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osmo-mgw/src/libosmo-mgcp/mgcp_network.c

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/* A Media Gateway Control Protocol Media Gateway: RFC 3435 */
/* The protocol implementation */
/*
* (C) 2009-2012 by Holger Hans Peter Freyther <zecke@selfish.org>
* (C) 2009-2012 by On-Waves
* All Rights Reserved
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU Affero General Public License as published by
* the Free Software Foundation; either version 3 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU Affero General Public License for more details.
*
* You should have received a copy of the GNU Affero General Public License
* along with this program. If not, see <http://www.gnu.org/licenses/>.
*
*/
#include <string.h>
#include <stdlib.h>
#include <unistd.h>
#include <errno.h>
#include <time.h>
#include <limits.h>
#include <arpa/inet.h>
#include <osmocom/core/msgb.h>
#include <osmocom/core/select.h>
#include <osmocom/core/socket.h>
#include <osmocom/core/byteswap.h>
#include <osmocom/netif/rtp.h>
#include <osmocom/netif/amr.h>
#include <osmocom/mgcp/mgcp.h>
#include <osmocom/mgcp/mgcp_common.h>
#include <osmocom/mgcp/mgcp_network.h>
#include <osmocom/mgcp/mgcp_protocol.h>
#include <osmocom/mgcp/mgcp_stat.h>
#include <osmocom/mgcp/osmux.h>
#include <osmocom/mgcp/mgcp_conn.h>
#include <osmocom/mgcp/mgcp_endp.h>
#include <osmocom/mgcp/mgcp_trunk.h>
#include <osmocom/mgcp/mgcp_codec.h>
#include <osmocom/mgcp/debug.h>
#include <osmocom/codec/codec.h>
#include <osmocom/mgcp/mgcp_e1.h>
#include <osmocom/mgcp/mgcp_iuup.h>
#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);
}
static int rx_rtp(struct msgb *msg);
bool mgcp_rtp_end_remote_addr_available(const struct mgcp_rtp_end *rtp_end)
{
return (osmo_sockaddr_port(&rtp_end->addr.u.sa) != 0) &&
(osmo_sockaddr_is_any(&rtp_end->addr) == 0);
}
/*! 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->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;
}
/*! dummy callback to disable transcoding (see also cfg->rtp_processing_cb).
* \param[in] associated endpoint.
* \param[in] destination RTP end.
* \param[in,out] pointer to buffer with voice data.
* \param[in] voice data length.
* \param[in] maximum size of caller provided voice data buffer.
* \returns ignores input parameters, return always 0. */
int mgcp_rtp_processing_default(struct mgcp_endpoint *endp,
struct mgcp_rtp_end *dst_end,
char *data, int *len, int buf_size)
{
LOGPENDP(endp, DRTP, LOGL_DEBUG, "transcoding disabled\n");
return 0;
}
/*! dummy callback to disable transcoding (see also cfg->setup_rtp_processing_cb).
* \param[in] associated endpoint.
* \param[in] destination RTP connnection.
* \param[in] source RTP connection.
* \returns ignores input parameters, return always 0. */
int mgcp_setup_rtp_processing_default(struct mgcp_endpoint *endp,
struct mgcp_conn_rtp *conn_dst,
struct mgcp_conn_rtp *conn_src)
{
LOGPENDP(endp, DRTP, LOGL_DEBUG, "transcoding disabled\n");
return 0;
}
void mgcp_get_net_downlink_format_default(struct mgcp_endpoint *endp,
const struct mgcp_rtp_codec **codec,
const char **fmtp_extra,
struct mgcp_conn_rtp *conn)
{
LOGPENDP(endp, DRTP, LOGL_DEBUG, "conn:%s using format defaults\n",
mgcp_conn_dump(conn->conn));
*codec = conn->end.codec;
*fmtp_extra = conn->end.fmtp_extra;
}
void mgcp_rtp_annex_count(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
* that is valid for the destination connection (conn_dst) */
static int mgcp_patch_pt(struct mgcp_conn_rtp *conn_src,
struct mgcp_conn_rtp *conn_dst, struct msgb *msg)
{
struct rtp_hdr *rtp_hdr;
uint8_t pt_in;
int pt_out;
if (msgb_length(msg) < sizeof(struct rtp_hdr)) {
LOG_CONN_RTP(conn_src, LOGL_ERROR, "RTP packet too short (%u < %zu)\n",
msgb_length(msg), sizeof(struct rtp_hdr));
return -EINVAL;
}
rtp_hdr = (struct rtp_hdr *)msgb_data(msg);
pt_in = rtp_hdr->payload_type;
pt_out = mgcp_codec_pt_translate(conn_src, conn_dst, pt_in);
if (pt_out < 0)
return -EINVAL;
rtp_hdr->payload_type = (uint8_t) pt_out;
return 0;
}
/* The RFC 3550 Appendix A assumes there are multiple sources but
* some of the supported endpoints (e.g. the nanoBTS) can only handle
* one source and this code will patch RTP header to appear as if there
* is only one source.
