Provided by: rtpengine-daemon_11.5.1.18-1build5_amd64 
NAME
rtpengine - NGCP proxy for RTP and other UDP based media traffic
SYNOPSIS
rtpengine --interface=addr... --listen-tcp|--listen-udp|--listen-ng|--listen-tcp-
ng|--listen-http|--listen-https=addr... [option...]
DESCRIPTION
The Sipwise NGCP rtpengine is a proxy for RTP traffic and other UDP based media traffic.
It is meant to be used with the Kamailio SIP proxy and forms a drop-in replacement for any
of the other available RTP and media proxies.
OPTIONS
Most of these options are indeed optional, with two exceptions. It’s mandatory to specify
at least one local IP address through --interface, and at least one of the --listen-...
options must be given.
All options can (and should) be provided in a config file instead of at the command line.
See the --config-file option below for details.
• --help
Print the usage information.
• -v, --version
If called with this option, the rtpengine daemon will simply print its version number
and exit.
• --codecs
Print a list of supported codecs and exit.
• --config-file=FILE
Specifies the location of a config file to be used. The config file is an .ini style
config file, with all command-line options listed here also being valid options in the
config file. For all command-line options, the long name version instead of the single-
character version (e.g. table instead of just t) must be used in the config file. For
boolean options that are either present or not (e.g. no-fallback), a boolean value
(either true or false) must be used in the config file. If an option is given in both
the config file and at the command line, the command-line value overrides the value from
the config file. Options that can be specified multiple times on the command line must
be given only once in the config file, with the multiple values separated by semicolons
(see section INTERFACES (https://metacpan.org/pod/INTERFACES) below for an example).
As a special value, none can be passed here to suppress loading of the default config
file /etc/rtpengine/rtpengine.conf.
• --config-section=STRING
Specifies the .ini style section to be used in the config file. Multiple sections can
be present in the config file, but only one can be used at a time. The default value is
rtpengine. A config file section is started in the config file using square brackets
(e.g. [rtpengine]).
• -t, --table=INT
Takes an integer argument and specifies which kernel table to use for in-kernel packet
forwarding. See the section on in-kernel operation in the README.md for more detail.
Optional and defaults to zero. If in-kernel operation is not desired, a negative number
can be specified.
• -F, --no-fallback
Will prevent fallback to userspace-only operation if the kernel module is unavailable.
In this case, startup of the daemon will fail with an error if this option is given.
• -S, --save-interface-ports
Will bind ports only on the first available local interface, of desired family, of
logical interface. If no ports available on any local interface of desired family, give
an error message.
In this case, ICE will be broken.
• -i, --interface=[NAME/]IP[!IP]
Specifies a local network interface for RTP. At least one must be given, but multiple
can be specified. See the section INTERFACES (https://metacpan.org/pod/INTERFACES) just
below for details.
• -l, --listen-tcp=[IP:]PORT
• -u, --listen-udp=[IP46:]PORT
• -n, --listen-ng=[IP46:]PORT
• -n, --listen-tcp-ng=[IP46:]PORT
These options each enable one of the 4 available control protocols if given and each
take either just a port number as argument, or an address:port pair, separated by colon.
At least one of these 3 options must be given.
The tcp protocol is obsolete. It was used by old versions of OpenSER and its mediaproxy
module. It is provided for backwards compatibility.
The udp protocol is used by Kamailio’s rtpproxy module. In this mode, rtpengine can be
used as a drop-in replacement for any other compatible RTP proxy.
The ng protocol is an advanced control protocol and can be used with Kamailio’s
rtpengine module. With this protocol, the complete SDP body is passed to rtpengine,
rewritten and passed back to Kamailio. Several additional features are available with
this protocol, such as ICE handling, SRTP bridging, etc.
The tcp-ng protocol is in fact the ng protocol but transported over TCP.
It is recommended to specify not only a local port number, but also 127.0.0.1 as
interface to bind to.
Each option can be given multiple times to open multiple control ports of the same type.
In the config file, the option can be given only once, with multiple addresses and ports
separated by semicolons.
• -c, --listen-cli=[IP46:]PORT
TCP IP and port to listen for the CLI (command line interface).
This option can be given multiple times to open multiple CLI ports. In the config file,
the option can be given only once, with multiple addresses and ports separated by
semicolons.
• -g, --graphite=IP46:PORT
Address of the graphite statistics server.
• -w, --graphite-interval=INT
Interval of the time when information is sent to the graphite server.
• --graphite-prefix=STRING
Add a prefix for every graphite line.
• --graphite-timeout=INT
Sets after how much time (seconds) to force fail graphite socket connection, when
graphite server is filtered out. If set to 0, there are no changes.
• -t, --tos=INT
Takes an integer as argument and if given, specifies the TOS value that should be set in
outgoing packets. The default is to leave the TOS field untouched. A typical value is
184 (Expedited Forwarding).
• --control-tos=INT
Takes an integer as argument and if given, specifies the TOS value that should be set in
the control-ng interface packets. The default is to leave the TOS field untouched.
This parameter can also be set or listed via rtpengine-ctl.
