.h> |
| #include <stdlib.h> |
| #include <unistd.h> |
| #include <errno.h> |
| #include <time.h> |
| #include <limits.h> |
| #include <arpa/inet.h> |
| |
| #include <osmocom/core/msgb.h> |
| #include <osmocom/core/select.h> |
| #include <osmocom/core/socket.h> |
| #include <osmocom/core/byteswap.h> |
| #include <osmocom/netif/rtp.h> |
| #include <osmocom/netif/amr.h> |
| #include <osmocom/mgcp/mgcp.h> |
| #include <osmocom/mgcp/mgcp_common.h> |
| #include <osmocom/mgcp/mgcp_internal.h> |
| #include <osmocom/mgcp/mgcp_stat.h> |
| #include <osmocom/mgcp/osmux.h> |
| #include <osmocom/mgcp/mgcp_conn.h> |
| #include <osmocom/mgcp/mgcp_endp.h> |
| #include <osmocom/mgcp/mgcp_trunk.h> |
| #include <osmocom/mgcp/mgcp_codec.h> |
| #include <osmocom/mgcp/debug.h> |
| #include <osmocom/codec/codec.h> |
| |
| |
| #define RTP_SEQ_MOD (1 << 16) |
| #define RTP_MAX_DROPOUT 3000 |
| #define RTP_MAX_MISORDER 100 |
| #define RTP_BUF_SIZE 4096 |
| |
| enum rtp_proto { |
| MGCP_PROTO_RTP, |
| MGCP_PROTO_RTCP, |
| }; |
| |
| static void rtpconn_rate_ctr_add(struct mgcp_conn_rtp *conn_rtp, struct mgcp_endpoint *endp, |
| int id, int inc) |
| { |
| struct rate_ctr_group *conn_stats = conn_rtp->rate_ctr_group; |
| struct rate_ctr_group *mgw_stats = endp->trunk->ratectr.all_rtp_conn_stats; |
| |
| /* add to both the per-connection and the global stats */ |
| rate_ctr_add(&conn_stats->ctr[id], inc); |
| rate_ctr_add(&mgw_stats->ctr[id], inc); |
| } |
| |
| static void rtpconn_rate_ctr_inc(struct mgcp_conn_rtp *conn_rtp, struct mgcp_endpoint *endp, int id) |
| { |
| rtpconn_rate_ctr_add(conn_rtp, endp, id, 1); |
| } |
| |
| static int rx_rtp(struct msgb *msg); |
| |
| /*! Determine the local rtp bind IP-address. |
| * \param[out] addr caller provided memory to store the resulting IP-Address. |
| * \param[in] endp mgcp endpoint, that holds a copy of the VTY parameters. |
| * |
| * The local bind IP-address is automatically selected by probing the |
| * IP-Address of the interface that is pointing towards the remote IP-Address, |
| * if no remote IP-Address is known yet, the statically configured |
| * IP-Addresses are used as fallback. */ |
| void mgcp_get_local_addr(char *addr, struct mgcp_conn_rtp *conn) |
| { |
| |
| struct mgcp_endpoint *endp; |
| int rc; |
| endp = conn->conn->endp; |
| |
| /* Try probing the local IP-Address */ |
| if (endp->cfg->net_ports.bind_addr_probe && conn->end.addr.s_addr != 0) { |
| rc = osmo_sock_local_ip(addr, inet_ntoa(conn->end.addr)); |
| if (rc < 0) |
| LOGPCONN(conn->conn, DRTP, LOGL_ERROR, |
| "local interface auto detection failed, using configured addresses...\n"); |
| else { |
| LOGPCONN(conn->conn, DRTP, LOGL_DEBUG, |
| "selected local rtp bind ip %s by probing using remote ip %s\n", |
| addr, inet_ntoa(conn->end.addr)); |
| return; |
| } |
| } |
| |
| /* Select from preconfigured IP-Addresses. We don't have bind_addr for Osmux (yet?). */ |
| if (endp->cfg->net_ports.bind_addr) { |
| /* Check there is a bind IP for the RTP traffic configured, |
| * if so, use that IP-Address */ |
| osmo_strlcpy(addr, endp->cfg->net_ports.bind_addr, INET_ADDRSTRLEN); |
| LOGPCONN(conn->conn, DRTP, LOGL_DEBUG, |
| "using configured rtp bind ip as local bind ip %s\n", |
| addr); |
| } else { |
| /* No specific bind IP is configured for the RTP traffic, so |
| * assume the IP where we listen for incoming MGCP messages |
| * as bind IP */ |
| osmo_strlcpy(addr, endp->cfg->source_addr, INET_ADDRSTRLEN); |
| LOGPCONN(conn->conn, DRTP, LOGL_DEBUG, |
| "using mgcp bind ip as local rtp bind ip: %s\n", addr); |
| } |
| } |
| |