* There is also no probation period for new sources. Every RTP header
* we receive will be seen as a switch in streams. */
void mgcp_patch_and_count(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;
int payload = rtp_end->codec->payload_type;
unsigned int len = msgb_length(msg);
if (len < sizeof(*rtp_hdr))
return;
rtp_hdr = (struct rtp_hdr *)msgb_data(msg);
seq = ntohs(rtp_hdr->sequence);
timestamp = ntohl(rtp_hdr->timestamp);
arrival_time = mgcp_get_current_ts(rtp_end->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->in_stream.last_seq = seq - 1;
state->in_stream.ssrc = state->patch.orig_ssrc = ssrc;
state->in_stream.last_tsdelta = 0;
state->packet_duration =
mgcp_rtp_packet_duration(endp, rtp_end);
state->out_stream.last_seq = seq - 1;
state->out_stream.ssrc = state->patch.orig_ssrc = ssrc;
state->out_stream.last_tsdelta = 0;
state->out_stream.last_timestamp = timestamp;
state->out_stream.ssrc = ssrc - 1; /* force output SSRC change */
LOGPENDP(endp, DRTP, LOGL_INFO,
"initializing stream, SSRC: %u timestamp: %u "
"pkt-duration: %d, from %s:%d\n",
state->in_stream.ssrc,
state->patch.seq_offset, state->packet_duration,
osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
osmo_sockaddr_port(&addr->u.sa));
if (state->packet_duration == 0) {
state->packet_duration =
rtp_end->codec->rate * 20 / 1000;
LOGPENDP(endp, DRTP, LOGL_NOTICE,
"fixed packet duration is not available, "
"using fixed 20ms instead: %d from %s:%d\n",
state->packet_duration,
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 (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, marker_bit);
state->patch.patch_ssrc = true;
ssrc = state->patch.orig_ssrc;
if (rtp_end->force_constant_ssrc != -1)
rtp_end->force_constant_ssrc -= 1;
LOGPENDP(endp, DRTP, LOGL_NOTICE,
"SSRC patching enabled, SSRC: %u "
"SeqNo offset: %d, TS offset: %d "
"from %s:%d\n", state->in_stream.ssrc,
state->patch.seq_offset, state->patch.timestamp_offset,
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;
}
/* For AMR RTP two framing modes are defined RFC3267. There is a bandwith
* efficient encoding scheme where all fields are packed together one after
* another and an octet aligned mode where all fields are aligned to octet
* boundaries. This function is used to convert between the two modes */
static int amr_oa_bwe_convert(struct mgcp_endpoint *endp, struct msgb *msg,
bool target_is_oa)
{
/* NOTE: the msgb has an allocated length of RTP_BUF_SIZE, so there is
* plenty of space available to store the slightly larger, converted
* data */
struct rtp_hdr *rtp_hdr;
unsigned int payload_len;
int rc;
if (msgb_length(msg) < sizeof(struct rtp_hdr)) {
LOGPENDP(endp, DRTP, LOGL_ERROR, "AMR RTP packet too short (%d < %zu)\n", msgb_length(msg), sizeof(struct rtp_hdr));
return -EINVAL;
}
rtp_hdr = (struct rtp_hdr *)msgb_data(msg);
payload_len = msgb_length(msg) - sizeof(struct rtp_hdr);
if (osmo_amr_is_oa(rtp_hdr->data, payload_len)) {
if (!target_is_oa)
/* Input data is oa an target format is bwe
* ==> convert */
rc = osmo_amr_oa_to_bwe(rtp_hdr->data, payload_len);
else
/* Input data is already bew, but we accept it anyway
