/* A Media Gateway Control Protocol Media Gateway: RFC 3435 */ /* The protocol implementation */ /* * (C) 2009-2012 by Holger Hans Peter Freyther * (C) 2009-2012 by On-Waves * (C) 2013-2024 by sysmocom - s.f.m.c. GmbH * 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 . * */ #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #define RTP_SEQ_MOD (1 << 16) #define RTP_MAX_DROPOUT 3000 #define RTP_MAX_MISORDER 100 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->ctrg; 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(rate_ctr_group_get_ctr(conn_stats, id), inc); rate_ctr_add(rate_ctr_group_get_ctr(mgw_stats, id), inc); } 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); } /* wrapper around libosmocore msgb_copy_c, which [at least before libosmocore.git Change-Id * I68328adb952ca8833ba047cb3b49ccc6f8a1f1b5] doesn't copy the cb */ static inline struct msgb *mgw_msgb_copy_c(void *ctx, struct msgb *msg, const char *name) { struct msgb *msg2 = msgb_copy_c(ctx, msg, name); if (OSMO_UNLIKELY(!msg2)) return NULL; memcpy(msg2->cb, msg->cb, sizeof(msg2->cb)); return msg2; } 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. * \ returns 0 on success, -1 if no local address could be provided. * * 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. */ int mgcp_get_local_addr(char *addr, struct mgcp_conn_rtp *conn) { const struct mgcp_endpoint *endp = conn->conn->endp; const struct mgcp_config *cfg = endp->trunk->cfg; char ipbuf[INET6_ADDRSTRLEN]; int rc; bool rem_addr_set = osmo_sockaddr_is_any(&conn->end.addr) == 0; const char *bind_addr; /* Osmux: No smart IP addresses allocation is supported yet. Simply * return the one set in VTY config: */ if (mgcp_conn_rtp_is_osmux(conn)) { if (rem_addr_set) { /* Match IP version with what was requested from remote: */ bind_addr = conn->end.addr.u.sa.sa_family == AF_INET6 ? cfg->osmux.local_addr_v6 : cfg->osmux.local_addr_v4; } else { /* Choose any of the bind addresses, preferring v6 over v4 if available: */ bind_addr = cfg->osmux.local_addr_v6; if (!bind_addr) bind_addr = cfg->osmux.local_addr_v4; } if (!bind_addr) { LOGPCONN(conn->conn, DOSMUX, LOGL_ERROR, "Unable to locate local Osmux address, check your configuration! v4=%u v6=%u remote_known=%s\n", !!cfg->osmux.local_addr_v4, !!cfg->osmux.local_addr_v6, rem_addr_set ? osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf) : "no"); return -1; } LOGPCONN(conn->conn, DOSMUX, LOGL_DEBUG, "Using configured osmux bind ip as local bind ip %s\n", bind_addr); osmo_strlcpy(addr, bind_addr, INET6_ADDRSTRLEN); return 0; } /* Try probing the local IP-Address */ if (cfg->net_ports.bind_addr_probe && rem_addr_set) { rc = osmo_sock_local_ip(addr, osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf)); 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, osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf)); return 0; } } /* Select from preconfigured IP-Addresses. */ if (rem_addr_set) { /* Check there is a bind IP for the RTP traffic configured, * if so, use that IP-Address */ bind_addr = conn->end.addr.u.sa.sa_family == AF_INET6 ? cfg->net_ports.bind_addr_v6 : cfg->net_ports.bind_addr_v4; } else { /* Choose any of the bind addresses, preferring v6 over v4 */ bind_addr = cfg->net_ports.bind_addr_v6; if (!strlen(bind_addr)) bind_addr = cfg->net_ports.bind_addr_v4; } if (strlen(bind_addr)) { LOGPCONN(conn->conn, DRTP, LOGL_DEBUG, "using configured rtp bind ip as local bind ip %s\n", bind_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 */ bind_addr = cfg->source_addr; LOGPCONN(conn->conn, DRTP, LOGL_DEBUG, "using mgcp bind ip as local rtp bind ip: %s\n", bind_addr); } osmo_strlcpy(addr, bind_addr, INET6_ADDRSTRLEN); return 0; } /* This does not need to be a precision timestamp and * is allowed to wrap quite fast. The returned value is * 1/codec_rate seconds. */ uint32_t mgcp_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; } /* Compute timestamp alignment error */ static int32_t ts_alignment_error(const 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(const struct mgcp_endpoint *endp, const struct mgcp_rtp_state *state, const struct mgcp_rtp_stream_state *sstate, const struct mgcp_rtp_end *rtp_end, const struct osmo_sockaddr *addr, uint16_t seq, uint32_t timestamp, const char *text, int32_t * tsdelta_out) { int32_t tsdelta; int32_t timestamp_error; char ipbuf[INET6_ADDRSTRLEN]; /* 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, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); } 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, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); 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, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); } } 