/* 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 * 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 #define RTP_SEQ_MOD (1 << 16) #define RTP_MAX_DROPOUT 3000 #define RTP_MAX_MISORDER 100 #define RTP_BUF_SIZE 4096 enum { MGCP_PROTO_RTP, MGCP_PROTO_RTCP, }; /*! 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) LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x CI:%s local interface auto detection failed, using configured addresses...\n", ENDPOINT_NUMBER(endp), conn->conn->id); else { LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x CI:%s selected local rtp bind ip %s by probing using remote ip %s\n", ENDPOINT_NUMBER(endp), conn->conn->id, addr, inet_ntoa(conn->end.addr)); return; } } /* Select from preconfigured IP-Addresses */ 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); LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x CI:%s using configured rtp bind ip as local bind ip %s\n", ENDPOINT_NUMBER(endp), conn->conn->id, 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); LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x CI:%s using mgcp bind ip as local rtp bind ip: %s\n", ENDPOINT_NUMBER(endp), conn->conn->id, 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 * \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); LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x sending dummy packet...\n", ENDPOINT_NUMBER(endp)); LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x conn:%s\n", ENDPOINT_NUMBER(endp), 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->tcfg->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: LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x Failed to send dummy %s packet.\n", ENDPOINT_NUMBER(endp), 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); LOGP(DRTP, LOGL_ERROR, "The %s timestamp delta is != 0 but the sequence " "number %d is the same, " "TS offset: %d, SeqNo offset: %d " "on 0x%x SSRC: %u timestamp: %u " "from %s:%d\n", text, seq, state->patch.timestamp_offset, state->patch.seq_offset, ENDPOINT_NUMBER(endp), 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 */ LOGP(DRTP, LOGL_NOTICE, "The %s timestamp delta is %d " "on 0x%x SSRC: %u timestamp: %u " "from %s:%d\n", text, tsdelta, ENDPOINT_NUMBER(endp), sstate->ssrc, timestamp, inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); return 0; } if (sstate->last_tsdelta != tsdelta) { if (sstate->last_tsdelta) { LOGP(DRTP, LOGL_INFO, "The %s timestamp delta changes from %d to %d " "on 0x%x SSRC: %u timestamp: %u from %s:%d\n", text, sstate->last_tsdelta, tsdelta, ENDPOINT_NUMBER(endp), 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); LOGP(DRTP, LOGL_NOTICE, "The %s timestamp has an alignment error of %d " "on 0x%x SSRC: %u " "SeqNo delta: %d, TS delta: %d, dTS/dSeq: %d " "from %s:%d. ptime: %d\n", text, timestamp_error, ENDPOINT_NUMBER(endp), 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) { LOGP(DRTP, LOGL_NOTICE, "A fixed packet duration is not available on 0x%x, " "using last output timestamp delta instead: %d " "from %s:%d\n", ENDPOINT_NUMBER(endp), tsdelta, inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); } else { tsdelta = rtp_end->codec->rate * 20 / 1000; LOGP(DRTP, LOGL_NOTICE, "Fixed packet duration and last timestamp delta " "are not available on 0x%x, " "using fixed 20ms instead: %d " "from %s:%d\n", ENDPOINT_NUMBER(endp), 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; LOGP(DRTP, LOGL_NOTICE, "Timestamp offset change on 0x%x SSRC: %u " "SeqNo delta: %d, TS offset: %d, " "from %s:%d\n", ENDPOINT_NUMBER(endp), 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; LOGP(DRTP, LOGL_NOTICE, "Corrected timestamp alignment error of %d on 0x%x SSRC: %u " "new TS offset: %d, " "from %s:%d\n", ts_error, ENDPOINT_NUMBER(endp), 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) { LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x