* ==> no conversion needed */
rc = payload_len;
} else {
if (target_is_oa)
/* Input data is bwe an target format is oa
* ==> convert */
rc = osmo_amr_bwe_to_oa(rtp_hdr->data, payload_len,
RTP_BUF_SIZE);
else
/* Input data is already oa, but we accept it anyway
* ==> no conversion needed */
rc = payload_len;
}
if (rc < 0) {
LOGPENDP(endp, DRTP, LOGL_ERROR,
"AMR RTP packet conversion failed\n");
return -EINVAL;
}
return msgb_trim(msg, rc + sizeof(struct rtp_hdr));
}
/* Check if a conversion between octet-aligned and bandwith-efficient mode is
* indicated. */
static bool amr_oa_bwe_convert_indicated(struct mgcp_rtp_codec *codec)
{
if (codec->param_present == false)
return false;
if (!codec->param.amr_octet_aligned_present)
return false;
if (strcmp(codec->subtype_name, "AMR") != 0)
return false;
return true;
}
/* Return whether an RTP packet with AMR payload is in octet-aligned mode.
* Return 0 if in bandwidth-efficient mode, 1 for octet-aligned mode, and negative if the RTP data is invalid. */
static int amr_oa_check(char *data, int len)
{
struct rtp_hdr *rtp_hdr;
unsigned int payload_len;
if (len < sizeof(struct rtp_hdr))
return -EINVAL;
rtp_hdr = (struct rtp_hdr *)data;
payload_len = len - sizeof(struct rtp_hdr);
if (payload_len < sizeof(struct amr_hdr))
return -EINVAL;
return osmo_amr_is_oa(rtp_hdr->data, payload_len) ? 1 : 0;
}
/* Forward data to a debug tap. This is debug function that is intended for
* debugging the voice traffic with tools like gstreamer */
void forward_data_tap(int fd, struct mgcp_rtp_tap *tap, struct msgb *msg)
{
int rc;
if (!tap->enabled)
return;
rc = sendto(fd, msgb_data(msg), msgb_length(msg), 0, (struct sockaddr *)&tap->forward,
sizeof(tap->forward));
if (rc < 0)
LOGP(DRTP, LOGL_ERROR,
"Forwarding tapped (debug) voice data failed.\n");
}
/* 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->codec->payload_type;
hdr->timestamp = osmo_htonl(mgcp_get_current_ts(rtp_end->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) {
switch (conn->conn->mode) {
case MGCP_CONN_LOOPBACK:
/* HACK: for IuUP, we want to reply with an IuUP Initialization ACK upon the first RTP
* message received. We currently hackishly accomplish that by putting the endpoint in
* loopback mode and patching over the looped back RTP message to make it look like an
* ack. We don't know the femto cell's IP address and port until the RAB Assignment
* Response is received, but the nano3G expects an IuUP Initialization Ack before it even
* sends the RAB Assignment Response. Hence, if the remote address is 0.0.0.0 and the
* MGCP port is in loopback mode, allow looping back the packet to any source. */
LOGPCONN(conn->conn, DRTP, LOGL_ERROR,
"In loopback mode and remote address not set:"
" allowing data from address: %s\n",
osmo_sockaddr_ntop(&addr->u.sa, ipbuf));
return 0;
default:
/* Receiving early media before the endpoint is configured. Instead of logging
* this as an error that occurs on every call, keep it more low profile to not
* confuse humans with expected errors. */
LOGPCONN(conn->conn, DRTP, LOGL_INFO,
"Rx RTP from %s, but remote address not set:"