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, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa), state->packet_duration); } return 1; } /* Set the timestamp offset according to the packet duration. */ static int adjust_rtp_timestamp_offset(const struct mgcp_endpoint *endp, struct mgcp_rtp_state *state, const struct mgcp_rtp_end *rtp_end, const struct osmo_sockaddr *addr, int16_t delta_seq, uint32_t in_timestamp, bool marker_bit) { int32_t tsdelta = state->packet_duration; int timestamp_offset; uint32_t out_timestamp; char ipbuf[INET6_ADDRSTRLEN]; if (marker_bit) { /* If RTP pkt contains marker bit, the timestamps are not longer * in sync, so we can erase timestamp offset patching. */ state->patch.timestamp_offset = 0; return 0; } 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, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); } else { tsdelta = rtp_end->cset.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, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); } } 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, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); } return timestamp_offset; } /* Set the timestamp offset according to the packet duration. */ static int align_rtp_timestamp_offset(const struct mgcp_endpoint *endp, struct mgcp_rtp_state *state, const struct mgcp_rtp_end *rtp_end, const struct osmo_sockaddr *addr, uint32_t timestamp, bool marker_bit) { char ipbuf[INET6_ADDRSTRLEN]; int ts_error = 0; int ts_check = 0; int ptime = state->packet_duration; if (marker_bit) { /* If RTP pkt contains marker bit, the timestamps are not longer * in sync, so no alignment is needed. */ return 0; } /* 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, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); } /* 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; } void mgcp_rtp_annex_count(const struct mgcp_endpoint *endp, struct mgcp_rtp_state *state, const uint16_t seq, const int32_t transit, const uint32_t ssrc, const bool marker_bit) { int32_t d; /* initialize or re-initialize */ if (!state->stats.initialized || state->stats.ssrc != ssrc || marker_bit) { 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 * of the codec that is set for the destination connection (conn_dst) */ static int mgcp_patch_pt(struct mgcp_conn_rtp *conn_dst, struct msgb *msg) { struct rtp_hdr *rtp_hdr; if (msgb_length(msg) < sizeof(struct rtp_hdr)) { LOG_CONN_RTP(conn_dst, LOGL_NOTICE, "RTP packet too short (%u < %zu)\n", msgb_length(msg), sizeof(struct rtp_hdr)); return -EINVAL; } rtp_hdr = (struct rtp_hdr *)msgb_data(msg); if (!conn_dst->end.cset.codec) { LOG_CONN_RTP(conn_dst, LOGL_NOTICE, "no codec set on destination connection!\n"); return -EINVAL; } rtp_hdr->payload_type = (uint8_t) conn_dst->end.cset.codec->payload_type; 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(const struct mgcp_endpoint *endp, struct mgcp_rtp_state *state, struct mgcp_rtp_end *rtp_end, struct osmo_sockaddr *addr, struct msgb *msg) { char ipbuf[INET6_ADDRSTRLEN]; uint32_t arrival_time; int32_t transit; uint16_t seq; uint32_t timestamp, ssrc; bool marker_bit; struct rtp_hdr *rtp_hdr; struct mgcp_rtp_codec *codec = rtp_end->cset.codec; int payload = 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 = mgcp_get_current_ts(codec->rate); ssrc = ntohl(rtp_hdr->ssrc); marker_bit = !!rtp_hdr->marker; transit = arrival_time - timestamp; mgcp_rtp_annex_count(endp, state, seq, transit, ssrc, marker_bit); if (!state->initialized) { state->initialized = 1; state->packet_duration = mgcp_rtp_packet_duration(endp, rtp_end); state->in_stream.last_seq = seq - 1; state->in_stream.ssrc = ssrc; state->in_stream.last_tsdelta = 0; state->out_stream.last_seq = seq - 1; state->out_stream.last_tsdelta = 0; state->out_stream.last_timestamp = timestamp; state->out_stream.ssrc = ssrc - 1; /* force output SSRC change */ state->patch.orig_ssrc = ssrc; state->patch.patch_ssrc = rtp_end->force_constant_ssrc; LOGPENDP(endp, DRTP, LOGL_INFO, "initializing stream, SSRC: %u timestamp: %u " "pkt-duration: %d, from %s:%d\n", state->in_stream.ssrc, state->patch.seq_offset, state->packet_duration, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); if (state->packet_duration == 0) { state->packet_duration = codec->rate * 20 / 1000; LOGPENDP(endp, DRTP, LOGL_NOTICE, "fixed packet duration is not available, " "using fixed 20ms instead: %d from %s:%d\n", state->packet_duration, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); } } else if (state->in_stream.ssrc != ssrc) { LOGPENDP(endp, DRTP, LOGL_NOTICE, "SSRC changed: %u -> %u " "from %s:%d\n", state->in_stream.ssrc, rtp_hdr->ssrc, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); state->in_stream.ssrc = ssrc; if (state->patch.patch_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, marker_bit); ssrc = state->patch.orig_ssrc; LOGPENDP(endp, DRTP, LOGL_NOTICE, "SSRC patching enabled, SSRC: %u " "SeqNo offset: %d, TS offset: %d " "from %s:%d\n", state->in_stream.ssrc, state->patch.seq_offset, state->patch.timestamp_offset, osmo_sockaddr_ntop(&addr->u.sa, ipbuf), osmo_sockaddr_port(&addr->u.sa)); } state->in_stream.last_tsdelta = 0; } else { if (!marker_bit) { /* 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); } else { state->in_stream.last_tsdelta = 0; } 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, marker_bit); /* 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 (!marker_bit) { 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); } else { state->out_stream.last_tsdelta = 0; } /* 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, "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; } /*! Convert msg to AMR RTP framing mode specified by target_is_oa. * \param[in] endp MGCP Endpoint where this message belongs to (used for logging purposes) * \param[in] msg Message containing an AMR RTP payload (in octet-aligned or bandwidth-efficient format). * \param[in] target_is_oa the target framing mode that msg will contain after this function succeeds. * \returns The size of the new RTP AMR content on success, negative on error. * * For AMR RTP two framing modes are defined RFC3267. There is a bandwidth * 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. */ 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; bool orig_is_oa; 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); orig_is_oa = osmo_amr_is_oa(rtp_hdr->data, payload_len); if (orig_is_oa) { 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, "RTP AMR packet conversion %s->%s failed: %s\n", orig_is_oa ? "OA" : "BWE", target_is_oa ? "OA" : "BWE", osmo_hexdump(rtp_hdr->data, payload_len)); return -EINVAL; } return msgb_trim(msg, rc + sizeof(struct rtp_hdr)); } /* 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 */ void forward_data_tap(struct osmo_io_fd *iofd, struct mgcp_rtp_tap *tap, struct msgb *msg) { int rc; if (!tap->enabled) return; struct msgb *msg2 = msgb_copy(msg, "RTP TAP Tx"); if (!msg2) return; rc = osmo_iofd_sendto_msgb(iofd, msg2, 0, &tap->forward); if (rc < 0) LOGP(DRTP, LOGL_ERROR, "Forwarding tapped (debug) voice data failed.\n"); } /* Generate an RTP header if it is missing */ static void gen_rtp_header(struct msgb *msg, struct mgcp_rtp_end *rtp_end, struct mgcp_rtp_state *state) { struct rtp_hdr *hdr = (struct rtp_hdr *)msgb_data(msg); if (hdr->version > 0) return; hdr->version = 2; hdr->payload_type = rtp_end->cset.codec->payload_type; hdr->timestamp = osmo_htonl(mgcp_get_current_ts(rtp_end->cset.codec->rate)); hdr->sequence = osmo_htons(state->alt_rtp_tx_sequence); hdr->ssrc = state->alt_rtp_tx_ssrc; } /* Check if the origin (addr) matches the address/port data of the RTP * connections. */ static int check_rtp_origin(struct mgcp_conn_rtp *conn, struct osmo_sockaddr *addr) { char ipbuf[INET6_ADDRSTRLEN]; if (osmo_sockaddr_is_any(&conn->end.addr) != 0 || osmo_sockaddr_port(&conn->end.addr.u.sa) == 0) { if (mgcp_conn_rtp_is_iuup(conn) && !conn->iuup.configured) { /* Allow IuUP Initialization to get through even if we don't have a remote address set yet. * This is needed because hNodeB doesn't announce its IuUP remote IP addr to the MGCP client * (RAB Assignment Response at HNBGW) until it has gone through IuUP Initialization against * this MGW here. Hence the MGW may not yet know the remote IuUP address and port at the time * of receiving IuUP Initialization from the hNodeB. */ LOGPCONN(conn->conn, DRTP, LOGL_INFO, "Rx RTP from %s: allowing unknown src for IuUP Initialization\n", osmo_sockaddr_to_str(addr)); return 0; } /* Receiving early media before the endpoint is configured. Instead of logging * this as an error that occurs on every call, keep it more low profile to not * confuse humans with expected errors. */ LOGPCONN(conn->conn, DRTP, LOGL_INFO, "Rx RTP from %s, but remote address not set: dropping early media\n", osmo_sockaddr_to_str(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.u.sa.sa_family != addr->u.sa.sa_family || (conn->end.addr.u.sa.sa_family == AF_INET && conn->end.addr.u.sin.sin_addr.s_addr != addr->u.sin.sin_addr.s_addr) || (conn->end.addr.u.sa.sa_family == AF_INET6 && memcmp(&conn->end.addr.u.sin6.sin6_addr, &addr->u.sin6.sin6_addr, sizeof(struct in6_addr)))) { LOGPCONN(conn->conn, DRTP, LOGL_ERROR, "data from wrong src %s, expected IP Address %s. Packet tossed.