transcoding disabled\n", ENDPOINT_NUMBER(endp)); 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) { LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x transcoding disabled\n", ENDPOINT_NUMBER(endp)); return 0; } void mgcp_get_net_downlink_format_default(struct mgcp_endpoint *endp, int *payload_type, const char **audio_name, const char **fmtp_extra, struct mgcp_conn_rtp *conn) { LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x conn:%s using format defaults\n", ENDPOINT_NUMBER(endp), mgcp_conn_dump(conn->conn)); *payload_type = conn->end.codec->payload_type; *audio_name = conn->end.codec->audio_name; *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) { LOGP(DRTP, LOGL_NOTICE, "RTP seqno made a very large jump on 0x%x delta: %u\n", ENDPOINT_NUMBER(endp), 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, char *data, int len) { struct rtp_hdr *rtp_hdr; uint8_t pt_in; int pt_out; OSMO_ASSERT(len >= sizeof(struct rtp_hdr)); rtp_hdr = (struct rtp_hdr *)data; 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, char *data, int len) { uint32_t arrival_time; int32_t transit; uint16_t seq; uint32_t timestamp, ssrc; struct rtp_hdr *rtp_hdr; int payload = rtp_end->codec->payload_type; if (len < sizeof(*rtp_hdr)) return; rtp_hdr = (struct rtp_hdr *)data; seq = ntohs(rtp_hdr->sequence); timestamp = ntohl(rtp_hdr->timestamp); arrival_time = get_current_ts(rtp_end->codec->rate); ssrc = ntohl(rtp_hdr->ssrc); transit = arrival_time - timestamp; mgcp_rtp_annex_count(endp, state, seq, transit, ssrc); if (!state->initialized) { state->initialized = 1; state->in_stream.last_seq = seq - 1; state->in_stream.ssrc = state->patch.orig_ssrc = ssrc; state->in_stream.last_tsdelta = 0; state->packet_duration = mgcp_rtp_packet_duration(endp, rtp_end); state->out_stream.last_seq = seq - 1; state->out_stream.ssrc = state->patch.orig_ssrc = ssrc; state->out_stream.last_tsdelta = 0; state->out_stream.last_timestamp = timestamp; state->out_stream.ssrc = ssrc - 1; /* force output SSRC change */ LOGP(DRTP, LOGL_INFO, "endpoint:0x%x initializing stream, SSRC: %u timestamp: %u " "pkt-duration: %d, from %s:%d\n", ENDPOINT_NUMBER(endp), state->in_stream.ssrc, state->patch.seq_offset, state->packet_duration, inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); if (state->packet_duration == 0) { state->packet_duration = rtp_end->codec->rate * 20 / 1000; LOGP(DRTP, LOGL_NOTICE, "endpoint:0x%x fixed packet duration is not available, " "using fixed 20ms instead: %d from %s:%d\n", ENDPOINT_NUMBER(endp), state->packet_duration, inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); } } else if (state->in_stream.ssrc != ssrc) { LOGP(DRTP, LOGL_NOTICE, "endpoint:0x%x SSRC changed: %u -> %u " "from %s:%d\n", ENDPOINT_NUMBER(endp), state->in_stream.ssrc, rtp_hdr->ssrc, inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); state->in_stream.ssrc = ssrc; if (rtp_end->force_constant_ssrc) { int16_t delta_seq; /* Always increment seqno by 1 */ state->patch.seq_offset = (state->out_stream.last_seq + 1) - seq; /* Estimate number of packets that would have been sent */ delta_seq = (arrival_time - state->in_stream.last_arrival_time + state->packet_duration / 2) / state->packet_duration; adjust_rtp_timestamp_offset(endp, state, rtp_end, addr, delta_seq, timestamp); state->patch.patch_ssrc = 1; ssrc = state->patch.orig_ssrc; if (rtp_end->force_constant_ssrc != -1) rtp_end->force_constant_ssrc -= 1; LOGP(DRTP, LOGL_NOTICE, "endpoint:0x%x SSRC patching enabled, SSRC: %u " "SeqNo offset: %d, TS offset: %d " "from %s:%d\n", ENDPOINT_NUMBER(endp), state->in_stream.ssrc, state->patch.seq_offset, state->patch.timestamp_offset, inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); } state->in_stream.last_tsdelta = 0; } else { /* Compute current per-packet timestamp delta */ check_rtp_timestamp(endp, state, &state->in_stream, rtp_end, addr, seq, timestamp, "input", &state->in_stream.last_tsdelta); if (state->patch.patch_ssrc) ssrc = state->patch.orig_ssrc; } /* Save before patching */ state->in_stream.last_timestamp = timestamp; state->in_stream.last_seq = seq; state->in_stream.last_arrival_time = arrival_time; if (rtp_end->force_aligned_timing && state->out_stream.ssrc == ssrc && state->packet_duration) /* Align the timestamp offset */ align_rtp_timestamp_offset(endp, state, rtp_end, addr, timestamp); /* Store the updated SSRC back to the packet */ if (state->patch.patch_ssrc) rtp_hdr->ssrc = htonl(ssrc); /* Apply the offset and store it back to the packet. * This won't change anything if the offset is 0, so the conditional is * omitted. */ seq += state->patch.seq_offset; rtp_hdr->sequence = htons(seq); timestamp += state->patch.timestamp_offset; rtp_hdr->timestamp = htonl(timestamp); /* Check again, whether the timestamps are still valid */ if (state->out_stream.ssrc == ssrc) check_rtp_timestamp(endp, state, &state->out_stream, rtp_end, addr, seq, timestamp, "output", &state->out_stream.last_tsdelta); /* Save output values */ state->out_stream.last_seq = seq; state->out_stream.last_timestamp = timestamp; state->out_stream.ssrc = ssrc; if (payload < 0) return; #if 0 DEBUGP(DRTP, "endpoint:0x%x payload hdr payload %u -> endp payload %u\n", ENDPOINT_NUMBER(endp), rtp_hdr->payload_type, payload); rtp_hdr->payload_type = payload; #endif } /* Forward data to a debug tap. This is debug function that is intended for * debugging the voice traffic with tools like gstreamer */ static void forward_data(int fd, struct mgcp_rtp_tap *tap, const char *buf, int len) { int rc; if (!tap->enabled) return; rc = sendto(fd, buf, len, 0, (struct sockaddr *)&tap->forward, sizeof(tap->forward)); if (rc < 0) LOGP(DRTP, LOGL_ERROR, "Forwarding tapped (debug) voice data failed.\n"); } /*! Send RTP/RTCP data to a specified destination connection. * \param[in] endp associated endpoint (for configuration, logging) * \param[in] is_rtp flag to specify if the packet is of type RTP or RTCP * \param[in] spoofed source address (set to NULL to disable) * \param[in] buf buffer that contains the RTP/RTCP data * \param[in] len length of the buffer that contains the RTP/RTCP data * \param[in] conn_src associated source connection * \param[in] conn_dst associated destination connection * \returns 0 on success, -1 on ERROR */ int mgcp_send(struct mgcp_endpoint *endp, int is_rtp, struct sockaddr_in *addr, char *buf, int len, struct mgcp_conn_rtp *conn_src, struct mgcp_conn_rtp *conn_dst) { /*! When no destination connection is available (e.g. when only one * connection in loopback mode exists), then the source connection * shall be specified as destination connection */ struct mgcp_trunk_config *tcfg = endp->tcfg; struct mgcp_rtp_end *rtp_end; struct mgcp_rtp_state *rtp_state; char *dest_name; int rc; OSMO_ASSERT(conn_src); OSMO_ASSERT(conn_dst); if (is_rtp) { LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x delivering RTP packet...\n", ENDPOINT_NUMBER(endp)); } else { LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x delivering RTCP packet...\n", ENDPOINT_NUMBER(endp)); } LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x loop:%d, mode:%d%s\n", ENDPOINT_NUMBER(endp), tcfg->audio_loop, conn_src->conn->mode, conn_src->conn->mode == MGCP_CONN_LOOPBACK ? " (loopback)" : ""); /* 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, buf, len); if (rc < 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x can not patch PT because no suitable egress codec was found.