" dropping early media\n",
osmo_sockaddr_ntop(&addr->u.sa, ipbuf));
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 address: %s, ",
osmo_sockaddr_ntop(&addr->u.sa, ipbuf));
LOGPC(DRTP, LOGL_ERROR, "expected: %s\n",
osmo_sockaddr_ntop(&conn->end.addr.u.sa, ipbuf));
LOGPCONN(conn->conn, DRTP, LOGL_ERROR, "packet tossed\n");
return -1;
}
/* Note: Usually the remote remote port of the data we receive will be
* the same as the remote port where we transmit outgoing RTP traffic
* to (set by MDCX). We use this to check the origin of the data for
* plausibility. */
if (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 source port: %d, ",
osmo_sockaddr_port(&addr->u.sa));
LOGPC(DRTP, LOGL_ERROR,
"expected: %d for RTP or %d for RTCP\n",
osmo_sockaddr_port(&conn->end.addr.u.sa), ntohs(conn->end.rtcp_port));
LOGPCONN(conn->conn, DRTP, LOGL_ERROR, "packet tossed\n");
return -1;
}
return 0;
}
/* Check the if the destination address configuration of an RTP connection
* makes sense */
static int check_rtp_destin(struct mgcp_conn_rtp *conn)
{
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;
}
/* Send RTP data. Possible options are standard RTP packet
* transmission or trsmission via an osmux connection */
static int mgcp_send_rtp(struct mgcp_conn_rtp *conn_dst, struct msgb *msg)
{
struct osmo_rtp_msg_ctx *mc = OSMO_RTP_MSG_CTX(msg);
enum rtp_proto proto = mc->proto;
struct mgcp_conn_rtp *conn_src = mc->conn_src;
struct mgcp_endpoint *endp = conn_src->conn->endp;
LOGPENDP(endp, DRTP, LOGL_DEBUG, "destin conn:%s\n",
mgcp_conn_dump(conn_dst->conn));
/* Before we try to deliver the packet, we check if the destination
* port and IP-Address make sense at all. If not, we will be unable
* to deliver the packet. */
if (check_rtp_destin(conn_dst) != 0)
return -1;
/* Depending on the RTP connection type, deliver the RTP packet to the
* destination connection. */
switch (conn_dst->type) {
case MGCP_RTP_DEFAULT:
LOGPENDP(endp, DRTP, LOGL_DEBUG,
"endpoint type is MGCP_RTP_DEFAULT, "
"using mgcp_send() to forward data directly\n");
return mgcp_send(endp, proto == MGCP_PROTO_RTP,
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 osmux_xfrm_to_osmux((char*)msgb_data(msg), msgb_length(msg), conn_dst);
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");
return -1;
}
/*! send udp packet.
* \param[in] fd 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 bytes sent, -1 on error. */
int mgcp_udp_send(int fd, const struct osmo_sockaddr *addr, const char *buf, int len)
{
char ipbuf[INET6_ADDRSTRLEN];
size_t addr_len;
LOGP(DRTP, LOGL_DEBUG,
"sending %i bytes length packet to %s:%u ...\n", len,
osmo_sockaddr_ntop(&addr->u.sa, ipbuf),
osmo_sockaddr_port(&addr->u.sa));
if (addr->u.sa.sa_family == AF_INET6) {
addr_len = sizeof(addr->u.sin6);
} else {
addr_len = sizeof(addr->u.sin);
}
return sendto(fd, buf, len, 0, &addr->u.sa, addr_len);
}
/*! 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)
{
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.fd, &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.fd, &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.