\n", osmo_sockaddr_to_str(addr), osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf)); 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 (osmo_sockaddr_port(&conn->end.addr.u.sa) != osmo_sockaddr_port(&addr->u.sa) && ntohs(conn->end.rtcp_port) != osmo_sockaddr_port(&addr->u.sa)) { LOGPCONN(conn->conn, DRTP, LOGL_ERROR, "data from wrong src %s, expected port: %u for RTP or %u for RTCP. Packet tossed.\n", osmo_sockaddr_to_str(addr), osmo_sockaddr_port(&conn->end.addr.u.sa), ntohs(conn->end.rtcp_port)); return -1; } return 0; } /* Check the if the destination address configuration of an RTP connection * makes sense */ static int check_rtp_destin(struct mgcp_conn_rtp *conn) { char ipbuf[INET6_ADDRSTRLEN]; bool ip_is_any = osmo_sockaddr_is_any(&conn->end.addr) != 0; uint16_t port = osmo_sockaddr_port(&conn->end.addr.u.sa); /* Note: it is legal to create a connection but never setting a port * and IP-address for outgoing data. */ if (ip_is_any && port == 0) { LOGPCONN(conn->conn, DRTP, LOGL_DEBUG, "destination IP-address and rtp port is not (yet) known (%s:%u)\n", osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf), port); return -1; } if (ip_is_any) { LOGPCONN(conn->conn, DRTP, LOGL_ERROR, "destination IP-address is invalid (%s:%u)\n", osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf), port); return -1; } if (port == 0) { LOGPCONN(conn->conn, DRTP, LOGL_ERROR, "destination rtp port is invalid (%s:%u)\n", osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf), 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; } /*! Dispatch msg bridged from the sister conn in the endpoint. Takes ownership of msgb. * \param[in] conn_dst The destination conn that should handle and transmit the content to * its peer outside MGW. * \param[in] msg msgb containing an RTP pkt received by the sister conn in the endpoint, * \returns bytes sent, -1 on error. * * Possible options are standard RTP packet transmission, transmission * via IuUP or transmission via an osmux connection. */ static int mgcp_conn_rtp_dispatch_rtp(struct mgcp_conn_rtp *conn_dst, struct msgb *msg) { struct osmo_rtp_msg_ctx *mc = OSMO_RTP_MSG_CTX(msg); enum rtp_proto proto = mc->proto; struct mgcp_conn_rtp *conn_src = mc->conn_src; struct mgcp_endpoint *endp = conn_src->conn->endp; LOGPENDP(endp, DRTP, LOGL_DEBUG, "destin conn:%s\n", mgcp_conn_dump(conn_dst->conn)); /* Before we try to deliver the packet, we check if the destination * port and IP-Address make sense at all. If not, we will be unable * to deliver the packet. */ if (check_rtp_destin(conn_dst) != 0) { msgb_free(msg); return -1; } /* Depending on the RTP connection type, deliver the RTP packet to the * destination connection. */ switch (conn_dst->type) { case MGCP_RTP_DEFAULT: LOGPENDP(endp, DRTP, LOGL_DEBUG, "endpoint type is MGCP_RTP_DEFAULT, " "using mgcp_send() to forward data directly\n"); return mgcp_send(endp, proto == MGCP_PROTO_RTP, mc->from_addr, msg, conn_src, conn_dst); case MGCP_RTP_OSMUX: LOGPENDP(endp, DRTP, LOGL_DEBUG, "endpoint type is MGCP_RTP_OSMUX, " "using osmux_xfrm_to_osmux() to forward data through OSMUX\n"); return conn_osmux_send_rtp(conn_dst, msg); case MGCP_RTP_IUUP: if (proto == MGCP_PROTO_RTP) { LOGPENDP(endp, DRTP, LOGL_DEBUG, "endpoint type is MGCP_RTP_IUUP, " "using mgcp_conn_iuup_send_rtp() to forward data over IuUP\n"); return mgcp_conn_iuup_send_rtp(conn_src, conn_dst, msg); } /* RTCP: we forward as usual for regular RTP connection */ LOGPENDP(endp, DRTP, LOGL_DEBUG, "endpoint type is MGCP_RTP_IUUP and proto!=MGCP_PROTO_RTP, " "using mgcp_send() to forward data directly\n"); return mgcp_send(endp, false, mc->from_addr, msg, conn_src, 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"); msgb_free(msg); return -1; } /*! send message buffer via udp socket. If it succeeds, it takes ownership of the msgb and internally calls * msgb_free() after the aynchronous sendto() completes. In case of error, the msgb is still owned by the * caller and must be free'd accordingly. * \param[in] iofd associated file descriptor. * \param[in] addr destination ip-address. * \param[in] msg message buffer that holds the data to be send. * \returns 0 in case of success (takes msgb ownership), -1 on error (doesn't take msgb ownership). */ static int mgcp_udp_send_msg(struct osmo_io_fd *iofd, const struct osmo_sockaddr *addr, struct msgb *msg) { LOGP(DRTP, LOGL_DEBUG, "sending %d bytes length packet to %s ...\n", msgb_length(msg), osmo_sockaddr_to_str(addr)); return osmo_iofd_sendto_msgb(iofd, msg, 0, addr); } /*! send udp packet from raw buffer/length. * \param[in] iofd associated file descriptor. * \param[in] addr destination ip-address. * \param[in] buf buffer that holds the data to be send. * \param[in] len length of the data to be sent. * \returns 0 in case of success, -1 on error. */ int mgcp_udp_send(struct osmo_io_fd *iofd, const struct osmo_sockaddr *addr, const char *buf, int len) { struct msgb *msg = msgb_alloc_c(iofd, len, "mgcp_udp_send"); if (!msg) return -ENOMEM; memcpy(msg->tail, buf, len); msgb_put(msg, len); return mgcp_udp_send_msg(iofd, addr, msg); } /*! 