| /* This does not need to be a precision timestamp and |
| * is allowed to wrap quite fast. The returned value is |
| * 1/codec_rate seconds. */ |
| static uint32_t get_current_ts(unsigned codec_rate) |
| { |
| struct timespec tp; |
| uint64_t ret; |
| |
| if (!codec_rate) |
| return 0; |
| |
| memset(&tp, 0, sizeof(tp)); |
| if (clock_gettime(CLOCK_MONOTONIC, &tp) != 0) |
| LOGP(DRTP, LOGL_NOTICE, "Getting the clock failed.\n"); |
| |
| /* convert it to 1/unit seconds */ |
| ret = tp.tv_sec; |
| ret *= codec_rate; |
| ret += (int64_t) tp.tv_nsec * codec_rate / 1000 / 1000 / 1000; |
| |
| return ret; |
| } |
| |
| /*! send udp packet. |
| * \param[in] fd associated file descriptor. |
| * \param[in] addr destination ip-address. |
| * \param[in] port destination UDP port (network byte order). |
| * \param[in] buf buffer that holds the data to be send. |
| * \param[in] len length of the data to be sent. |
| * \returns bytes sent, -1 on error. */ |
| int mgcp_udp_send(int fd, struct in_addr *addr, int port, char *buf, int len) |
| { |
| struct sockaddr_in out; |
| |
| LOGP(DRTP, LOGL_DEBUG, |
| "sending %i bytes length packet to %s:%u ...\n", |
| len, inet_ntoa(*addr), ntohs(port)); |
| |
| out.sin_family = AF_INET; |
| out.sin_port = port; |
| memcpy(&out.sin_addr, addr, sizeof(*addr)); |
| |
| return sendto(fd, buf, len, 0, (struct sockaddr *)&out, sizeof(out)); |
| } |
| |
| /*! send RTP dummy packet (to keep NAT connection open). |
| * \param[in] endp mcgp endpoint that holds the RTP connection. |
| * \param[in] conn associated RTP connection. |
| * \returns bytes sent, -1 on error. */ |
| int mgcp_send_dummy(struct mgcp_endpoint *endp, struct mgcp_conn_rtp *conn) |
| { |
| static char buf[] = { MGCP_DUMMY_LOAD }; |
| int rc; |
| int was_rtcp = 0; |
| |
| OSMO_ASSERT(endp); |
| OSMO_ASSERT(conn); |
| |
| LOGPCONN(conn->conn, DRTP, LOGL_DEBUG,"sending dummy packet... %s\n", |
| mgcp_conn_dump(conn->conn)); |
| |
| rc = mgcp_udp_send(conn->end.rtp.fd, &conn->end.addr, |
| conn->end.rtp_port, buf, 1); |
| |
| if (rc == -1) |
| goto failed; |
| |
| if (endp->trunk->omit_rtcp) |
| return rc; |
| |
| was_rtcp = 1; |
| rc = mgcp_udp_send(conn->end.rtcp.fd, &conn->end.addr, |
| conn->end.rtcp_port, buf, 1); |
| |
| if (rc >= 0) |
| return rc; |
| |
| failed: |
| LOGPCONN(conn->conn, DRTP, LOGL_ERROR, |
| "Failed to send dummy %s packet.\n", |
| was_rtcp ? "RTCP" : "RTP"); |
| |
| return -1; |
| } |
| |
| /* Compute timestamp alignment error */ |
| static int32_t ts_alignment_error(struct mgcp_rtp_stream_state *sstate, |
| int ptime, uint32_t timestamp) |
| { |
| int32_t timestamp_delta; |
| |
| if (ptime == 0) |
| return 0; |
| |
| /* Align according to: T - Tlast = k * Tptime */ |
| timestamp_delta = timestamp - sstate->last_timestamp; |
| |
| return timestamp_delta % ptime; |
| } |
| |
| /* Check timestamp and sequence number for plausibility */ |
| static int check_rtp_timestamp(struct mgcp_endpoint *endp, |
| struct mgcp_rtp_state *state, |
| struct mgcp_rtp_stream_state *sstate, |
| struct mgcp_rtp_end *rtp_end, |
| struct sockaddr_in *addr, |
| uint16_t seq, uint32_t timestamp, |
| const char *text, int32_t * tsdelta_out) |
| { |
| int32_t tsdelta; |
| int32_t timestamp_error; |
| |
| /* Not fully intialized, skip */ |
| if (sstate->last_tsdelta == 0 && timestamp == sstate->last_timestamp) |
| return 0; |
| |
| if (seq == sstate->last_seq) { |
| if (timestamp != sstate->last_timestamp) { |
| rate_ctr_inc(sstate->err_ts_ctr); |
| LOGPENDP(endp, DRTP, LOGL_ERROR, |
| "The %s timestamp delta is != 0 but the sequence " |
| "number %d is the same, " |