* \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 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");
/* 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 && conn_dst->type != MGCP_RTP_IUUP) {
rc = mgcp_patch_pt(conn_src, conn_dst, msg);
if (rc < 0) {
LOGPENDP(endp, DRTP, LOGL_DEBUG,
"can not patch PT because no suitable egress codec was found.\n");
}
}
/* Note: In case of loopback configuration, both, the source and the
* destination will point to the same connection. */
rtp_end = &conn_dst->end;
rtp_state = &conn_src->state;
dest_name = conn_dst->conn->name;
/* 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) {
int cont;
int nbytes = 0;
int buflen = msgb_length(msg);
/* 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);
do {
/* Run transcoder */
cont = endp->trunk->cfg->rtp_processing_cb(endp, rtp_end, (char *)msgb_data(msg), &buflen, RTP_BUF_SIZE);
if (cont < 0)
break;
if (addr)
mgcp_patch_and_count(endp, rtp_state, rtp_end,
addr, msg);
if (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 (amr_oa_bwe_convert_indicated(conn_dst->end.codec)) {
rc = amr_oa_bwe_convert(endp, msg,
conn_dst->end.codec->param.amr_octet_aligned);
if (rc < 0) {
LOGPENDP(endp, DRTP, LOGL_ERROR,
"Error in AMR octet-aligned <-> bandwidth-efficient mode conversion\n");
break;
}
} else if (rtp_end->rfc5993_hr_convert &&
strcmp(conn_src->end.codec->subtype_name, "GSM-HR-08") == 0) {
rc = rfc5993_hr_convert(endp, msg);
if (rc < 0) {
LOGPENDP(endp, DRTP, LOGL_ERROR, "Error while converting to GSM-HR-08\n");
break;
}
}
LOGPENDP(endp, DRTP, LOGL_DEBUG,
"process/send to %s %s "
"rtp_port:%u rtcp_port:%u\n",
dest_name,
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.fd, &conn_src->tap_out,
msg);
len = mgcp_udp_send(rtp_end->rtp.fd, &rtp_end->addr,
(char *)msgb_data(msg), msgb_length(msg));
if (len <= 0)
return len;
rtpconn_rate_ctr_inc(conn_dst, endp, RTP_PACKETS_TX_CTR);
rtpconn_rate_ctr_add(conn_dst, endp, RTP_OCTETS_TX_CTR, len);
rtp_state->alt_rtp_tx_sequence++;
nbytes += len;
buflen = cont;
} while (buflen > 0);
return nbytes;
} else if (!trunk->omit_rtcp) {
struct osmo_sockaddr rtcp_addr = rtp_end->addr;
osmo_sockaddr_set_port(&rtcp_addr.u.sa, 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 = mgcp_udp_send(rtp_end->rtcp.fd, &rtcp_addr,
(char *)msgb_data(msg), msgb_length(msg));
rtpconn_rate_ctr_inc(conn_dst, endp, RTP_PACKETS_TX_CTR);
rtpconn_rate_ctr_add(conn_dst, endp, RTP_OCTETS_TX_CTR, len);
rtp_state->alt_rtp_tx_sequence++;
return len;
}
return 0;
}
/*! 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 osmo_sockaddr *from_addr = mc->from_addr;
char ipbuf[INET6_ADDRSTRLEN];
/*! 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->u.rtp.end.addr.u.sa) == 0) {
memcpy(&conn->u.rtp.end.addr, from_addr,
sizeof(conn->u.rtp.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->u.rtp.end.addr.u.sa));
}
return mgcp_send_rtp(conn_src, msg);
}
/* Find a destination connection. */
/* NOTE: This code path runs every time an RTP packet is received. The
* function mgcp_find_dst_conn() we use to determine the detination
* connection will iterate the connection list inside the endpoint.
* Since list iterations are quite costly, we will figure out the
* destination only once and use the optional private data pointer of
* the connection to cache the destination connection pointer. */
if (!conn->priv) {
conn_dst = mgcp_find_dst_conn(conn);
conn->priv = conn_dst;
} else {
conn_dst = (struct mgcp_conn *)conn->priv;
}
/* There is no destination conn, stop here */
if (!conn_dst) {
LOGPCONN(conn, DRTP, LOGL_DEBUG,
"no connection to forward an incoming RTP packet to\n");
return -1;
}
/* The destination conn is not an RTP connection */
if (conn_dst->type != MGCP_CONN_TYPE_RTP) {
LOGPCONN(conn, DRTP, LOGL_ERROR,
"unable to find suitable destination conn\n");
return -1;
}
/* Dispatch RTP packet to destination RTP connection */
return mgcp_send_rtp(&conn_dst->u.rtp, msg);
}
/*! dispatch incoming RTP packet to E1 subslot, handle RTCP packets locally.
* \param[in] proto protocol (MGCP_CONN_TYPE_RTP or MGCP_CONN_TYPE_RTCP).
* \param[in] addr socket address where the RTP packet has been received from.
* \param[in] buf buffer that hold the RTP payload.
* \param[in] buf_size size data length of buf.
* \param[in] conn originating connection.