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 0 in case of success, -1 on error. */ int mgcp_send_dummy(struct mgcp_endpoint *endp, struct mgcp_conn_rtp *conn) { int rc; int was_rtcp = 0; struct osmo_sockaddr rtcp_addr; OSMO_ASSERT(endp); OSMO_ASSERT(conn); LOGPCONN(conn->conn, DRTP, LOGL_DEBUG, "sending dummy packet... %s\n", mgcp_conn_dump(conn->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) != 0) goto failed; if (mgcp_conn_rtp_is_iuup(conn)) rc = mgcp_conn_iuup_send_dummy(conn); else rc = mgcp_udp_send(conn->end.rtp, &conn->end.addr, rtp_dummy_payload, sizeof(rtp_dummy_payload)); if (rc == -1) goto failed; if (endp->trunk->omit_rtcp) return rc; was_rtcp = 1; rtcp_addr = conn->end.addr; osmo_sockaddr_set_port(&rtcp_addr.u.sa, ntohs(conn->end.rtcp_port)); rc = mgcp_udp_send(conn->end.rtcp, &rtcp_addr, rtp_dummy_payload, sizeof(rtp_dummy_payload)); 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; } /*! Send RTP/RTCP data to a specified destination connection. Takes ownership of msg. * \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] addr spoofed source address (set to NULL to disable). * \param[in] msg message 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, negative on ERROR. */ int mgcp_send(struct mgcp_endpoint *endp, int is_rtp, struct osmo_sockaddr *addr, struct msgb *msg, struct mgcp_conn_rtp *conn_src, struct mgcp_conn_rtp *conn_dst) { /*! When no destination connection is available (e.g. when only one * connection in loopback mode exists), then the source connection * shall be specified as destination connection */ struct mgcp_trunk *trunk = endp->trunk; struct mgcp_rtp_end *rtp_end; struct mgcp_rtp_state *rtp_state; char ipbuf[INET6_ADDRSTRLEN]; char *dest_name; int rc; int len; OSMO_ASSERT(conn_src); OSMO_ASSERT(conn_dst); if (is_rtp) LOGPENDP(endp, DRTP, LOGL_DEBUG, "delivering RTP packet...\n"); else LOGPENDP(endp, DRTP, LOGL_DEBUG, "delivering RTCP packet...\n"); /* Patch the payload type number: translate from conn_src to conn_dst. * Do not patch for IuUP, where the correct payload type number is already set in bridge_iuup_to_rtp_peer(): * IuUP -> AMR: calls this function, skip patching if conn_src is IuUP. * {AMR or IuUP} -> IuUP: calls mgcp_udp_send() directly, skipping this function: No need to examine dst. */ if (is_rtp && !mgcp_conn_rtp_is_iuup(conn_src)) { if (mgcp_patch_pt(conn_dst, msg) < 0) { LOGPENDP(endp, DRTP, LOGL_NOTICE, "unable to patch payload type RTP packet, discarding...\n"); msgb_free(msg); return -EINVAL; } } /* Note: In case of loopback configuration, both, the source and the * destination will point to the same connection. */ rtp_end = &conn_dst->end; rtp_state = &conn_src->state; dest_name = conn_dst->conn->name; /* Ensure we have an alternative SSRC in case we need it, see also * gen_rtp_header() */ if (rtp_state->alt_rtp_tx_ssrc == 0) rtp_state->alt_rtp_tx_ssrc = rand(); if (!rtp_end->output_enabled) { rtpconn_rate_ctr_inc(conn_dst, endp, RTP_DROPPED_PACKETS_CTR); LOGPENDP(endp, DRTP, LOGL_DEBUG, "output disabled, drop to %s %s " "rtp_port:%u rtcp_port:%u\n", dest_name, osmo_sockaddr_ntop(&rtp_end->addr.u.sa, ipbuf), osmo_sockaddr_port(&rtp_end->addr.u.sa), ntohs(rtp_end->rtcp_port) ); } else if (is_rtp) { struct mgcp_rtp_codec *src_codec; struct mgcp_rtp_codec *dst_codec; /* Make sure we have a valid RTP header, in cases where no RTP * header is present, we will generate one. */ gen_rtp_header(msg, rtp_end, rtp_state); /* TODO: Run transcoder */ if (addr) mgcp_patch_and_count(endp, rtp_state, rtp_end, addr, msg); src_codec = conn_src->end.cset.codec; dst_codec = conn_dst->end.cset.codec; if (mgcp_conn_rtp_is_iuup(conn_dst) || mgcp_conn_rtp_is_iuup(conn_src)) { /* the iuup code will correctly transform to the correct AMR mode */ } else if (mgcp_codec_amr_align_mode_is_indicated(dst_codec)) { rc = amr_oa_bwe_convert(endp, msg, dst_codec->param.amr_octet_aligned); if (rc < 0) { LOGPENDP(endp, DRTP, LOGL_ERROR, "Error in AMR octet-aligned <-> bandwidth-efficient mode conversion (target=%s)\n", dst_codec->param.amr_octet_aligned ? "octet-aligned" : "bandwidth-efficient"); msgb_free(msg); return rc; } } else if (rtp_end->rfc5993_hr_convert && strcmp(src_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"); msgb_free(msg); return rc; } } LOGPENDP(endp, DRTP, LOGL_DEBUG, "process/send to %s %s " "rtp_port:%u