\n", ENDPOINT_NUMBER(endp)); } } /* Note: In case of loopback configuration, both, the source and the * destination will point to the same connection. */ rtp_end = &conn_dst->end; rtp_state = &conn_src->state; dest_name = conn_dst->conn->name; if (!rtp_end->output_enabled) { rate_ctr_inc(&conn_dst->rate_ctr_group->ctr[RTP_DROPPED_PACKETS_CTR]); LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x output disabled, drop to %s %s " "rtp_port:%u rtcp_port:%u\n", ENDPOINT_NUMBER(endp), dest_name, inet_ntoa(rtp_end->addr), ntohs(rtp_end->rtp_port), ntohs(rtp_end->rtcp_port) ); } else if (is_rtp) { int cont; int nbytes = 0; int buflen = len; do { /* Run transcoder */ cont = endp->cfg->rtp_processing_cb(endp, rtp_end, buf, &buflen, RTP_BUF_SIZE); if (cont < 0) break; if (addr) mgcp_patch_and_count(endp, rtp_state, rtp_end, addr, buf, buflen); LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x process/send to %s %s " "rtp_port:%u rtcp_port:%u\n", ENDPOINT_NUMBER(endp), dest_name, inet_ntoa(rtp_end->addr), ntohs(rtp_end->rtp_port), ntohs(rtp_end->rtcp_port) ); /* Forward a copy of the RTP data to a debug ip/port */ forward_data(rtp_end->rtp.fd, &conn_src->tap_out, buf, buflen); /* FIXME: HACK HACK HACK. See OS#2459. * The ip.access nano3G needs the first RTP payload's first two bytes to read hex * 'e400', or it will reject the RAB assignment. It seems to not harm other femto * cells (as long as we patch only the first RTP payload in each stream). */ if (!rtp_state->patched_first_rtp_payload && conn_src->conn->mode == MGCP_CONN_LOOPBACK) { uint8_t *data = (uint8_t *) & buf[12]; if (data[0] == 0xe0) { data[0] = 0xe4; data[1] = 0x00; rtp_state->patched_first_rtp_payload = true; LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x Patching over first two bytes" " to fake an IuUP Initialization Ack\n", ENDPOINT_NUMBER(endp)); } } len = mgcp_udp_send(rtp_end->rtp.fd, &rtp_end->addr, rtp_end->rtp_port, buf, buflen); if (len <= 0) return len; rate_ctr_inc(&conn_dst->rate_ctr_group->ctr[RTP_PACKETS_TX_CTR]); rate_ctr_add(&conn_dst->rate_ctr_group->ctr[RTP_OCTETS_TX_CTR], len); nbytes += len; buflen = cont; } while (buflen > 0); return nbytes; } else if (!tcfg->omit_rtcp) { LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x send to %s %s rtp_port:%u rtcp_port:%u\n", ENDPOINT_NUMBER(endp), dest_name, inet_ntoa(rtp_end->addr), ntohs(rtp_end->rtp_port), ntohs(rtp_end->rtcp_port) ); len = mgcp_udp_send(rtp_end->rtcp.fd, &rtp_end->addr, rtp_end->rtcp_port, buf, len); rate_ctr_inc(&conn_dst->rate_ctr_group->ctr[RTP_PACKETS_TX_CTR]); rate_ctr_add(&conn_dst->rate_ctr_group->ctr[RTP_OCTETS_TX_CTR], len); return len; } return 0; } /* Helper function for mgcp_recv(), Receive one RTP Packet + Originating address from file descriptor */ static int receive_from(struct mgcp_endpoint *endp, int fd, struct sockaddr_in *addr, char *buf, int bufsize) { int rc; socklen_t slen = sizeof(*addr); struct sockaddr_in addr_sink; char buf_sink[RTP_BUF_SIZE]; bool tossed = false; if (!addr) addr = &addr_sink; if (!buf) { tossed = true; buf = buf_sink; bufsize = sizeof(buf_sink); } rc = recvfrom(fd, buf, bufsize, 0, (struct sockaddr *)addr, &slen); LOGP(DRTP, LOGL_DEBUG, "receiving %u bytes length packet from %s:%u ...\n", rc, inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); if (rc < 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x failed to receive packet, errno: %d/%s\n", ENDPOINT_NUMBER(endp), errno, strerror(errno)); return -1; } if (tossed) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x packet tossed\n", ENDPOINT_NUMBER(endp)); } return rc; } /* Check if the origin (addr) matches the address/port data of the RTP * connections. */ static int check_rtp_origin(struct mgcp_conn_rtp *conn, struct sockaddr_in *addr) { struct mgcp_endpoint *endp; endp = conn->conn->endp; struct sockaddr_in zero_addr = {}; if (memcmp(&zero_addr.sin_addr, &conn->end.addr, sizeof(zero_addr.sin_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. */ LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x In loopback mode and remote address not set:" " allowing data from address: %s\n", ENDPOINT_NUMBER(endp), 