• --control-pmtu=want|dont
Forces a specific PMTU discovery behaviour on IPv4 UDP control sockets, overriding the
system-wide default. If set to want then path MTU discovery is performed, initially
enabling the DF (don’t fragment) bit on outgoing IPv4 packets until the path MTU has
been discovered through reception of a “fragmentation needed” ICMP packet. If set to
dont then path MTU discovery is disabled, leaving the DF bit unset, and relying on the
routers within the network path to perform any necessary fragmentation.
The setting of dont is useful in broken IPv4 environments without functioning PMTU
discovery, for example in networks which unconditionally block all ICMP.
• -o, --timeout=SECS
Takes the number of seconds as argument after which a media stream should be considered
dead if no media traffic has been received. If all media streams belonging to a
particular call go dead, then the call is removed from rtpengine’s internal state table.
Defaults to 60 seconds.
• -s, --silent-timeout=SECS
Ditto as the --timeout option, but applies to muted or inactive media streams. Defaults
to 3600 (one hour).
• -a, --final-timeout=SECS
The number of seconds since call creation, after call is deleted. Useful for limiting
the lifetime of a call. This feature can be disabled by setting the parameter to 0. By
default this timeout is disabled.
• --offer-timeout=SECS
This timeout (in seconds) is applied to calls which only had an offer but no answer.
Defaults to 3600 (one hour).
• -p, --pidfile=FILE
Specifies a path and file name to write the daemon’s PID number to.
• -f, --foreground
If given, prevents the daemon from daemonizing, meaning it will stay in the foreground.
Useful for debugging.
• -m, --port-min=INT
• -M, --port-max=INT
Both take an integer as argument and together define the local port range from which
rtpengine will allocate UDP ports for media traffic relay. Default to 30000 and 40000
respectively.
• -L, --log-level=INT
Takes an integer as argument and controls the highest log level which will be sent to
syslog. This is merely the default log level used for logging subsystems (see below)
that don’t explicitly have a separate log level configured.
The log levels correspond to the ones found in the syslog(3)
(http://man.he.net/man3/syslog) man page. The default value is 6, equivalent to
LOG_INFO. The highest possible value is 7 (LOG_DEBUG) which will log everything.
During runtime, the log level can be decreased by sending the signal SIGURS1 to the
daemon and can be increased with the signal SIGUSR2.
• --log-level-subsystem=INT
Configures a log level for one of the logging subsystems. A logging subsystem which
doesn’t have a log level configured explicitly takes its default value from the log-
level setting described above, with the exception of the internals subsystem which by
default has all logging disabled.
The full list of logging subsystems can be viewed by pulling up the --help online help.
Some (if not all) subsystems are: core, spandsp (messages generated by SpanDSP itself),
ffmpeg (messages generated by ffmpeg libraries themselves), transcoding (messages
related to RTP/media transcoding), codec (messages related to codec negotiation), rtcp,
ice, crypto (messages related to crypto/SRTP/SDES/DTLS negotiation), srtp (messages
related to RTP/SRTP en/decryption), internals (disabled by default), http (includes
WebSocket), control (messages related to control protocols, including SDP exchanges),
dtx.
• --log-facilty=daemon|local0|...|local7|...
The syslog facilty to use when sending log messages to the syslog daemon. Defaults to
daemon.
• --log-facilty-cdr=daemon|local0|...|local7|...
Same as --log-facility with the difference that only CDRs are written to this log
facility.
• --log-facilty-rtcp=daemon|local0|...|local7|...
Same as --log-facility with the difference that only RTCP data is written to this log
facility. Be careful with this parameter since there may be a lot of information
written to it.
• --log-facilty-dtmf=daemon|local0|...|local7|...
Same as --log-facility with the difference that only DTMF events are written to this log
facility. DTMF events are extracted from RTP packets conforming to RFC 4733, are
encoded in JSON format, and written as soon as the end of an event is detected.
• --log-format=default|parsable
Selects between multiple log output styles. The default is to prefix log lines with a
description of the relevant entity, such as [CALLID] or [CALLID port 12345]. The
parsable output style is similar, but makes the ID easier to parse by enclosing it in
quotes, such as [ID=“CALLID”] or [ID=“CALLID” port=“12345”].
• --dtmf-log-dest=IP46:PORT
Configures a target address for logging detected DTMF event. Similar to the feature
enabled by --log-facilty-dtmf, but instead of writing detected DTMF events to syslog,
this sends the JSON payload to the given address as UDP packets.
• --dtmf-log-ng-tcp
If --listen-tcp-ng is enabled, this will send DTMF events to all connected clients
encoded in bencode format.
• --dtmf-no-log-injects If --dtmf-no-log-injects is enabled, DTMF events resulting from a
call to inject-DTMF won’t be sent to --dtmf-log-dest= or --listen-tcp-ng
• --dtmf-no-suppress
Some RTP clients continue to send audio RTP packets during a DTMF event, resulting in
both audio packets and DTMF packets appearing simultaneously. By default, when
transcoding, rtpengine suppresses audio packets during a DTMF event and will only send
DTMF packets until the DTMF event is over. Setting this option disables this feature.
• --log-srtp-keys
Write SRTP keys to error log instead of debug log.
• -E, --log-stderr
Log to stderr instead of syslog. Only useful in combination with --foreground.