| "TS offset: %d, SeqNo offset: %d " |
| "on SSRC: %u timestamp: %u " |
| "from %s:%d\n", |
| text, seq, |
| state->patch.timestamp_offset, state->patch.seq_offset, |
| sstate->ssrc, timestamp, inet_ntoa(addr->sin_addr), |
| ntohs(addr->sin_port)); |
| } |
| return 0; |
| } |
| |
| tsdelta = |
| (int32_t)(timestamp - sstate->last_timestamp) / |
| (int16_t)(seq - sstate->last_seq); |
| |
| if (tsdelta == 0) { |
| /* Don't update *tsdelta_out */ |
| LOGPENDP(endp, DRTP, LOGL_NOTICE, |
| "The %s timestamp delta is %d " |
| "on SSRC: %u timestamp: %u " |
| "from %s:%d\n", |
| text, tsdelta, |
| sstate->ssrc, timestamp, |
| inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); |
| |
| return 0; |
| } |
| |
| if (sstate->last_tsdelta != tsdelta) { |
| if (sstate->last_tsdelta) { |
| LOGPENDP(endp, DRTP, LOGL_INFO, |
| "The %s timestamp delta changes from %d to %d " |
| "on SSRC: %u timestamp: %u from %s:%d\n", |
| text, sstate->last_tsdelta, tsdelta, |
| sstate->ssrc, timestamp, |
| inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); |
| } |
| } |
| |
| if (tsdelta_out) |
| *tsdelta_out = tsdelta; |
| |
| timestamp_error = |
| ts_alignment_error(sstate, state->packet_duration, timestamp); |
| |
| if (timestamp_error) { |
| rate_ctr_inc(sstate->err_ts_ctr); |
| LOGPENDP(endp, DRTP, LOGL_NOTICE, |
| "The %s timestamp has an alignment error of %d " |
| "on SSRC: %u " |
| "SeqNo delta: %d, TS delta: %d, dTS/dSeq: %d " |
| "from %s:%d. ptime: %d\n", |
| text, timestamp_error, |
| sstate->ssrc, |
| (int16_t)(seq - sstate->last_seq), |
| (int32_t)(timestamp - sstate->last_timestamp), |
| tsdelta, |
| inet_ntoa(addr->sin_addr), ntohs(addr->sin_port), |
| state->packet_duration); |
| } |
| return 1; |
| } |
| |
| /* Set the timestamp offset according to the packet duration. */ |
| static int adjust_rtp_timestamp_offset(struct mgcp_endpoint *endp, |
| struct mgcp_rtp_state *state, |
| struct mgcp_rtp_end *rtp_end, |
| struct sockaddr_in *addr, |
| int16_t delta_seq, uint32_t in_timestamp) |
| { |
| int32_t tsdelta = state->packet_duration; |
| int timestamp_offset; |
| uint32_t out_timestamp; |
| |
| if (tsdelta == 0) { |
| tsdelta = state->out_stream.last_tsdelta; |
| if (tsdelta != 0) { |
| LOGPENDP(endp, DRTP, LOGL_NOTICE, |
| "A fixed packet duration is not available, " |
| "using last output timestamp delta instead: %d " |
| "from %s:%d\n", tsdelta, |
| inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); |
| } else { |
| tsdelta = rtp_end->codec->rate * 20 / 1000; |
| LOGPENDP(endp, DRTP, LOGL_NOTICE, |
| "Fixed packet duration and last timestamp delta " |
| "are not available, " |
| "using fixed 20ms instead: %d " |
| "from %s:%d\n", tsdelta, |
| inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); |
| } |
| } |
| |
| out_timestamp = state->out_stream.last_timestamp + delta_seq * tsdelta; |
| timestamp_offset = out_timestamp - in_timestamp; |
| |
| if (state->patch.timestamp_offset != timestamp_offset) { |
| state->patch.timestamp_offset = timestamp_offset; |
| |
| LOGPENDP(endp, DRTP, LOGL_NOTICE, |
| "Timestamp offset change on SSRC: %u " |
| "SeqNo delta: %d, TS offset: %d, " |
| "from %s:%d\n", state->in_stream.ssrc, |
| delta_seq, state->patch.timestamp_offset, |
| inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); |
| } |
| |
| return timestamp_offset; |
| } |
| |
| /* Set the timestamp offset according to the packet duration. */ |
| static int align_rtp_timestamp_offset(struct mgcp_endpoint *endp, |
| struct mgcp_rtp_state *state, |
| struct mgcp_rtp_end *rtp_end, |
| struct sockaddr_in *addr, |