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. */ LOGP(DRTP, LOGL_INFO, "endpoint:0x%x I:%s Rx RTP from %s, but remote address not set:" " dropping early media\n", ENDPOINT_NUMBER(endp), conn->conn->id, 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 (memcmp(&addr->sin_addr, &conn->end.addr, sizeof(addr->sin_addr)) != 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x data from wrong address: %s, ", ENDPOINT_NUMBER(endp), inet_ntoa(addr->sin_addr)); LOGPC(DRTP, LOGL_ERROR, "expected: %s\n", inet_ntoa(conn->end.addr)); LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x packet tossed\n", ENDPOINT_NUMBER(endp)); return -1; } /* Note: Usually the remote remote port of the data we receive will be * the same as the remote port where we transmit outgoing RTP traffic * to (set by MDCX). We use this to check the origin of the data for * plausibility. */ if (conn->end.rtp_port != addr->sin_port && conn->end.rtcp_port != addr->sin_port) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x data from wrong source port: %d, ", ENDPOINT_NUMBER(endp), ntohs(addr->sin_port)); LOGPC(DRTP, LOGL_ERROR, "expected: %d for RTP or %d for RTCP\n", ntohs(conn->end.rtp_port), ntohs(conn->end.rtcp_port)); LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x packet tossed\n", ENDPOINT_NUMBER(endp)); return -1; } return 0; } /* Check the if the destination address configuration of an RTP connection * makes sense */ static int check_rtp_destin(struct mgcp_conn_rtp *conn) { struct mgcp_endpoint *endp; endp = conn->conn->endp; /* Note: it is legal to create a connection but never setting a port * and IP-address for outgoing data. */ if (strcmp(inet_ntoa(conn->end.addr), "0.0.0.0") == 0 && conn->end.rtp_port == 0) { LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x destination IP-address and rtp port is (not yet) known\n", ENDPOINT_NUMBER(endp)); return -1; } if (strcmp(inet_ntoa(conn->end.addr), "0.0.0.0") == 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x destination IP-address is invalid\n", ENDPOINT_NUMBER(endp)); return -1; } if (conn->end.rtp_port == 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x destination rtp port is invalid\n", ENDPOINT_NUMBER(endp)); return -1; } return 0; } /* Do some basic checks to make sure that the RTCP packets we are going to * process are not complete garbage */ static int check_rtcp(char *buf, unsigned int buf_size) { 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 (buf_size < 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 *)buf; len = (osmo_ntohs(hdr->length) + 1) * 4; if (len > buf_size) 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)) 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(char *buf, unsigned int buf_size) { /* RTP packets that are just a header without data do not make * any sense. */ if (buf_size < sizeof(struct rtp_hdr)) return -EINVAL; /* 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; } /* Receive RTP data from a specified source connection and dispatch it to a * destination connection. */ static int mgcp_recv(int *proto, struct sockaddr_in *addr, char *buf, unsigned int buf_size, struct osmo_fd *fd) { struct mgcp_endpoint *endp; struct mgcp_conn_rtp *conn; struct mgcp_trunk_config *tcfg; int rc; conn = (struct mgcp_conn_rtp*) fd->data; endp = conn->conn->endp; tcfg = endp->tcfg; LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x receiving RTP/RTCP packet...