• --split-logs
Split multi-line log messages into individual log messages so that each line receives
its own log line prefix.
• --max-log-line-length=INT
Split log lines into multiple lines when they exceed the character count given here.
Can be set to a negative value to allow unlimited length log lines. Set to zero for the
default value, which is unlimited if logging to stderr, or 500 if logging to syslog.
• --no-log-timestamps
Don’t add timestamps to log lines written to stderr. Only useful in combination with
--log-stderr.
• --log-name=STRING
Set the id to be printed in syslog. Defaults to rtpengine.
• --log-mark-prefix=STRING
Prefix to be added to particular data fields in log files that are deemed sensitive
and/or private information. Defaults to an empty string.
• --log-mark-suffix=STRING
Suffix to be added to particular data fields in log files that are deemed sensitive
and/or private information. Defaults to an empty string.
• --num-threads=INT
How many worker threads to create, must be at least one. The default is to create as
many threads as there are CPU cores available. If the number of CPU cores cannot be
determined or if it is less than four, then the default is four.
• --media-num-threads=INT
Number of threads to launch for media playback. Defaults to the same number as num-
threads. This can be set to zero if no media playback functionality is desired.
Media playback is actually handled by two threads: One for reading and decoding the
media file, and another to schedule and send out RTP packets. So for example, if this
option is set to 4, in total 8 threads will be launched.
• --poller-size=INT
Set the maximum number of event items (file descriptors) to retrieve from the underlying
system poll mechanism per iteration. Defaults to 128. A lower number can lead to
improved load-balancing among a large number of threads.
• --thread-stack=INT
Set the stack size of each thread to the value given in kB. Defaults to 2048 kB. Can
be set to -1 to leave the default provided by the OS unchanged.
• --evs-lib-path=FILE
Points to the shared object file (.so) containing the reference implementation for the
EVS codec. See the README for more details.
• --sip-source
The original rtpproxy as well as older version of rtpengine by default did not honour IP
addresses given in the SDP body, and instead used the source address of the received SIP
message as default endpoint address. Newer versions of rtpengine reverse this behaviour
and honour the addresses given in the SDP body by default. This option restores the old
behaviour.
• --dtls-passive
Enables the DTLS=passive flag for all calls unconditionally.
• -d, --delete-delay=INT
Delete the call after the specified delay from memory. Can be set to zero for immediate
call deletion.
• -r, --redis=[PW@]IP:PORT/INT
Connect to specified Redis database (with the given database number) and use it for
persistence storage. The format of this option is ADDRESS:PORT/DBNUM, for example
127.0.0.1:6379/12 to connect to the Redis DB number 12 running on localhost on the
default Redis port.
If the Redis database is protected with an authentication password, the password can be
supplied by prefixing the argument value with the password, separated by an @ symbol,
for example foobar@127.0.0.1:6379/12. Note that this leaves the password visible in the
process list, posing a security risk if untrusted users access the same system. As an
alternative, the password can also be supplied in the shell environment through the
environment variable RTPENGINE*REDIS*AUTH*PW.
On startup, rtpengine will read the contents of this database and restore all calls
stored therein. During runtime operation, rtpengine will continually update the
database’s contents to keep it current, so that in case of a service disruption, the
last state can be restored upon a restart.
When this option is given, rtpengine will delay startup until the Redis database adopts
the master role (but see below).
• -w, --redis-write=[PW@]IP:PORT/INT
Configures a second Redis database for write operations. If this option is given in
addition to the first one, then the first database will be used for read operations
(i.e. to restore calls from) while the second one will be used for write operations (to
update states in the database).
For password protected Redis servers, the environment variable for the password is
RTPENGINE*REDIS*WRITE*AUTH*PW.
When both options are given, rtpengine will start and use the Redis database regardless
of the database’s role (master or slave).
• -k, --subscribe-keyspace=INT
List of redis keyspaces to subscribe. If this is not present, no keyspaces are
subscribed (default behaviour). Further subscriptions could be added/removed via
rtpengine-ctl ksadd/ksrm. This may lead to enabling/disabling of the redis keyspace
notification feature.
• --redis-num-threads=INT
How many redis restore threads to create. The default is 4.
• --redis-expires=INT
Expire time in seconds for redis keys. Default is 86400.
• --active-switchover
With this option enabled, any activity (such as signalling or media) on a call that was
created through a Redis keyspace notification will make rtpengine take control of that
call. Without this option, an explicit command is required for rtpengine to take (or
relinquish) control of a call.
• -q, --no-redis-required
When this parameter is present or NO*REDIS*REQUIRED=`yes' or `1' in the config file,
rtpengine starts even if there is no initial connection to redis databases (either to -r
or to -w or to both redis).
Be aware that if the -r redis cannot be initially connected, sessions are not reloaded
upon rtpengine startup, even though rtpengine still starts.
• --redis-allowed-errors
If this parameter is present and has a value >= 0, it will configure how many
consecutive errors are allowed when communicating with a redis server before the redis
communication will be temporarily disabled for that server. While the communication is
disabled there will be no attempts to reconnect to redis or send commands to that
server. Default value is -1, meaning that this feature is disabled. This parameter can
also be set or listed via rtpengine-ctl.