* \returns 0 on success, -1 on ERROR. */
int mgcp_dispatch_e1_bridge_cb(struct msgb *msg)
{
struct osmo_rtp_msg_ctx *mc = OSMO_RTP_MSG_CTX(msg);
struct mgcp_conn_rtp *conn_src = mc->conn_src;
struct mgcp_conn *conn = conn_src->conn;
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->u.rtp.end.addr.u.sa) == 0) {
memcpy(&conn->u.rtp.end.addr, from_addr,
sizeof(conn->u.rtp.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->u.rtp.end.addr.u.sa));
}
return mgcp_send_rtp(conn_src, msg);
}
/* Forward to E1 */
return mgcp_e1_send_rtp(conn->endp, conn->u.rtp.end.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 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 osmo_sockaddr addr;
socklen_t slen = sizeof(addr);
char ipbuf[INET6_ADDRSTRLEN];
int ret;
enum rtp_proto proto;
struct osmo_rtp_msg_ctx *mc;
struct msgb *msg;
int rc;
conn_src = (struct mgcp_conn_rtp *)fd->data;
OSMO_ASSERT(conn_src);
endp = conn_src->conn->endp;
OSMO_ASSERT(endp);
msg = msgb_alloc_c(endp->trunk, RTP_BUF_SIZE, "RTP-rx");
proto = (fd == &conn_src->end.rtp)? MGCP_PROTO_RTP : MGCP_PROTO_RTCP;
ret = recvfrom(fd->fd, msgb_data(msg), msg->data_len, 0, (struct sockaddr *)&addr.u.sa, &slen);
if (ret <= 0) {
LOG_CONN_RTP(conn_src, LOGL_ERROR, "recvfrom error: %s\n", strerror(errno));
rc = -1;
goto out;
}
msgb_put(msg, ret);
LOG_CONN_RTP(conn_src, LOGL_DEBUG, "%s: rx %u bytes from %s:%u\n",
proto == MGCP_PROTO_RTP ? "RTP" : "RTCP",
msgb_length(msg), osmo_sockaddr_ntop(&addr.u.sa, ipbuf),
osmo_sockaddr_port(&addr.u.sa));
if ((proto == MGCP_PROTO_RTP && check_rtp(conn_src, msg))
|| (proto == MGCP_PROTO_RTCP && check_rtcp(conn_src, msg))) {
/* Logging happened in the two check_ functions */
rc = -1;
goto out;
}
if (mgcp_is_rtp_dummy_payload(msg)) {
LOG_CONN_RTP(conn_src, LOGL_DEBUG, "rx dummy packet (dropped)\n");
rc = 0;
goto out;
}
/* Since the msgb remains owned and freed by this function, the msg ctx data struct can just be on the stack and
* needs not be allocated with the msgb. */
mc = OSMO_RTP_MSG_CTX(msg);
*mc = (struct osmo_rtp_msg_ctx){
.proto = proto,
.conn_src = conn_src,
.from_addr = &addr,
};
LOG_CONN_RTP(conn_src, LOGL_DEBUG, "msg ctx: %d %p %s\n",
mc->proto, mc->conn_src,
osmo_hexdump((void*)mc->from_addr,
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(fd->fd, &conn_src->tap_in, msg);
rc = rx_rtp(msg);
out:
msgb_free(msg);
return rc;
}
/* Note: This function is able to handle RTP and RTCP */
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));
mgcp_conn_watchdog_kick(conn_src->conn);
/* If AMR is configured for the ingress connection a conversion of the
* framing mode (octet-aligned vs. bandwith-efficient is explicitly
* define, then we check if the incoming payload matches that
* expectation. */
if (mc->proto == MGCP_PROTO_RTP &&
amr_oa_bwe_convert_indicated(conn_src->end.codec)) {
int oa = amr_oa_check((char*)msgb_data(msg), msgb_length(msg));
if (oa < 0)
return -1;
if (((bool)oa) != conn_src->end.codec->param.amr_octet_aligned)
return -1;
}
/* Check if the origin of the RTP packet seems plausible */
if (!trunk->rtp_accept_all && check_rtp_origin(conn_src, from_addr))
return -1;
/* Execute endpoint specific implementation that handles the
* dispatching of the RTP data */
return conn->endp->type->dispatch_rtp_cb(msg);
}
/*! 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.
* \param[in] dscp IP DSCP value to use.
* \param[in] prio socket priority to use.