\n", ENDPOINT_NUMBER(endp)); rc = receive_from(endp, fd->fd, addr, buf, buf_size); if (rc <= 0) return -1; /* FIXME: The way how we detect the protocol looks odd. We should look * into the packet header. Also we should introduce a packet type * MGCP_PROTO_IUUP because currently we handle IUUP packets like RTP * packets which is problematic. */ *proto = fd == &conn->end.rtp ? MGCP_PROTO_RTP : MGCP_PROTO_RTCP; if (*proto == MGCP_PROTO_RTP) { if (check_rtp(buf, rc) < 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x invalid RTP packet received -- packet tossed\n", ENDPOINT_NUMBER(endp)); return -1; } } else if (*proto == MGCP_PROTO_RTCP) { if (check_rtcp(buf, rc) < 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x invalid RTCP packet received -- packet tossed\n", ENDPOINT_NUMBER(endp)); return -1; } } LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x ", ENDPOINT_NUMBER(endp)); LOGPC(DRTP, LOGL_DEBUG, "receiving from %s %s %d\n", conn->conn->name, inet_ntoa(addr->sin_addr), ntohs(addr->sin_port)); LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x conn:%s\n", ENDPOINT_NUMBER(endp), mgcp_conn_dump(conn->conn)); /* Check if the origin of the RTP packet seems plausible */ if (tcfg->rtp_accept_all == 0) { if (check_rtp_origin(conn, addr) != 0) return -1; } /* Filter out dummy message */ if (rc == 1 && buf[0] == MGCP_DUMMY_LOAD) { LOGP(DRTP, LOGL_NOTICE, "endpoint:0x%x dummy message received\n", ENDPOINT_NUMBER(endp)); LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x packet tossed\n", ENDPOINT_NUMBER(endp)); return 0; } /* Increment RX statistics */ rate_ctr_inc(&conn->rate_ctr_group->ctr[RTP_PACKETS_RX_CTR]); rate_ctr_add(&conn->rate_ctr_group->ctr[RTP_OCTETS_RX_CTR], rc); /* Forward a copy of the RTP data to a debug ip/port */ forward_data(fd->fd, &conn->tap_in, buf, rc); return rc; } /* Send RTP data. Possible options are standard RTP packet * transmission or trsmission via an osmux connection */ static int mgcp_send_rtp(int proto, struct sockaddr_in *addr, char *buf, unsigned int buf_size, struct mgcp_conn_rtp *conn_src, struct mgcp_conn_rtp *conn_dst) { struct mgcp_endpoint *endp; endp = conn_src->conn->endp; LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x destin conn:%s\n", ENDPOINT_NUMBER(endp), mgcp_conn_dump(conn_dst->conn)); /* Before we try to deliver the packet, we check if the destination * port and IP-Address make sense at all. If not, we will be unable * to deliver the packet. */ if (check_rtp_destin(conn_dst) != 0) return -1; /* Depending on the RTP connection type, deliver the RTP packet to the * destination connection. */ switch (conn_dst->type) { case MGCP_RTP_DEFAULT: LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x endpoint type is MGCP_RTP_DEFAULT, " "using mgcp_send() to forward data directly\n", ENDPOINT_NUMBER(endp)); return mgcp_send(endp, proto == MGCP_PROTO_RTP, addr, buf, buf_size, conn_src, conn_dst); case MGCP_OSMUX_BSC_NAT: case MGCP_OSMUX_BSC: LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x endpoint type is MGCP_OSMUX_BSC_NAT, " "using osmux_xfrm_to_osmux() to forward data through OSMUX\n", ENDPOINT_NUMBER(endp)); return osmux_xfrm_to_osmux(buf, buf_size, conn_dst); } /* If the data has not been handled/forwarded until here, it will * be discarded, this should not happen, normally the MGCP type * should be properly set */ LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x bad MGCP type -- data discarded!\n", ENDPOINT_NUMBER(endp)); return -1; } /*! dispatch incoming RTP packet to opposite RTP connection. * \param[in] proto protocol (MGCP_CONN_TYPE_RTP or MGCP_CONN_TYPE_RTCP) * \param[in] addr socket address where the RTP packet has been received from * \param[in] buf buffer that hold the RTP payload * \param[in] buf_size size data length of buf * \param[in] conn originating connection * \returns 0 on success, -1 on ERROR */ int mgcp_dispatch_rtp_bridge_cb(int proto, struct sockaddr_in *addr, char *buf, unsigned int buf_size, struct mgcp_conn *conn) { struct mgcp_conn *conn_dst; struct mgcp_endpoint *endp; endp = conn->endp; /*! NOTE: This callback function implements the endpoint specific * dispatch bahviour of an rtp bridge/proxy endpoint. It is assumed * that the endpoint will hold only two connections. This premise * is used to determine the opposite connection (it is always