• --redis-disable-time
This parameter configures the number of seconds redis communication is disabled because
of errors. This works together with redis-allowed-errors parameter. The default value
is 10. This parameter can also be set or listed via rtpengine-ctl.
• --redis-cmd-timeout=INT
If this parameter is set to a non-zero value it will set the timeout, in milliseconds,
for each command to the redis server. If redis does not reply within the specified
timeout the command will fail. The default value is 0, meaning that the commands will
be blocking without timeout. This parameter can also be set or listed via rtpengine-
ctl; note that setting the parameter to 0 will require a reconnect on all configured
redis servers.
• --redis-connect-timeout=INT
This parameter sets the timeout value, in milliseconds, when connecting to a redis
server. If the connection cannot be made within the specified timeout the connection
will fail. Note that in case of failure, when reconnecting to redis, a PING command is
issued before attempting to connect so the --redis-cmd-timeout value will also be added
to the total waiting time. This is useful if using --redis-allowed-errors, when
attempting to estimate the total lost time in case of redis failures. The default value
for the connection timeout is 1000ms. This parameter can also be set or listed via
rtpengine-ctl.
• -b, --b2b-url=STRING
Enables and sets the URI for an XMLRPC callback to be made when a call is torn down due
to packet timeout. The special code %% can be used in place of an IP address, in which
case the source address of the originating request (or alternatively the address
specified using the xmlrpc-callback ng protocol option) will be used.
• -x, --xmlrpc-format=INT
Selects the internal format of the XMLRPC callback message for B2BUA call teardown. 0
is for SEMS, 1 is for a generic format containing the call-ID only, 2 is for Kamailio.
• --max-sessions=INT
Limit the number of maximum concurrent sessions. Set at startup via max-sessions in
config file. Set at runtime via rtpengine-ctl util. Setting the rtpengine-ctl set
maxsessions 0 can be used in draining rtpengine sessions. Enable feature: max-
sessions=1000 Enable feature: rtpengine-ctl set maxsessions >= 0 Disable feature:
rtpengine-ctl set maxsessions -1 By default, the feature is disabled (i.e. maxsessions
== -1).
• --max-load=FLOAT
If the current 1-minute load average exceeds the value given here, reject new sessions
until the load average drops below the threshold.
• --max-cpu=FLOAT
If the current CPU usage (in percent) exceeds the value given here, reject new sessions
until the CPU usage drops below the threshold. CPU usage is sampled in 0.5-second
intervals. Only supported on systems providing a Linux-style /proc/stat.
• --max-bandwidth=INT
If the current bandwidth usage (in bytes per second) exceeds the value given here,
reject new sessions until the bandwidth usage drops below the threshold. Bandwidth
usage is sampled in 1-second intervals and is based on received packets, not sent
packets.
• --homer=IP46:PORT
Enables sending the decoded contents of RTCP packets to a Homer SIP capture server. The
transport is HEP version 3 and payload format is JSON. This argument takes an IP
address and a port number as value.
• --homer-protocol=udp|tcp
Can be either udp or tcp with udp being the default.
• --homer-id=INT
The HEP protocol used by Homer contains a “capture ID” used to distinguish different
sources of capture data. This ID can be specified using this argument.
• --recording-dir=FILE
An optional argument to specify a path to a directory where PCAP recording files and
recording metadata files should be stored. If not specified, support for call recording
will be disabled.
rtpengine supports multiple mechanisms for recording calls. See recording-method below
for a list. The default recording method pcap is described in this section.
PCAP files will be stored within a pcap subdirectory and metadata within a metadata
subdirectory.
The format for a metadata file is (with a trailing newline):
/path/to/recording-pcap.pcap
SDP mode: offer
SDP before RTP packet: 1
first SDP
SDP mode: answer
SDP before RTP packet: 1
second SDP
...
SDP mode: answer
SDP before RTP packet: 100
n-th and final SDP
start timestamp (YYYY-MM-DDThh:mm:ss)
end timestamp (YYYY-MM-DDThh:mm:ss)
generic metadata
There are two empty lines between each logic block of metadata. We write out all answer
SDP, each separated from one another by one empty line. The generic metadata at the end
can be any length with any number of lines. Metadata files will appear in the
subdirectory when the call completes. PCAP files will be written to the subdirectory as
the call is being recorded.
Since call recording via this method happens entirely in userspace, in-kernel packet
forwarding cannot be used for calls that are currently being recorded and packet
forwarding will thus be done in userspace only.
• --recording-method=pcap|proc|all
Multiple methods of call recording are supported and this option can be used to select
one. Currently supported are the method pcap, proc and all. The default method is pcap
and is the one described above.
The recording method proc works by writing metadata files directly into the recording-
dir (i.e. not into a subdirectory) and instead of recording RTP packet data into pcap
files, the packet data is exposed via a special interface in the /proc filesystem.
Packets must then be retrieved from this interface by a dedicated userspace component
(usually a daemon such as recording-daemon included in this repository).
Packet data is held in kernel memory until retrieved by the userspace component, but
only a limited number of packets (default 10) per media stream. If packets are not
retrieved in time, they will be simply discarded. This makes it possible to flag all
calls to be recorded and then leave it to the userspace component to decided whether to
use the packet data for any purpose or not.
