From 218e4b4aa0fc6de842ff820dec8e97d1f083268a Mon Sep 17 00:00:00 2001 From: Neels Hofmeyr Date: Tue, 4 Jul 2017 23:08:44 +0200 Subject: move openbsc/* to repos root This is the first step in creating this repository from the legacy openbsc.git. Like all other Osmocom repositories, keep the autoconf and automake files in the repository root. openbsc.git has been the sole exception, which ends now. Change-Id: I9c6f2a448d9cb1cc088cf1cf6918b69d7e69b4e7 --- src/libtrau/rtp_proxy.c | 764 ++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 764 insertions(+) create mode 100644 src/libtrau/rtp_proxy.c (limited to 'src/libtrau/rtp_proxy.c') diff --git a/src/libtrau/rtp_proxy.c b/src/libtrau/rtp_proxy.c new file mode 100644 index 000000000..6b38ee5fe --- /dev/null +++ b/src/libtrau/rtp_proxy.c @@ -0,0 +1,764 @@ +/* RTP proxy handling for ip.access nanoBTS */ + +/* (C) 2009-2013 by Harald Welte + * 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 /* gettimeofday() */ +#include /* get..() */ +#include /* clock() */ +#include /* uname() */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include + +/* attempt to determine byte order */ +#include +#include + +static LLIST_HEAD(rtp_sockets); + +/* should we mangle the CNAME inside SDES of RTCP packets? We disable + * this by default, as it seems to be not needed */ +static int mangle_rtcp_cname = 0; + +enum rtp_bfd_priv { + RTP_PRIV_NONE, + RTP_PRIV_RTP, + RTP_PRIV_RTCP +}; + +#define RTP_ALLOC_SIZE 1500 + +#define RTCP_TYPE_SDES 202 + +#define RTCP_IE_CNAME 1 + + +#define RTP_VERSION 2 + +/* 33 for FR, all other codecs have smaller size */ +#define MAX_RTP_PAYLOAD_LEN 33 + +/* decode an rtp frame and create a new buffer with payload */ +static int rtp_decode(struct msgb *msg, uint32_t callref, struct msgb **data) +{ + struct msgb *new_msg; + struct gsm_data_frame *frame; + struct rtp_hdr *rtph = (struct rtp_hdr *)msg->data; + struct rtp_x_hdr *rtpxh; + uint8_t *payload, *payload_out; + int payload_len; + int msg_type; + int x_len; + + if (msg->len < 12) { + DEBUGPC(DLMUX, "received RTP frame too short (len = %d)\n", + msg->len); + return -EINVAL; + } + if (rtph->version != RTP_VERSION) { + DEBUGPC(DLMUX, "received RTP version %d not supported.\n", + rtph->version); + return -EINVAL; + } + payload = msg->data + sizeof(struct rtp_hdr) + (rtph->csrc_count << 2); + payload_len = msg->len - sizeof(struct rtp_hdr) - (rtph->csrc_count << 2); + if (payload_len < 0) { + DEBUGPC(DLMUX, "received RTP frame too short (len = %d, " + "csrc count = %d)\n", msg->len, rtph->csrc_count); + return -EINVAL; + } + if (rtph->extension) { + if (payload_len < sizeof(struct rtp_x_hdr)) { + DEBUGPC(DLMUX, "received RTP frame too short for " + "extension header\n"); + return -EINVAL; + } + rtpxh = (struct rtp_x_hdr *)payload; + x_len = ntohs(rtpxh->length) * 4 + sizeof(struct rtp_x_hdr); + payload += x_len; + payload_len -= x_len; + if (payload_len < 0) { + DEBUGPC(DLMUX, "received RTP frame too short, " + "extension header exceeds frame length\n"); + return -EINVAL; + } + } + if (rtph->padding) { + if (payload_len < 1) { + DEBUGPC(DLMUX, "received RTP frame too short for " + "padding length\n"); + return -EINVAL; + } + payload_len -= payload[payload_len - 1]; + if (payload_len < 0) { + DEBUGPC(DLMUX, "received RTP frame with padding " + "greater than payload\n"); + return -EINVAL; + } + } + + switch (rtph->payload_type) { + case RTP_PT_GSM_FULL: + msg_type = GSM_TCHF_FRAME; + if (payload_len != RTP_LEN_GSM_FULL) { + DEBUGPC(DLMUX, "received RTP full rate frame with " + "payload length != %d (len = %d)\n", + RTP_LEN_GSM_FULL, payload_len); + return -EINVAL; + } + break; + case RTP_PT_GSM_EFR: + msg_type = GSM_TCHF_FRAME_EFR; + if (payload_len != RTP_LEN_GSM_EFR) { + DEBUGPC(DLMUX, "received RTP extended full rate frame " + "with payload length != %d (len = %d)\n", + RTP_LEN_GSM_EFR, payload_len); + return -EINVAL; + } + break; + case RTP_PT_GSM_HALF: + msg_type = GSM_TCHH_FRAME; + if (payload_len != RTP_LEN_GSM_HALF) { + DEBUGPC(DLMUX, "received RTP half rate frame with " + "payload length != %d (len = %d)\n", + RTP_LEN_GSM_HALF, payload_len); + return -EINVAL; + } + break; + case RTP_PT_AMR: + msg_type = GSM_TCH_FRAME_AMR; + break; + default: + DEBUGPC(DLMUX, "received RTP frame with unknown payload " + "type %d\n", rtph->payload_type); + return -EINVAL; + } + + if (payload_len > MAX_RTP_PAYLOAD_LEN || + (rtph->payload_type == RTP_PT_AMR && + payload_len > MAX_RTP_PAYLOAD_LEN - 1)) { + DEBUGPC(DLMUX, "RTP payload too large (%d octets)\n", + payload_len); + return -EINVAL; + } + + /* always allocate for the maximum possible size to avoid + * fragmentation */ + new_msg = msgb_alloc(sizeof(struct gsm_data_frame) + + MAX_RTP_PAYLOAD_LEN+1, "GSM-DATA (TCH)"); + + if (!new_msg) + return -ENOMEM; + frame = (struct gsm_data_frame *) msgb_put(new_msg, sizeof(struct gsm_data_frame)); + frame->msg_type = msg_type; + frame->callref = callref; + if (rtph->payload_type == RTP_PT_AMR) { + /* for FR/HR/EFR the length is implicit. In AMR, we + * need to make it explicit by using the first byte of + * the data[] buffer as length byte */ + uint8_t *data0 = msgb_put(new_msg, 1); + *data0 = payload_len; + } + payload_out = msgb_put(new_msg, payload_len); + memcpy(payload_out, payload, payload_len); + + *data = new_msg; + return 0; +} + +/*! \brief encode and send a rtp frame + * \param[in] rs RTP socket through which we shall send + * \param[in] frame GSM RTP frame to be sent + */ +int rtp_send_frame(struct rtp_socket *rs, struct gsm_data_frame *frame) +{ + struct rtp_sub_socket *rss = &rs->rtp; + struct msgb *msg; + struct rtp_hdr *rtph; + uint8_t *payload; + int payload_type; + int payload_len; + int duration; /* in samples */ + int is_bfi = 0; + + if (rs->tx_action != RTP_SEND_DOWNSTREAM) { + /* initialize sequences */ + rs->tx_action = RTP_SEND_DOWNSTREAM; + rs->transmit.ssrc = rand(); + rs->transmit.sequence = random(); + rs->transmit.timestamp = random(); + } + + switch (frame->msg_type) { + case GSM_TCHF_FRAME: + payload_type = RTP_PT_GSM_FULL; + payload_len = RTP_LEN_GSM_FULL; + duration = RTP_GSM_DURATION; + break; + case GSM_TCHF_FRAME_EFR: + payload_type = RTP_PT_GSM_EFR; + payload_len = RTP_LEN_GSM_EFR; + duration = RTP_GSM_DURATION; + break; + case GSM_TCHH_FRAME: + payload_type = RTP_PT_GSM_HALF; + payload_len = RTP_LEN_GSM_HALF; + duration = RTP_GSM_DURATION; + break; + case GSM_TCH_FRAME_AMR: + payload_type = RTP_PT_AMR; + payload_len = frame->data[0]; + duration = RTP_GSM_DURATION; + break; + case GSM_BAD_FRAME: + payload_type = 0; + payload_len = 0; + duration = RTP_GSM_DURATION; + is_bfi = 1; + break; + default: + DEBUGPC(DLMUX, "unsupported message type %d\n", + frame->msg_type); + return -EINVAL; + } + + if (payload_len > MAX_RTP_PAYLOAD_LEN) { + DEBUGPC(DLMUX, "RTP payload too large (%d octets)\n", + payload_len); + return -EINVAL; + } + + if (is_bfi) { + /* In case of a bad frame, just count and drop packet. */ + rs->transmit.timestamp += duration; + rs->transmit.sequence++; + return 0; + } + + msg = msgb_alloc(sizeof(struct rtp_hdr) + payload_len, "RTP-GSM"); + if (!msg) + return -ENOMEM; + rtph = (struct rtp_hdr *) msgb_put(msg, sizeof(struct rtp_hdr)); + rtph->version = RTP_VERSION; + rtph->padding = 0; + rtph->extension = 0; + rtph->csrc_count = 0; + rtph->marker = 0; + rtph->payload_type = payload_type; + rtph->sequence = htons(rs->transmit.sequence++); + rtph->timestamp = htonl(rs->transmit.timestamp); + rs->transmit.timestamp += duration; + rtph->ssrc = htonl(rs->transmit.ssrc); + + payload = msgb_put(msg, payload_len); + if (frame->msg_type == GSM_TCH_FRAME_AMR) + memcpy(payload, frame->data + 1, payload_len); + else + memcpy(payload, frame->data, payload_len); + msgb_enqueue(&rss->tx_queue, msg); + rss->bfd.when |= BSC_FD_WRITE; + + return 0; +} + +/* iterate over all chunks in one RTCP message, look for CNAME IEs and + * replace all of those with 'new_cname' */ +static int rtcp_sdes_cname_mangle(struct msgb *msg, struct rtcp_hdr *rh, + uint16_t *rtcp_len, const char *new_cname) +{ + uint8_t *rtcp_end; + uint8_t *cur = (uint8_t *) rh; + uint8_t tag, len = 0; + + rtcp_end = cur + *rtcp_len; + /* move cur to end of RTP header */ + cur += sizeof(*rh); + + /* iterate over Chunks */ + while (cur+4 < rtcp_end) { + /* skip four bytes SSRC/CSRC */ + cur += 4; + + /* iterate over IE's inside the chunk */ + while (cur+1 < rtcp_end) { + tag = *cur++; + if (tag == 0) { + /* end of chunk, skip additional zero */ + while ((*cur++ == 0) && (cur < rtcp_end)) { } + break; + } + len = *cur++; + + if (tag == RTCP_IE_CNAME) { + /* we've found the CNAME, lets mangle it */ + if (len < strlen(new_cname)) { + /* we need to make more space */ + int increase = strlen(new_cname) - len; + + msgb_push(msg, increase); + memmove(cur+len+increase, cur+len, + rtcp_end - (cur+len)); + /* FIXME: we have to respect RTCP + * padding/alignment rules! */ + len += increase; + *(cur-1) += increase; + rtcp_end += increase; + *rtcp_len += increase; + } + /* copy new CNAME into message */ + memcpy(cur, new_cname, strlen(new_cname)); + /* FIXME: zero the padding in case new CNAME + * is smaller than old one !!! */ + } + cur += len; + } + } + + return 0; +} + +static int rtcp_mangle(struct msgb *msg, struct rtp_socket *rs) +{ + struct rtp_sub_socket *rss = &rs->rtcp; + struct rtcp_hdr *rtph; + uint16_t old_len; + int rc; + + if (!mangle_rtcp_cname) + return 0; + + printf("RTCP\n"); + /* iterate over list of RTCP messages */ + rtph = (struct rtcp_hdr *)msg->data; + while ((void *)rtph + sizeof(*rtph) <= (void *)msg->data + msg->len) { + old_len = (ntohs(rtph->length) + 1) * 4; + if ((void *)rtph + old_len > (void *)msg->data + msg->len) { + DEBUGPC(DLMUX, "received RTCP packet too short for " + "length element\n"); + return -EINVAL; + } + if (rtph->type == RTCP_TYPE_SDES) { + char new_cname[255]; + osmo_strlcpy(new_cname, + inet_ntoa(rss->sin_local.sin_addr), + sizeof(new_cname)); + rc = rtcp_sdes_cname_mangle(msg, rtph, &old_len, + new_cname); + if (rc < 0) + return rc; + } + rtph = (void *)rtph + old_len; + } + + return 0; +} + +/* read from incoming RTP/RTCP socket */ +static int rtp_socket_read(struct rtp_socket *rs, struct rtp_sub_socket *rss) +{ + int rc; + struct msgb *msg = msgb_alloc(RTP_ALLOC_SIZE, "RTP/RTCP"); + struct msgb *new_msg; + struct rtp_sub_socket *other_rss; + + if (!msg) + return -ENOMEM; + + rc = read(rss->bfd.fd, msg->data, RTP_ALLOC_SIZE); + if (rc == 0) { + rss->bfd.when &= ~BSC_FD_READ; + goto out_free; + } else if (rc < 0) { + /* Ignore "connection refused". this happens, If we open the + * socket faster than the remote side. */ + if (errno == ECONNREFUSED) + goto out_free; + DEBUGPC(DLMUX, "Read of RTP socket (%p) failed (errno %d, " + "%s)\n", rs, errno, strerror(errno)); + rss->bfd.when &= ~BSC_FD_READ; + goto out_free; + } + + msgb_put(msg, rc); + + switch (rs->rx_action) { + case RTP_PROXY: + if (!rs->proxy.other_sock) { + rc = -EIO; + goto out_free; + } + if (rss->bfd.priv_nr == RTP_PRIV_RTP) + other_rss = &rs->proxy.other_sock->rtp; + else if (rss->bfd.priv_nr == RTP_PRIV_RTCP) { + other_rss = &rs->proxy.other_sock->rtcp; + /* modify RTCP SDES CNAME */ + rc = rtcp_mangle(msg, rs); + if (rc < 0) + goto out_free; + } else { + rc = -EINVAL; + goto out_free; + } + msgb_enqueue(&other_rss->tx_queue, msg); + other_rss->bfd.when |= BSC_FD_WRITE; + break; + + case RTP_RECV_UPSTREAM: + if (!rs->receive.callref || !rs->receive.net) { + rc = -EIO; + goto out_free; + } + if (rss->bfd.priv_nr == RTP_PRIV_RTCP) { + if (!mangle_rtcp_cname) { + msgb_free(msg); + break; + } + /* modify RTCP SDES CNAME */ + rc = rtcp_mangle(msg, rs); + if (rc < 0) + goto out_free; + msgb_enqueue(&rss->tx_queue, msg); + rss->bfd.when |= BSC_FD_WRITE; + break; + } + if (rss->bfd.priv_nr != RTP_PRIV_RTP) { + rc = -EINVAL; + goto out_free; + } + rc = rtp_decode(msg, rs->receive.callref, &new_msg); + if (rc < 0) + goto out_free; + msgb_free(msg); + trau_tx_to_mncc(rs->receive.net, new_msg); + break; + + case RTP_NONE: /* if socket exists, but disabled by app */ + msgb_free(msg); + break; + } + + return 0; + +out_free: + msgb_free(msg); + return rc; +} + +/* \brief write from tx_queue to RTP/RTCP socket */ +static int rtp_socket_write(struct rtp_socket *rs, struct rtp_sub_socket *rss) +{ + struct msgb *msg; + int written; + + msg = msgb_dequeue(&rss->tx_queue); + if (!msg) { + rss->bfd.when &= ~BSC_FD_WRITE; + return 0; + } + + written = write(rss->bfd.fd, msg->data, msg->len); + if (written < msg->len) { + LOGP(DLMIB, LOGL_ERROR, "short write"); + msgb_free(msg); + return -EIO; + } + + msgb_free(msg); + + return 0; +} + + +/*! \brief callback for the select.c:bfd_* layer */ +static int rtp_bfd_cb(struct