| uint32_t timestamp) |
| { |
| int ts_error = 0; |
| int ts_check = 0; |
| int ptime = state->packet_duration; |
| |
| /* Align according to: T + Toffs - Tlast = k * Tptime */ |
| |
| ts_error = ts_alignment_error(&state->out_stream, ptime, |
| timestamp + state->patch.timestamp_offset); |
| |
| /* If there is an alignment error, we have to compensate it */ |
| if (ts_error) { |
| state->patch.timestamp_offset += ptime - ts_error; |
| |
| LOGPENDP(endp, DRTP, LOGL_NOTICE, |
| "Corrected timestamp alignment error of %d on SSRC: %u " |
| "new TS offset: %d, " |
| "from %s:%d\n", |
| ts_error, state->in_stream.ssrc, |
| state->patch.timestamp_offset, inet_ntoa(addr->sin_addr), |
| ntohs(addr->sin_port)); |
| } |
| |
| /* Check we really managed to compensate the timestamp |
| * offset. There should not be any remaining error, failing |
| * here would point to a serous problem with the alignment |
| * error computation function */ |
| ts_check = ts_alignment_error(&state->out_stream, ptime, |
| timestamp + state->patch.timestamp_offset); |
| OSMO_ASSERT(ts_check == 0); |
| |
| /* Return alignment error before compensation */ |
| return ts_error; |
| } |
| |
| /*! dummy callback to disable transcoding (see also cfg->rtp_processing_cb). |
| * \param[in] associated endpoint. |
| * \param[in] destination RTP end. |
| * \param[in,out] pointer to buffer with voice data. |
| * \param[in] voice data length. |
| * \param[in] maximum size of caller provided voice data buffer. |
| * \returns ignores input parameters, return always 0. */ |
| int mgcp_rtp_processing_default(struct mgcp_endpoint *endp, |
| struct mgcp_rtp_end *dst_end, |
| char *data, int *len, int buf_size) |
| { |
| LOGPENDP(endp, DRTP, LOGL_DEBUG, "transcoding disabled\n"); |
| return 0; |
| } |
| |
| /*! dummy callback to disable transcoding (see also cfg->setup_rtp_processing_cb). |
| * \param[in] associated endpoint. |
| * \param[in] destination RTP connnection. |
| * \param[in] source RTP connection. |
| * \returns ignores input parameters, return always 0. */ |
| int mgcp_setup_rtp_processing_default(struct mgcp_endpoint *endp, |
| struct mgcp_conn_rtp *conn_dst, |
| 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 sockaddr_in *addr, struct msgb *msg) |
| { |
| 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, |
| inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); |
| 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, |
| inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); |
| } |
| } 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, |
| 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; |
| |
| 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, |
| 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 |
| 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"); |
| } |
| |
| /*! 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, |
| 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 *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"); |
| } |
| |
| LOGPENDP(endp, DRTP, LOGL_DEBUG, "loop:%d, mode:%d%s\n", |
| trunk->audio_loop, conn_src->conn->mode, |
| conn_src->conn->mode == MGCP_CONN_LOOPBACK ? " (loopback)" : ""); |
| |
| /* 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; |
| |
| 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, |
| 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 = msgb_length(msg); |
| 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, 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, |
| 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); |
| |