In-kernel packet forwarding is fully supported with this recording method even for calls
being recorded.
The recording method all will enable both pcap and proc at the same time.
• --recording-format=raw|eth
When recording to pcap file in raw (default) format, there is no ethernet header. When
set to eth, a fake ethernet header is added, making each package 14 bytes larger.
• --record-egress
Apply media recording to egress media streams (as they are sent by rtpengine) instead of
media streams as they are received. This makes it possible to include manipulated and
generated media (such as from the play media command) in the recordings.
• --iptables-chain=STRING
This option enables explicit management of an iptables chain. When enabled, rtpengine
takes control of the given iptables chain, which must exist already prior to starting
the daemon. Upon startup, rtpengine will flush the chain, and then add one ACCEPT rule
for each media port (RTP/RTCP) opened. Each rule will exactly match the individual port
and destination IP address, and will be created with the call ID as iptables comment.
The rule will be deleted when the port is closed.
This option allows creating a firewall with a default DROP policy for the entire port
range used by rtpengine and then referencing the given iptables chain to only
selectively allow the ports actually in use.
Note that this applies only to media ports, and does not apply to any other ports (such
as the control ports) used by rtpengine.
Also note that the iptables API is not the most efficient one around and does not lend
itself to fast dynamic creation and deletion of rules. If you have a high call volume,
and especially many call attempts per second, you might experience significant
performance impact. This is not a shortcoming of rtpengine but rather of iptables and
its API implementation in the Linux kernel. In such a case, it is recommended to add a
static iptables rule for the entire media port range instead, and not use this option.
• --scheduling=default|...
• --priority=INT
• --idle-scheduling=default|...
• --idle-priority=INT
These options control various thread scheduling parameters. The scheduling and priority
settings are applied to the main worker threads, while the idle- versions of these
settings are applied to various lower priority threads, such as timer runs.
The scheduling settings take the name of one of the supported scheduler policies.
Setting it to default or none is equivalent to not setting the option at all and leaves
the system default in place. The strings fifo and rr refer to realtime scheduling
policies. other is the Linux default scheduling policy. batch is similar to other
except for a small wake-up scheduling penalty. idle is an extremely low priority
scheduling policy. The Linux-specific deadline policy is not supported by rtpengine.
Not all systems necessarily supports all scheduling policies; refer to your system’s
sched(7) man page for details.
The priority settings correspond to the scheduling priority for realtime (fifo or rr)
scheduling policies and must be in the range of 1 (low) through 99 (high). For all
other scheduling policies (including no policy specified), the priority settings
correspond to the nice value and should be in the range of -20 (high) through 19 (low).
Not all systems support thread-specific nice values; on such a system, using these
settings might have unexpected results. (Linux does support thread-specific nice
values.) Refer to your system’s sched(7) man page.
• --mysql-host=HOST|IP
• --mysql-port=INT
• --mysql-user=USERNAME
• --mysql-pass=PASSWORD
Configuration for playing back media files that are stored in a MySQL (or MariaDB)
database. At least mysql-host must be configured for this to work. The others are
optional and default to their respective values from the MySQL/MariaDB client library.
• --mysql-query=STRING
Query to be used for retrieving media files from the database. No default exist,
therefore this is a mandatory configuration for media playback from database. The
provided query string must contain the single format placeholder %llu and must not
contain any other format placeholders. The ID value passed to rtpengine in the db-id
key of the play media message will be used in place of the placeholder when querying the
database.
An example configuration might look like this:
mysql-query = select data from voip.files where id = %llu
• --endpoint-learning=delayed|immediate|off|heuristic
Chooses one of the available algorithms to learn RTP endpoint addresses. The legacy
setting is delayed which waits 3 seconds before committing to an endpoint address, which
is then learned from the first incoming RTP packet seen after this delay. The setting
immediate learns the endpoint address from the first incoming packet seen without the
3-second delay. Using off disables endpoint learning altogether, likely breaking
clients behind NAT. The setting heuristic includes the 3-second delay, but source
addresses seen from incoming RTP packets are ranked according to preference: If a packet
with a source address and port matching the SDP address is seen, this address is used.
Otherwise, if a packet with a matching source address (but a different port) is seen,
that address is used. Otherwise, if a packet with a matching source port (but different
address) is seen, that address is used. Otherwise, the source address of any incoming
packet seen is used.
• --jitter-buffer=INT
Size of (incoming) jitter buffer in packets. A value of zero (the default) disables the
jitter buffer. The jitter buffer is currently only implemented for userspace operation.
• --jb-clock-drift
Enable clock drift compensation for the jitter buffer.
• --debug-srtp
Enable extra log messages to help debug SRTP issues. Per-packet details such as
sequence numbers, ROC, payloads (plain text and encrypted), authentication tags, etc are
recorded to the log. Every RTCP packet is logged in this way, while every 512th RTP
packet is logged. Only applies to packets forwarded/processed in userspace.
• --reject-invalid-sdp
With this option set, refuse to process SDP bodies that could not be cleanly parsed,
instead of skipping over the parsing error and processing the SDP anyway. Currently
this only affects the processing of SDP bodies that end in a blank line.