the * connection that is not the originating connection). Once the * destination connection is known the RTP packet is sent via * the destination connection. */ /* Find a destination connection. */ /* NOTE: This code path runs every time an RTP packet is received. The * function mgcp_find_dst_conn() we use to determine the detination * connection will iterate the connection list inside the endpoint. * Since list iterations are quite costly, we will figure out the * destination only once and use the optional private data pointer of * the connection to cache the destination connection pointer. */ if (!conn->priv) { conn_dst = mgcp_find_dst_conn(conn); conn->priv = conn_dst; } else { conn_dst = (struct mgcp_conn *)conn->priv; } /* There is no destination conn, stop here */ if (!conn_dst) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x unable to find destination conn\n", ENDPOINT_NUMBER(endp)); return -1; } /* The destination conn is not an RTP connection */ if (conn_dst->type != MGCP_CONN_TYPE_RTP) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x unable to find suitable destination conn\n", ENDPOINT_NUMBER(endp)); return -1; } /* Dispatch RTP packet to destination RTP connection */ return mgcp_send_rtp(proto, addr, buf, buf_size, &conn->u.rtp, &conn_dst->u.rtp); } /*! cleanup an endpoint when a connection on an RTP bridge endpoint is removed. * \param[in] endp Endpoint on which the connection resides. * \param[in] conn Connection that is about to be removed (ignored). * \returns 0 on success, -1 on ERROR. */ void mgcp_cleanup_rtp_bridge_cb(struct mgcp_endpoint *endp, struct mgcp_conn *conn) { struct mgcp_conn *conn_cleanup; /* In mgcp_dispatch_rtp_bridge_cb() we use conn->priv to cache the * pointer to the destination connection, so that we do not have * to go through the list every time an RTP packet arrives. To prevent * a use-after-free situation we invalidate this information for all * connections present when one connection is removed from the * endpoint. */ llist_for_each_entry(conn_cleanup, &endp->conns, entry) { conn_cleanup->priv = NULL; } } /* Handle incoming RTP data from NET */ static int rtp_data_net(struct osmo_fd *fd, unsigned int what) { /* NOTE: This is a generic implementation. RTP data is received. In * case of loopback the data is just sent back to its origin. All * other cases implement endpoint specific behaviour (e.g. how is the * destination connection determined?). That specific behaviour is * implemented by the callback function that is called at the end of * the function */ struct mgcp_conn_rtp *conn_src; struct mgcp_endpoint *endp; struct sockaddr_in addr; char buf[RTP_BUF_SIZE]; int proto; int len; conn_src = (struct mgcp_conn_rtp *)fd->data; OSMO_ASSERT(conn_src); endp = conn_src->conn->endp; OSMO_ASSERT(endp); LOGP(DRTP, LOGL_DEBUG, "endpoint:0x%x source conn:%s\n", ENDPOINT_NUMBER(endp), mgcp_conn_dump(conn_src->conn)); /* Receive packet */ len = mgcp_recv(&proto, &addr, buf, sizeof(buf), fd); if (len < 0) return -1; /* Check if the connection is in loopback mode, if yes, just send the * incoming data back to the origin */ if (conn_src->conn->mode == MGCP_CONN_LOOPBACK) { /* When we are in loopback mode, we loop back all incoming * packets back to their origin. We will use the originating * address data from the UDP packet header to patch the * outgoing address in connection on the fly */ if (conn_src->end.rtp_port == 0) { conn_src->end.addr = addr.sin_addr; conn_src->end.rtp_port = addr.sin_port; } return mgcp_send_rtp(proto, &addr, buf, len, conn_src, conn_src); } /* Execute endpoint specific implementation that handles the * dispatching of the RTP data */ return endp->type->dispatch_rtp_cb(proto, &addr, buf, len, conn_src->conn); } /*! set IP Type of Service parameter. * \param[in] fd associated file descriptor * \param[in] tos dscp value * \returns 0 on success, -1 