| 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, 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, (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); |
| |
| 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 sockaddr_in *addr) |
| { |
| if (conn->end.addr.s_addr == 0) { |
| 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", inet_ntoa(addr->sin_addr)); |
| 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", inet_ntoa(addr->sin_addr)); |
| 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.s_addr != addr->sin_addr.s_addr) { |
| LOGPCONN(conn->conn, DRTP, LOGL_ERROR, |
| "data from wrong address: %s, ", inet_ntoa(addr->sin_addr)); |
| LOGPC(DRTP, LOGL_ERROR, "expected: %s\n", |
| inet_ntoa(conn->end.addr)); |
| 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 (conn->end.rtp_port != addr->sin_port && |
| conn->end.rtcp_port != addr->sin_port) { |
| LOGPCONN(conn->conn, DRTP, LOGL_ERROR, |
| "data from wrong source port: %d, ", 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)); |
| 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) |
| { |
| /* 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) { |
| LOGPCONN(conn->conn, DRTP, LOGL_DEBUG, |
| "destination IP-address and rtp port is (not yet) known (%s:%u)\n", |
| inet_ntoa(conn->end.addr), conn->end.rtp_port); |
| return -1; |
| } |
| |
| if (strcmp(inet_ntoa(conn->end.addr), "0.0.0.0") == 0) { |
| LOGPCONN(conn->conn, DRTP, LOGL_ERROR, |
| "destination IP-address is invalid (%s:%u)\n", |
| inet_ntoa(conn->end.addr), 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", |
| inet_ntoa(conn->end.addr), 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 sockaddr_in *from_addr = mc->from_addr; |
| 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, |
| 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 sockaddr_in *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) { |
| conn->u.rtp.end.addr = from_addr->sin_addr; |
| conn->u.rtp.end.rtp_port = from_addr->sin_port; |
| } |
| 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; |
| |
| /* FIXME: integrate E1 support from libsomoabis, also implement |
| * handling for RTCP packets, which can not converted to E1. */ |
| LOGPCONN(conn, DRTP, LOGL_FATAL, |
| "cannot dispatch! E1 support is not implemented yet!\n"); |
| return -1; |
| } |
| |
| /*! 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) { |
| 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) |
| { |
| LOGPCONN(conn, DRTP, LOGL_FATAL, |
| "cannot dispatch! E1 support is not implemented yet!\n"); |
| } |
| |
| 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 sockaddr_in addr; |
| socklen_t slen = sizeof(addr); |
| 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, &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), inet_ntoa(addr.sin_addr), ntohs(addr.sin_port)); |
| |
| 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, 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 sockaddr_in *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_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, 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); |
| |
| rtp_end->rtp.when = OSMO_FD_READ; |
| 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; |
| } |
| |
| rtp_end->rtcp.when = OSMO_FD_READ; |
| 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; |
| 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) { |
| 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; |
| 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, 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); |
| } |
| } |