• --listen-http=[IP|HOSTNAME:]PORT
• --listen-https=[IP|HOSTNAME:]PORT
Enable listening for HTTP or WebSocket connections, or their TLS-secured counterparts
HTTPS and WSS. If no interface is specified, then the listening socket will be bound to
all interfaces.
The HTTP listener supports both HTTP and WS, while the HTTPS listener supports both
HTTPS and WSS.
If HTTPS/WSS is enabled, a certificate must also be provided using the options below.
• --https-cert=FILE
• --https-key=FILE
Provide a server certificate and corresponding private key for the HTTPS/WSS listener,
in PEM format.
• --http-threads=INT
Number of worker threads for HTTP/HTTPS/WS/WSS. If not specified, then the same number
as given under num-threads will be used. If no HTTP listeners are enabled, then no
threads are created.
• --software-id=STRING
Sets a free-form string that is used to identify this software towards external systems
with, for example in outgoing ICE/STUN requests. Defaults to rtpengine-VERSION. The
string is sanitised to replace all non-alphanumeric characters with a dash to make it
universally usable.
• --dtx-delay=INT
Processing delay in milliseconds to handle discontinuous transmission (DTX) or other
transmission gaps. Defaults to zero (disabled) and is applicable to transcoded audio
streams only. When enabled, delays processing of received packets for the specified
time (much like a jitter buffer) in order to trigger DTX handling when a transmission
gap occurs. The decoder is then instructed to fill in the missing time during a
transmission gap, for example by generating comfort noise. The delay should be
configured to be higher than the expected incoming jitter.
• --max-dtx=INT
Maximum duration for DTX handling in seconds. If no further RTP media is received
within this time frame, then DTX processing will stop. Can be set to zero or negative
to disable and keep DTX processing on indefinitely. Defaults to 30 seconds.
• --dtx-buffer=INT
• --dtx-lag=INT
These two options together control the maximum number of packets and amount of audio
that is allowed to be held in the DTX buffer. The dtx-buffer option limits the number
of packets held in the DTX buffer, while the dtx-lag option limits the amount of audio
(in milliseconds) to be held in the DTX buffer. A DTX buffer overflow is declared when
both limits are exceeded, in which case DTX processing is sped up by dtx-shift
milliseconds.
The defaults are 10 packets and 100 milliseconds.
• --dtx-shift=INT
Amount of time in milliseconds that DTX processing is shifted forward (sped up) or
backwards (delayed) in case of a DTX buffer overflow or underflow. An underflow occurs
when RTP packets are received slower than expected, while an overflow occurs when
packets are received faster than expected.
If this value is set to zero then no adjustments of the DTX timer will be made.
Instead, in order to keep up with the flow of received RTP packets, packets will be
dropped or additional DTX audio will be generated as needed.
• --dtx-cn-params=INT
Specify one comfort noise parameter. This option follows the same format as cn-payload
described below.
This option is applicable to audio generated to fill in transmission gaps during a DTX
event. The default setting is no value, which means silence will be generated to fill
in DTX gaps.
If any CN parameters are configured, the parameters will be passed to an RFC 3389 CN
decoder, and the generated comfort noise will be used to fill in DTX gaps.
• --amr-dtx=native|CN
Select the DTX behaviour for AMR codecs. The default is use the codec’s internal
processing: during a DTX event, a “no data” frame is passed to the decoder and the
output is used as audio data.
If CN is selected here, the same DTX mechanism as other codecs use is used for AMR,
which is to fill in DTX gaps with either silence or RFC 3389 comfort noise (see dtx-cn-
params). This also affects processing of received SID frames: SID frames would not be
passed to the codec but instead be replaced by generated silence or comfort noise.
• --silence-detect=FLOAT
Enable silence detection and specify threshold in percent. This option is applicable to
transcoded stream only and defaults to zero (disabled).
When enabled, silence detection will be performed on all transcoded audio streams. The
threshold specified here is the sensitivity for detecting silence: higher thresholds
result in more audio to be detected as silence, while lower thresholds result in less
audio to be detected as silence. The threshold is specified as percent between zero and
100. If set to 100, then all audio would be detected as silence; if set to 50, then any
audio that is quieter than 50% of the maximum volume would be detected as silence; and
so on. Setting it to zero disables silence detection. To only detect silence that is
very near or equal to absolute silence, set this value to a low number such as 0.01.
(For certain codecs such as PCMA, a higher minimum threshold is required to detect
complete silence, as their compressed payloads don’t decode to actual silence but
instead have a residual DC offset. For PCMA the minimum value is 0.013.)
Audio that is detected as silence will be replaced by comfort noise as specified by the
cn-payload option (see below). Currently this is applicable only to RTP peers that have
advertised support for the CN RTP payload type, in which case the silence audio frames
will be replaced by CN RTP frames.
• --cn-payload=INT
Specify one comfort noise parameter. This option can be given multiple times and the
format follows RFC 3389. When specified at the command line, list the --cn-payload=
option multiple times, each one specifying a single CN parameter. When used in the
config file, list the option only a single time and list multiple CN parameters
separated by semicolons (e.g. cn-payload = 20;40;60).