rtcp_port:%u\n", dest_name, osmo_sockaddr_ntop(&rtp_end->addr.u.sa, ipbuf), osmo_sockaddr_port(&rtp_end->addr.u.sa), ntohs(rtp_end->rtcp_port) ); /* Forward a copy of the RTP data to a debug ip/port */ forward_data_tap(rtp_end->rtp, &conn_src->tap_out, msg); len = msgb_length(msg); rc = mgcp_udp_send_msg(rtp_end->rtp, &rtp_end->addr, msg); if (rc < 0) { msgb_free(msg); return rc; } rtpconn_rate_ctr_inc(conn_dst, endp, RTP_PACKETS_TX_CTR); rtpconn_rate_ctr_add(conn_dst, endp, RTP_OCTETS_TX_CTR, len); rtp_state->alt_rtp_tx_sequence++; return 0; } else if (!trunk->omit_rtcp) { struct osmo_sockaddr rtcp_addr = rtp_end->addr; osmo_sockaddr_set_port(&rtcp_addr.u.sa, ntohs(rtp_end->rtcp_port)); LOGPENDP(endp, DRTP, LOGL_DEBUG, "send to %s %s rtp_port:%u rtcp_port:%u\n", dest_name, osmo_sockaddr_ntop(&rtcp_addr.u.sa, ipbuf), osmo_sockaddr_port(&rtp_end->addr.u.sa), osmo_sockaddr_port(&rtcp_addr.u.sa) ); len = msgb_length(msg); rc = mgcp_udp_send_msg(rtp_end->rtcp, &rtcp_addr, msg); if (rc < 0) { msgb_free(msg); return rc; } rtpconn_rate_ctr_inc(conn_dst, endp, RTP_PACKETS_TX_CTR); rtpconn_rate_ctr_add(conn_dst, endp, RTP_OCTETS_TX_CTR, len); rtp_state->alt_rtp_tx_sequence++; return 0; } msgb_free(msg); return 0; } /*! determine if there's only a single recipient in endp for data received via conn_src. * The function returns NULL in case there is no recipient, or in case there are multiple recipients. * \param endp The MGCP endpoint whose connections to analyze * \param conn_src The source MGCP connection [which shall not count in results] * \returns recipient donnection if there is only one; NULL in case there are multiple */ static struct mgcp_conn *rtpbridge_get_only_recipient(struct mgcp_endpoint *endp, struct mgcp_conn *conn_src) { struct mgcp_conn *conn_ret = NULL; struct mgcp_conn *conn_dst; llist_for_each_entry(conn_dst, &endp->conns, entry) { if (conn_dst == conn_src) continue; switch (conn_dst->mode) { case MGCP_CONN_SEND_ONLY: case MGCP_CONN_RECV_SEND: case MGCP_CONN_CONFECHO: if (conn_ret) return NULL; conn_ret = conn_dst; break; default: break; } } return conn_ret; } /*! Dispatch incoming RTP packet to opposite RTP connection. * \param[in] msg Message buffer to bridge, coming from source connection. * msg shall contain "struct osmo_rtp_msg_ctx *" attached in * "OSMO_RTP_MSG_CTX(msg)". * \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 mgcp_endpoint *endp = conn->endp; struct osmo_sockaddr *from_addr = mc->from_addr; char ipbuf[INET6_ADDRSTRLEN]; int rc = 0; /*! 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. */ /* If source is IuUP, we need to handle state, forward it through specific bridge path: */ if (mgcp_conn_rtp_is_iuup(conn_src) && mc->proto == MGCP_PROTO_RTP) return mgcp_conn_iuup_dispatch_rtp(msg); /* 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 (osmo_sockaddr_port(&conn_src->end.addr.u.sa) == 0) { memcpy(&conn_src->end.addr, from_addr, sizeof(conn_src->end.addr)); LOG_CONN_RTP(conn_src, LOGL_NOTICE, "loopback mode: implicitly using source address (%s:%u) as destination address\n", osmo_sockaddr_ntop(&from_addr->u.sa, ipbuf), osmo_sockaddr_port(&conn_src->end.addr.u.sa)); } return mgcp_conn_rtp_dispatch_rtp(conn_src, msg); } /* If the mode does not allow receiving RTP, we are done. */ switch (conn->mode) { case MGCP_CONN_RECV_ONLY: case MGCP_CONN_RECV_SEND: case MGCP_CONN_CONFECHO: break; default: return rc; } /* All the use cases above are 1:1 where we have one source msgb and we're sending that to one * destination. msgb ownership had been passed to the respective _*dospatch_rtp() function. * In the cases below, we actually [can] have multiple recipients, so we copy the original msgb * for each of the recipients. */ /* If the mode is "confecho", send RTP back to the sender. */ if (conn->mode == MGCP_CONN_CONFECHO) { struct msgb *msg2 = mgw_msgb_copy_c(conn, msg, "RTP confecho"); if (OSMO_LIKELY(msg2)) rc = mgcp_conn_rtp_dispatch_rtp(conn_src, msg2); } conn_dst = rtpbridge_get_only_recipient(endp, conn); if (OSMO_LIKELY(conn_dst)) { /* we only have a single recipient and cann hence send the original msgb without copying */ rc = mgcp_conn_rtp_dispatch_rtp(&conn_dst->u.rtp, msg); } else { /* Dispatch RTP packet to all other connection(s) that send audio. */ llist_for_each_entry(conn_dst, &endp->conns, entry) { struct msgb *msg2; if (conn_dst == conn) continue; switch (conn_dst->mode) { case MGCP_CONN_SEND_ONLY: case MGCP_CONN_RECV_SEND: case MGCP_CONN_CONFECHO: /* we have multiple recipients and must make copies for each recipient */ msg2 = mgw_msgb_copy_c(conn_dst, msg, "RTP Tx copy"); if (OSMO_LIKELY(msg2)) rc = mgcp_conn_rtp_dispatch_rtp(&conn_dst->u.rtp, msg2); break; default: break; } } /* as we only sent copies in the previous llist_for_each_entry() loop, we