* \returns 0 on success, -1 on ERROR. */
int mgcp_create_bind(const char *source_addr, struct osmo_fd *fd, 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:%i).\n",
source_addr, port);
return -1;
}
fd->fd = rc;
LOGP(DRTP, LOGL_DEBUG, "created socket + bound UDP port (%s:%i).\n", source_addr, port);
return 0;
}
/* Bind RTP and RTCP port (helper function for mgcp_bind_net_rtp_port()) */
static int bind_rtp(struct mgcp_config *cfg, const char *source_addr,
struct mgcp_rtp_end *rtp_end, struct mgcp_endpoint *endp)
{
/* NOTE: The port that is used for RTCP is the RTP port incremented by one
* (e.g. RTP-Port = 16000 ==> RTCP-Port = 16001) */
if (mgcp_create_bind(source_addr, &rtp_end->rtp, rtp_end->local_port,
cfg->endp_dscp, cfg->endp_priority) != 0) {
LOGPENDP(endp, DRTP, LOGL_ERROR,
"failed to create RTP port: %s:%d\n",
source_addr, rtp_end->local_port);
goto cleanup0;
}
if (mgcp_create_bind(source_addr, &rtp_end->rtcp, rtp_end->local_port + 1,
cfg->endp_dscp, cfg->endp_priority) != 0) {
LOGPENDP(endp, DRTP, LOGL_ERROR,
"failed to create RTCP port: %s:%d\n",
source_addr, rtp_end->local_port + 1);
goto cleanup1;
}
if (osmo_fd_register(&rtp_end->rtp) != 0) {
LOGPENDP(endp, DRTP, LOGL_ERROR,
"failed to register RTP port %d\n",
rtp_end->local_port);
goto cleanup2;
}
if (osmo_fd_register(&rtp_end->rtcp) != 0) {
LOGPENDP(endp, DRTP, LOGL_ERROR,
"failed to register RTCP port %d\n",
rtp_end->local_port + 1);
goto cleanup3;
}
return 0;
cleanup3:
osmo_fd_unregister(&rtp_end->rtp);
cleanup2:
close(rtp_end->rtcp.fd);
rtp_end->rtcp.fd = -1;
cleanup1:
close(rtp_end->rtp.fd);
rtp_end->rtp.fd = -1;
cleanup0:
return -1;
}
/*! bind RTP port to endpoint/connection.
* \param[in] endp endpoint that holds the RTP connection.
* \param[in] rtp_port port number to bind on.
* \param[in] conn associated RTP connection.
* \returns 0 on success, -1 on ERROR. */
int mgcp_bind_net_rtp_port(struct mgcp_endpoint *endp, int rtp_port,
struct mgcp_conn_rtp *conn)
{
char name[512];
struct mgcp_rtp_end *end;
snprintf(name, sizeof(name), "%s-%s", conn->conn->name, conn->conn->id);
end = &conn->end;
if (end->rtp.fd != -1 || end->rtcp.fd != -1) {
LOGPENDP(endp, DRTP, LOGL_ERROR, "%u was already bound on conn:%s\n",
rtp_port, mgcp_conn_dump(conn->conn));
/* Double bindings should never occour! Since we always allocate
* connections dynamically and free them when they are not
* needed anymore, there must be no previous binding leftover.
* Should there be a connection bound twice, we have a serious
* problem and must exit immediately! */
OSMO_ASSERT(false);
}
end->local_port = rtp_port;
osmo_fd_setup(&end->rtp, -1, OSMO_FD_READ, rtp_data_net, conn, 0);
osmo_fd_setup(&end->rtcp, -1, OSMO_FD_READ, rtp_data_net, conn, 0);
return bind_rtp(endp->trunk->cfg, conn->end.local_addr, end, endp);
}
/*! free allocated RTP and RTCP ports.
* \param[in] end RTP end */
void mgcp_free_rtp_port(struct mgcp_rtp_end *end)
{
if (end->rtp.fd != -1) {
close(end->rtp.fd);
end->rtp.fd = -1;
osmo_fd_unregister(&end->rtp);
}
if (end->rtcp.fd != -1) {
close(end->rtcp.fd);
end->rtcp.fd = -1;
osmo_fd_unregister(&end->rtcp);
}
}
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