The first CN payload value given is the noise level, specified as -dBov as per RFC 3389.
This means that a noise level of zero corresponds to maximum volume, while higher
numbers correspond to lower volumes. The highest allowable number is 127, corresponding
to -127 dBov, which is near silence.
Subsequent CN payload values carry spectral information (reflection coefficients) as per
RFC 3389. Allowable values for each coefficient are between 0 and 254. Specifying
spectral information is optional and the number of coefficients listed (model order) is
variable.
This option is applicable only to CN packets generated from the silence detection
mechanism described above. The configured CN parameters are used directly as payload of
CN packets sent by rtpengine.
The default values are 32 (-32 dBov) for the noise level and no spectral information.
• --player-cache
Enable caching of encoded media packets for media player. This is applicable for media
playback initiated through the play media command. When enabled rtpengine will not
simply decode given media files and then encode the media to RTP on demand and on the
fly, but will rather decode and encode each media file in full the first time playback
is requested, and then cache the resulting RTP packets in memory. This is done once for
each media file and for each output RTP codec requested.
Caching is done based on unique file name (with no consideration given to different file
names that may point to the same file), or integer index for media files played from
database. No verification of changing content of files or database entries is done.
Media files provided as binary blob are also cached, although in this case a hash over
the entire media file must be performed, therefore this usage is not recommended.
It’s not possible to choose a different start-pos for playback with this option enabled.
RTP data is cached and retained in memory for the lifetime of the process.
• audio-buffer-length=INT
Set the buffer length used by the audio player (see below) in milliseconds. The default
is 500 milliseconds.
The buffer must be long enough to accommodate at least two frames of audio from all
contributing sources, which means at least 40 ms or 60 ms for most cases. If media
playback (via the play media) command is desired, then the buffer must be able to
accommodate at least one full frame from the source media file, whose length can vary
depending on the format of the source media file. For 8 kHz .wav files this is 256 ms
(2048 samples). Therefore 500 ms is the recommended value.
• audio-buffer-delay=INT
Initial delay for new sources contributing to an audio buffer (used by the audio player,
see below) in milliseconds. The default is 5 ms.
The initial delay is meant to compensate for varying inter-arrival times of media
packets (jitter). If set too low, intermittent high jitter will result in gaps in the
output audio. If set too high, output audio will have an unnecessary latency added to
it.
• audio-player=on-demand|play-media|transcoding|always
Define when to enable the audio player if not explicitly instructed otherwise. The
default setting is on-demand.
Enabling the audio player for a party to a call makes rtpengine produce its own audio
RTP stream (instead of just forwarding an audio stream received from elsewhere). The
audio is generated from a circular audio buffer (see above) and all contributing audio
sources are mixed into that one audio buffer. Contributing audio sources are audio
streams received from elsewhere (that would otherwise simply be forwarded) and audio
produced by the play media command.
With this set to on-demand, the audio player is enabled only if explicitly requested by
the user for a particular call via the audio-player= option used in a signalling
message.
When set to play-media, the audio player is enabled only while media playback via the
play media command is active. After media playback is finished, the audio player is
again disabled and audio goes back to simply being forwarded.
Setting this option to transcoding leaves the audio player disabled unless any sort of
transcoding is required for a call.
With a setting of always, the audio player is enabled for all calls, unless explicitly
disabled via the audio-player= option used in a signalling message. This forces all
audio through the transcoding engine, even if input and output codecs are the same.
Audio player usage can be changed on a call-by-call basis by including the audio-player=
option in a signalling message. This option supports the values transcoding and always,
which result in the behaviour described just above, and off which forces the audio
player to be disabled regardless of this setting.
• --poller-per-thread
Enable `poller per thread' functionality: for every worker thread (see the \--num-
threads option) a poller will be created. With this option on, it is guaranteed that
only a single thread will ever read from a particular socket, thus maintaining the order
of the packets. Might help when having issues with DTMF packets (RFC 2833).
• --dtls-cert-cipher=prime256v1|RSA
Choose the type of key to use for the signature used by the self-signed certificate used
for DTLS. The previous default was RSA. The current default and the only other option
is prime256v1 which is a 256-bit elliptic-curve key.
• --dtls-signature=SHA-256|SHA-1
Choose the hash algorithm to use for the signature used by the self-signed certificate
used for DTLS. The default is SHA-256. Not to be confused with the hash algorithm used
for the certificate fingerprint inserted into the SDP (a=fingerprint:), which is
independent of the certificate’s signature and can be selected during runtime.
• --dtls-rsa-key-size=INT
Size in bits of the RSA key used by the DTLS certificate, if RSA is in use. Default is
2048 bits.
• --dtls-ciphers=STRING
Ciphers allowed during the DTLS key exchange (not to be confused with the cipher used by
the DTLS certificate). The format of this string is an OpenSSL cipher list. The
default is DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK
• --dtls-mtu=INT
Set DTLS MTU to enable fragmenting of large DTLS packets. Defaults to 1200. Minimum
value is 576 as the internet protocol requires that hosts must be able to process IP
datagrams of at least 576 bytes (for IPv4) or 1280 bytes (for IPv6). This does not
preclude link layers with an MTU smaller than this minimum MTU from conveying IP data.