on ERROR */ int mgcp_set_ip_tos(int fd, int tos) { int ret; ret = setsockopt(fd, IPPROTO_IP, IP_TOS, &tos, sizeof(tos)); if (ret < 0) return -1; return 0; } /*! bind RTP port to osmo_fd. * \param[in] source_addr source (local) address to bind on * \param[in] fd associated file descriptor * \param[in] port to bind on * \returns 0 on success, -1 on ERROR */ int mgcp_create_bind(const char *source_addr, struct osmo_fd *fd, int port) { int rc; rc = osmo_sock_init2(AF_INET, SOCK_DGRAM, IPPROTO_UDP, source_addr, port, NULL, 0, OSMO_SOCK_F_BIND); if (rc < 0) { LOGP(DRTP, LOGL_ERROR, "failed to bind UDP port (%s:%i).\n", source_addr, port); return -1; } fd->fd = rc; LOGP(DRTP, LOGL_DEBUG, "created socket + bound UDP port (%s:%i).\n", source_addr, port); return 0; } /* Bind RTP and RTCP port (helper function for mgcp_bind_net_rtp_port()) */ static int bind_rtp(struct mgcp_config *cfg, const char *source_addr, struct mgcp_rtp_end *rtp_end, int endpno) { /* NOTE: The port that is used for RTCP is the RTP port incremented by one * (e.g. RTP-Port = 16000 ==> RTCP-Port = 16001) */ if (mgcp_create_bind(source_addr, &rtp_end->rtp, rtp_end->local_port) != 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x failed to create RTP port: %s:%d\n", endpno, source_addr, rtp_end->local_port); goto cleanup0; } if (mgcp_create_bind(source_addr, &rtp_end->rtcp, rtp_end->local_port + 1) != 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x failed to create RTCP port: %s:%d\n", endpno, source_addr, rtp_end->local_port + 1); goto cleanup1; } /* Set Type of Service (DSCP-Value) as configured via VTY */ mgcp_set_ip_tos(rtp_end->rtp.fd, cfg->endp_dscp); mgcp_set_ip_tos(rtp_end->rtcp.fd, cfg->endp_dscp); rtp_end->rtp.when = BSC_FD_READ; if (osmo_fd_register(&rtp_end->rtp) != 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x failed to register RTP port %d\n", endpno, rtp_end->local_port); goto cleanup2; } rtp_end->rtcp.when = BSC_FD_READ; if (osmo_fd_register(&rtp_end->rtcp) != 0) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x failed to register RTCP port %d\n", endpno, rtp_end->local_port + 1); goto cleanup3; } return 0; cleanup3: osmo_fd_unregister(&rtp_end->rtp); cleanup2: close(rtp_end->rtcp.fd); rtp_end->rtcp.fd = -1; cleanup1: close(rtp_end->rtp.fd); rtp_end->rtp.fd = -1; cleanup0: return -1; } /*! bind RTP port to endpoint/connection. * \param[in] endp endpoint that holds the RTP connection * \param[in] rtp_port port number to bind on * \param[in] conn associated RTP connection * \returns 0 on success, -1 on ERROR */ int mgcp_bind_net_rtp_port(struct mgcp_endpoint *endp, int rtp_port, struct mgcp_conn_rtp *conn) { char name[512]; struct mgcp_rtp_end *end; char local_ip_addr[INET_ADDRSTRLEN]; snprintf(name, sizeof(name), "%s-%s", conn->conn->name, conn->conn->id); end = &conn->end; if (end->rtp.fd != -1 || end->rtcp.fd != -1) { LOGP(DRTP, LOGL_ERROR, "endpoint:0x%x %u was already bound on conn:%s\n", ENDPOINT_NUMBER(endp), rtp_port, mgcp_conn_dump(conn->conn)); /* Double bindings should never occour! Since we always allocate * connections dynamically and free them when they are not * needed anymore, there must be no previous binding leftover. * Should there be a connection bound twice, we have a serious * problem and must exit immediately! */ OSMO_ASSERT(false); } end->local_port = rtp_port; end->rtp.cb = rtp_data_net; end->rtp.data = conn; end->rtcp.data = conn; end->rtcp.cb = rtp_data_net; mgcp_get_local_addr(local_ip_addr, conn); return bind_rtp(endp->cfg, local_ip_addr, end, ENDPOINT_NUMBER(endp)); } /*! free allocated RTP and RTCP ports. * \param[in] end RTP end */ void mgcp_free_rtp_port(struct mgcp_rtp_end *end) { if (end->rtp.fd != -1) { close(end->rtp.fd); end->rtp.fd = -1; osmo_fd_unregister(&end->rtp); } if (end->rtcp.fd != -1) { close(end->rtcp.fd); end->rtcp.fd = -1; osmo_fd_unregister(&end->rtcp); } }