osmo_fd *bfd, unsigned int flags) +{ + struct rtp_socket *rs = bfd->data; + struct rtp_sub_socket *rss; + + switch (bfd->priv_nr) { + case RTP_PRIV_RTP: + rss = &rs->rtp; + break; + case RTP_PRIV_RTCP: + rss = &rs->rtcp; + break; + default: + return -EINVAL; + } + + if (flags & BSC_FD_READ) + rtp_socket_read(rs, rss); + + if (flags & BSC_FD_WRITE) + rtp_socket_write(rs, rss); + + return 0; +} + +/*! \brief initialize one rtp sub-socket */ +static void init_rss(struct rtp_sub_socket *rss, + struct rtp_socket *rs, int fd, int priv_nr) +{ + /* initialize bfd */ + rss->bfd.fd = fd; + rss->bfd.data = rs; + rss->bfd.priv_nr = priv_nr; + rss->bfd.cb = rtp_bfd_cb; +} + +/*! \brief create a new RTP/RTCP socket and bind it */ +struct rtp_socket *rtp_socket_create(void) +{ + int rc; + struct rtp_socket *rs; + + DEBUGP(DLMUX, "rtp_socket_create(): "); + + rs = talloc_zero(tall_bsc_ctx, struct rtp_socket); + if (!rs) + return NULL; + + INIT_LLIST_HEAD(&rs->rtp.tx_queue); + INIT_LLIST_HEAD(&rs->rtcp.tx_queue); + + rc = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP); + if (rc < 0) + goto out_free; + + init_rss(&rs->rtp, rs, rc, RTP_PRIV_RTP); + rc = osmo_fd_register(&rs->rtp.bfd); + if (rc < 0) + goto out_rtp_socket; + + rc = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP); + if (rc < 0) + goto out_rtp_bfd; + + init_rss(&rs->rtcp, rs, rc, RTP_PRIV_RTCP); + rc = osmo_fd_register(&rs->rtcp.bfd); + if (rc < 0) + goto out_rtcp_socket; + + DEBUGPC(DLMUX, "success\n"); + + rc = rtp_socket_bind(rs, INADDR_ANY); + if (rc < 0) + goto out_rtcp_bfd; + + return rs; + +out_rtcp_bfd: + osmo_fd_unregister(&rs->rtcp.bfd); +out_rtcp_socket: + close(rs->rtcp.bfd.fd); +out_rtp_bfd: + osmo_fd_unregister(&rs->rtp.bfd); +out_rtp_socket: + close(rs->rtp.bfd.fd); +out_free: + talloc_free(rs); + DEBUGPC(DLMUX, "failed\n"); + return NULL; +} + +static int rtp_sub_socket_bind(struct rtp_sub_socket *rss, uint32_t ip, + uint16_t port) +{ + int rc; + socklen_t alen = sizeof(rss->sin_local); + + rss->sin_local.sin_family = AF_INET; + rss->sin_local.sin_addr.s_addr = htonl(ip); + rss->sin_local.sin_port = htons(port); + rss->bfd.when |= BSC_FD_READ; + + rc = bind(rss->bfd.fd, (struct sockaddr *)&rss->sin_local, + sizeof(rss->sin_local)); + if (rc < 0) + return rc; + + /* retrieve the address we actually bound to, in case we + * passed INADDR_ANY as IP address */ + return getsockname(rss->bfd.fd, (struct sockaddr *)&rss->sin_local, + &alen); +} + +#define RTP_PORT_BASE 30000 +static unsigned int next_udp_port = RTP_PORT_BASE; + +/*! \brief bind a RTP socket to a specific local address + * \param[in] rs RTP socket to be bound + * \param[in] ip local IP address to which socket is to be bound + */ +int rtp_socket_bind(struct rtp_socket *rs, uint32_t ip) +{ + int rc = -EIO; + struct in_addr ia; + + ia.s_addr = htonl(ip); + DEBUGP(DLMUX, "rtp_socket_bind(rs=%p, IP=%s): ", rs, + inet_ntoa(ia)); + + /* try to bind to a consecutive pair of ports */ + for (next_udp_port = next_udp_port % 0xffff; + next_udp_port < 0xffff; next_udp_port += 2) { + rc = rtp_sub_socket_bind(&rs->rtp, ip, next_udp_port); + if (rc != 0) + continue; + + rc = rtp_sub_socket_bind(&rs->rtcp, ip, next_udp_port+1); + if (rc == 0) + break; + } + if (rc < 0) { + DEBUGPC(DLMUX, "failed\n"); + return rc; + } + + ia.s_addr = rs->rtp.sin_local.sin_addr.s_addr; + DEBUGPC(DLMUX, "BOUND_IP=%s, BOUND_PORT=%u\n", + inet_ntoa(ia), ntohs(rs->rtp.sin_local.sin_port)); + return ntohs(rs->rtp.sin_local.sin_port); +} + +static int rtp_sub_socket_connect(struct rtp_sub_socket *rss, + uint32_t ip, uint16_t port) +{ + int rc; + socklen_t alen = sizeof(rss->sin_local); + + rss->sin_remote.sin_family = AF_INET; + rss->sin_remote.sin_addr.s_addr = htonl(ip); + rss->sin_remote.sin_port = htons(port); + + rc = connect(rss->bfd.fd, (struct sockaddr *) &rss->sin_remote, + sizeof(rss->sin_remote)); + if (rc < 0) + return rc; + + return getsockname(rss->bfd.fd, (struct sockaddr *)&rss->sin_local, + &alen); +} + +/*! \brief 'connect' a RTP socket to a remote peer + * \param[in] rs RTP socket to be connected + * \param[in] ip remote IP address to which to connect + * \param[in] port remote UDP port number to which to connect + */ +int rtp_socket_connect(struct rtp_socket *rs, uint32_t ip, uint16_t port) +{ + int rc; + struct in_addr ia; + + ia.s_addr = htonl(ip); + DEBUGP(DLMUX, "rtp_socket_connect(rs=%p, ip=%s, port=%u)\n", + rs, inet_ntoa(ia), port); + + rc = rtp_sub_socket_connect(&rs->rtp, ip, port); + if (rc < 0) + return rc; + + return rtp_sub_socket_connect(&rs->rtcp, ip, port+1); +} + +/*! \brief bind two RTP/RTCP sockets together in the proxy + * \param[in] this First RTP socket + * \param[in] other Second RTP socket + */ +int rtp_socket_proxy(struct rtp_socket *this, struct rtp_socket *other) +{ + DEBUGP(DLMUX, "rtp_socket_proxy(this=%p, other=%p)\n", + this, other); + + this->rx_action = RTP_PROXY; + this->proxy.other_sock = other; + + other->rx_action = RTP_PROXY; + other->proxy.other_sock = this; + + return 0; +} + +/*! \brief bind RTP/RTCP socket to application, disabling proxy + * \param[in] this RTP socket + * \param[in] net gsm_network argument to trau_tx_to_mncc() + * \param[in] callref callref argument to trau_tx_to_mncc() + */ +int rtp_socket_upstream(struct rtp_socket *this, struct gsm_network *net, + uint32_t callref) +{ + DEBUGP(DLMUX, "rtp_socket_proxy(this=%p, callref=%u)\n", + this, callref); + + if (callref) { + this->rx_action = RTP_RECV_UPSTREAM; + this->receive.net = net; + this->receive.callref = callref; + } else + this->rx_action = RTP_NONE; + + return 0; +} + +static void free_tx_queue(struct rtp_sub_socket *rss) +{ + struct msgb *msg; + + while ((msg = msgb_dequeue(&rss->tx_queue))) + msgb_free(msg); +} + +/*! \brief Free/release a previously allocated RTP socket + * \param[in[] rs RTP/RTCP socket to be released + */ +int rtp_socket_free(struct rtp_socket *rs) +{ + DEBUGP(DLMUX, "rtp_socket_free(rs=%p)\n", rs); + + /* make sure we don't leave references dangling to us */ + if (rs->rx_action == RTP_PROXY && + rs->proxy.other_sock) + rs->proxy.other_sock->proxy.other_sock = NULL; + + osmo_fd_unregister(&rs->rtp.bfd); + close(rs->rtp.bfd.fd); + free_tx_queue(&rs->rtp); + + osmo_fd_unregister(&rs->rtcp.bfd); + close(rs->rtcp.bfd.fd); + free_tx_queue(&rs->rtcp); + + talloc_free(rs); + + return 0; +} -- cgit v1.2.3