must free the * original one */ msgb_free(msg); } return rc; } /*! dispatch incoming RTP packet to E1 subslot, handle RTCP packets locally. * \param[in] proto protocol (MGCP_CONN_TYPE_RTP or MGCP_CONN_TYPE_RTCP). * \param[in] addr socket address where the RTP packet has been received from. * \param[in] buf buffer that hold the RTP payload. * \param[in] buf_size size data length of buf. * \param[in] conn originating connection. * \returns 0 on success, -1 on ERROR. */ int mgcp_dispatch_e1_bridge_cb(struct msgb *msg) { struct osmo_rtp_msg_ctx *mc = OSMO_RTP_MSG_CTX(msg); struct mgcp_conn_rtp *conn_src = mc->conn_src; struct mgcp_conn *conn = conn_src->conn; struct osmo_sockaddr *from_addr = mc->from_addr; char ipbuf[INET6_ADDRSTRLEN]; /* 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 (osmo_sockaddr_port(&conn_src->end.addr.u.sa) == 0) { memcpy(&conn_src->end.addr, from_addr, sizeof(conn_src->end.addr)); LOG_CONN_RTP(conn_src, LOGL_NOTICE, "loopback mode: implicitly using source address (%s:%u) as destination address\n", osmo_sockaddr_ntop(&from_addr->u.sa, ipbuf), osmo_sockaddr_port(&conn_src->end.addr.u.sa)); } return mgcp_conn_rtp_dispatch_rtp(conn_src, msg); } /* Forward to E1 */ return mgcp_e1_send_rtp(conn->endp, conn_src->end.cset.codec, msg); } /*! cleanup an endpoint when a connection on an RTP bridge endpoint is removed. * \param[in] endp Endpoint on which the connection resides. * \param[in] conn Connection that is about to be removed (ignored). */ void mgcp_cleanup_rtp_bridge_cb(struct mgcp_endpoint *endp, struct mgcp_conn *conn) { struct mgcp_conn *conn_cleanup; /* In mgcp_dispatch_rtp_bridge_cb() we use conn->priv to cache the * pointer to the destination connection, so that we do not have * to go through the list every time an RTP packet arrives. To prevent * a use-after-free situation we invalidate this information for all * connections present when one connection is removed from the * endpoint. */ llist_for_each_entry(conn_cleanup, &endp->conns, entry) { if (conn_cleanup->priv == conn) conn_cleanup->priv = NULL; } } /*! cleanup an endpoint when a connection on an E1 endpoint is removed. * \param[in] endp Endpoint on which the connection resides. * \param[in] conn Connection that is about to be removed (ignored). */ void mgcp_cleanup_e1_bridge_cb(struct mgcp_endpoint *endp, struct mgcp_conn *conn) { /* Cleanup tasks for E1 are the same as for regular endpoint. The * shut down of the E1 part is handled separately. */ mgcp_cleanup_rtp_bridge_cb(endp, conn); } /* Handle incoming RTP data from NET */ static void rtp_recvfrom_cb(struct osmo_io_fd *iofd, int res, struct msgb *msg, const struct osmo_sockaddr *saddr) { /* 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; enum rtp_proto proto; struct osmo_rtp_msg_ctx *mc; conn_src = (struct mgcp_conn_rtp *) osmo_iofd_get_data(iofd); OSMO_ASSERT(conn_src); endp = conn_src->conn->endp; OSMO_ASSERT(endp); proto = (iofd == conn_src->end.rtp) ? MGCP_PROTO_RTP : MGCP_PROTO_RTCP; if (res <= 0) { LOG_CONN_RTP(conn_src, LOGL_ERROR, "recvfrom error: %s\n", strerror(-res)); goto out_free; } LOG_CONN_RTP(conn_src, LOGL_DEBUG, "%s: rx %u bytes from %s\n", proto == MGCP_PROTO_RTP ? "RTP" : "RTCP", msgb_length(msg), osmo_sockaddr_to_str(saddr)); 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 */ goto out_free; } if (mgcp_is_rtp_dummy_payload(msg)) { LOG_CONN_RTP(conn_src, LOGL_DEBUG, "rx dummy packet (dropped)\n"); goto out_free; } /* 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 = (struct osmo_sockaddr *) saddr, }; LOG_CONN_RTP(conn_src, LOGL_DEBUG, "msg ctx: %d %p %s\n", mc->proto, mc->conn_src, osmo_hexdump((void*)mc->from_addr, mc->from_addr->u.sa.sa_family == AF_INET6 ? sizeof(struct sockaddr_in6) : sizeof(struct sockaddr_in))); /* Increment RX statistics */ rate_ctr_inc(rate_ctr_group_get_ctr(conn_src->ctrg, RTP_PACKETS_RX_CTR)); rate_ctr_add(rate_ctr_group_get_ctr(conn_src->ctrg, 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_tap(iofd, &conn_src->tap_in, msg); rx_rtp(msg); return; out_free: msgb_free(msg); } /* Note: This function is able to handle RTP and RTCP. msgb ownership is transferred, so this function or its * downstream consumers must make sure to [eventually] free the msgb. */ static int rx_rtp(struct msgb *msg) { struct osmo_rtp_msg_ctx *mc = OSMO_RTP_MSG_CTX(msg); struct mgcp_conn_rtp *conn_src = mc->conn_src; struct osmo_sockaddr *from_addr = mc->from_addr; struct mgcp_conn *conn = conn_src->conn; struct mgcp_trunk *trunk = conn->endp->trunk; LOG_CONN_RTP(conn_src, LOGL_DEBUG, "rx_rtp(%u bytes)\n", msgb_length(msg)); /* Check if the origin of the RTP packet seems plausible */ if (!trunk->rtp_accept_all && check_rtp_origin(conn_src, from_addr)) goto out_free; /* Handle AMR frame format conversion (octet-aligned vs. bandwith-efficient) */ if (mc->proto == MGCP_PROTO_RTP && conn_src->end.cset.codec && mgcp_codec_amr_align_mode_is_indicated(conn_src->end.cset.codec)) { /* Make sure that the incoming AMR frame format matches the frame format that the call agent has * communicated via SDP when the connection was created/modfied. */ int oa = amr_oa_check((char*)msgb_data(msg), msgb_length(msg)); if (oa < 0) goto out_free; if (((bool)oa) != conn_src->end.cset.codec->param.amr_octet_aligned) { LOG_CONN_RTP(conn_src, LOGL_NOTICE, "rx_rtp(%u bytes): Expected RTP AMR octet-aligned=%u but got octet-aligned=%u." " check the config of your call-agent!\n", msgb_length(msg), conn_src->end.cset.codec->param.amr_octet_aligned, oa); goto out_free; } } mgcp_conn_watchdog_kick(conn_src->conn); /* Execute endpoint specific implementation that handles the * dispatching of the RTP data */ return conn->endp->type->dispatch_rtp_cb(msg); out_free: msgb_free(msg); return -1; } static void rtp_sendto_cb(struct osmo_io_fd *iofd, int res, struct msgb *msg, const struct osmo_sockaddr *daddr) { /* nothing; osmo_io takes care of msgb_free */ if (res < 0) { struct mgcp_conn_rtp *conn_rtp = (struct mgcp_conn_rtp *) osmo_iofd_get_data(iofd); int priv_nr = osmo_iofd_get_priv_nr(iofd); char errbuf[129]; strerror_r(-res, errbuf, sizeof(errbuf)); LOG_CONN_RTP(conn_rtp, LOGL_ERROR, "%s sendto(%s) failed: %s\n", priv_nr ? "RTCP" : "RTP", osmo_sockaddr_to_str(daddr), errbuf); } } static const struct osmo_io_ops rtp_ioops = { .recvfrom_cb = rtp_recvfrom_cb, .sendto_cb = rtp_sendto_cb, }; /*! bind RTP port to osmo_fd. * \param[in] source_addr source (local) address to bind on. * \param[in] port to bind on. * \param[in] dscp IP DSCP value to use. * \param[in] prio socket priority to use. * \returns file descriptor on success, -1 on ERROR. */ int mgcp_create_bind(const char *source_addr, int port, uint8_t dscp, uint8_t prio) { int rc; rc = osmo_sock_init2(AF_UNSPEC, SOCK_DGRAM, IPPROTO_UDP, source_addr, port, NULL, 0, OSMO_SOCK_F_BIND | OSMO_SOCK_F_DSCP(dscp) | OSMO_SOCK_F_PRIO(prio)); if (rc < 0) { LOGP(DRTP, LOGL_ERROR, "failed to bind UDP port (%s:%d).\n", source_addr, port); return -1; } LOGP(DRTP, LOGL_DEBUG, "created socket + bound UDP port (%s:%d).\n", source_addr, port); return rc; } /*! bind RTP port to endpoint/connection. * \param[in] conn associated RTP connection. * \param[in] rtp_port port number to bind on. * \returns 0 on success, -1 on ERROR. */ int mgcp_conn_rtp_bind_rtp_ports(struct mgcp_conn_rtp *conn_rtp, int rtp_port) { char name[512]; struct mgcp_conn *conn = conn_rtp->conn; struct mgcp_config *cfg = conn->endp->trunk->cfg; struct mgcp_rtp_end *end = &conn_rtp->end; int rc, rtp_fd, rtcp_fd; snprintf(name, sizeof(name), "%s-%s", conn->name, conn->id); if ((end->rtp && osmo_iofd_get_fd(end->rtp) != -1) || (end->rtcp && osmo_iofd_get_fd(end->rtcp) != -1)) { LOG_CONN_RTP(conn_rtp, LOGL_ERROR, "%u was already bound\n", rtp_port); /* 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 = osmo_iofd_setup(conn, -1, name, OSMO_IO_FD_MODE_RECVFROM_SENDTO, &rtp_ioops, conn_rtp); if (!end->rtp) goto free_iofd_ret; osmo_iofd_set_alloc_info(end->rtp, RTP_BUF_SIZE, 0); end->rtcp = osmo_iofd_setup(conn, -1, name, OSMO_IO_FD_MODE_RECVFROM_SENDTO, &rtp_ioops, conn_rtp); if (!end->rtcp) goto free_iofd_ret; osmo_iofd_set_alloc_info(end->rtcp, RTP_BUF_SIZE, 0); osmo_iofd_set_priv_nr(end->rtcp, 1); /* we use priv_nr as identifier for RTCP */ /* NOTE: The port that is used for RTCP is the RTP port incremented by one * (e.g. RTP-Port = 16000 ==> RTCP-Port = 16001) */ rc = mgcp_create_bind(end->local_addr, end->local_port, cfg->endp_dscp, cfg->endp_priority); if (rc < 0) { LOG_CONN_RTP(conn_rtp, LOGL_ERROR, "failed to create RTP port: %s:%d\n", end->local_addr, end->local_port); goto free_iofd_ret; } rtp_fd = rc; rc = mgcp_create_bind(end->local_addr, end->local_port + 1, cfg->endp_dscp, cfg->endp_priority); if (rc < 0) { LOG_CONN_RTP(conn_rtp, LOGL_ERROR, "failed to create RTCP port: %s:%d\n", end->local_addr, end->local_port + 1); goto cleanup1; } rtcp_fd = rc; if (osmo_iofd_register(end->rtp, rtp_fd) < 0) { LOG_CONN_RTP(conn_rtp, LOGL_ERROR, "failed to register RTP port %d\n", end->local_port); goto cleanup2; } if (osmo_iofd_register(end->rtcp, rtcp_fd) != 0) { LOG_CONN_RTP(conn_rtp, LOGL_ERROR, "failed to register RTCP port %d\n", end->local_port + 1); goto cleanup3; } return 0; cleanup3: osmo_iofd_unregister(end->rtp); cleanup2: close(rtcp_fd); cleanup1: close(rtp_fd); free_iofd_ret: osmo_iofd_free(end->rtcp); end->rtcp = NULL; osmo_iofd_free(end->rtp); end->rtp = NULL; return -EIO; }