Internet IPv4 path MTU is 68 bytes.
• --mqtt-host=HOST|IP
Host or IP address of the Mosquitto broker to connect to. Must be set to enable
exporting stats to Mosquitto.
• --mqtt-port=INT
Port of the Mosquitto broker. Defaults to 1883.
• --mqtt-id=STRING
Client ID to use for Mosquitto. Default is a generated random string.
• --mqtt-keepalive=INT
Keepalive interval in seconds. Defaults to 30.
• --mqtt-user=USERNAME
• --mqtt-pass=PASSWORD
Credentials to connect to Mosquitto broker. At least a username must be given to enable
authentication.
• --mqtt-cafile=FILE
• --mqtt-capath=PATH
• --mqtt-certfile=FILE
• --mqtt-keyfile=FILE
• --mqtt-tls-alpn=STRING
Enable TLS to connect to Mosquitto broker, optionally with client certificate
authentication. At least cafile or capath must be given to enable TLS. To enable
client certificate authentication, both certfile and keyfile must be set. All files
must be in PEM format. Password-proteted files are not supported. The tls-alpn can be
set (e.g. mqtt) if a service like AWS IoT Core shares the same TLS port for two
different network protocols.
• --mqtt-publish-qos=0|1|2
QoS value to use for publishing to Mosquitto. See Mosquitto docs for details.
• --mqtt-publish-topic=STRING
Topic string to use for publishing to Mosquitto. Must be set to a non-empty string.
• --mqtt-publish-interval=MILLISECONDS
Interval in milliseconds to publish to Mosquitto. Defaults to 5000 (5 seconds).
• --mqtt-publish-scope=global|summary|call|media
When set to summary, one message will be published to Mosquitto every interval
milliseconds containing all global stats. A setting of global has the same effect as
summary but will also contain a list of all running calls with stats for each call.
When set to call, one message per call will be published to Mosquitto with stats for
that call every interval milliseconds, plus one message every interval milliseconds with
global stats. When set to media, one message per call media (usually one media per call
participant, so usually 2 media per call) will be published to Mosquitto with stats for
that call media every interval milliseconds, plus one message every interval
milliseconds with global stats.
• --mos=CQ|LQ
MOS (Mean Opinion Score) calculation formula. Defaults to CQ (conversational quality)
which takes RTT into account and therefore requires peers to correctly send RTCP. If
set to LQ (listening quality) RTT is ignored, allowing a MOS to be calculated in the
absence of RTCP.
• --measure-rtp
Enable measuring RTP metrics even for plain RTP passthrough scenarios. Without that
option, RTP metrics are measured only in transcoding scenarios.
• --socket-cpu-affinity=INT
Enables setting the socket CPU affinity via the SO*INCOMING*CPU socket option if
available. The default value is zero which disables this feature. If set to a positive
number then the CPU affinity for all sockets belonging to the same call will be set to
the same value. The number specifies the upper limit of the affinity to be set, and
values will be used in a round-robin fashion (e.g. if set to 8 then the values 0 through
7 will be used to set the affinity). If this option is set to a negative number, then
the number of available CPU cores will be used.
INTERFACES
The command-line options -i or --interface, or equivalently the interface config file
option, specify local network interfaces for RTP. At least one must be given, but
multiple can be specified. The format of the value is [NAME/]IP[!IP] with IP being either
an IPv4 address, an IPv6 address, the name of a system network interface (such as eth0), a
DNS host name (such as test.example.com), or any.
The possibility of configuring a network interface by name rather than by address should
not be confused with the logical interface name used internally by rtpengine (as described
below). The NAME token in the syntax above refers to the internal logical interface name,
while the name of a system network interface is used in place of the first IP token in the
syntax above. For example, to configure a logical network interface called int using all
the addresses from the existing system network interface eth0, you would use the syntax
int/eth0. (Unless omitted, the second IP token used for the advertised address must be an
actual network address and cannot be an interface name.)
If DNS host names are used instead of addresses or interface names, the lookup will be
done only once during daemon start-up.
The special keyword any can be used to listen on any and all available local interface
addresses except from loopback devices. This keyword should only be given once in place
of a more explicit interface configuration.
To configure multiple interfaces using the command-line options, simply present multiple
-i or --interface options. When using the config file, only use a single interface line,
but specify multiple values separated by semicolons (e.g. interface =
internal/12.23.34.45;external/23.34.45.54).
If an interface option is given using a system interface name in place of a network
address, and if multiple network address are found configured on that network interface,
then rtpengine behaves as if multiple --interface options had been specified. For
example, if interface eth0 exists with both addresses 192.168.1.120 and
2001:db8:85a3::7334 configured on it, and if the option --interface=ext/eth0 is given,
then rtpengine would behave as if both options --interface=ext/192.168.1.120 and
--interface=ext/2001:db8:85a3::7334 had been specified.
The second IP address after the exclamation point is optional and can be used if the
address to advertise in outgoing SDP bodies should be different from the actual local
address. This can be useful in certain cases, such as your SIP proxy being behind NAT.
For example, --interface=10.65.76.2!192.0.2.4 means that 10.65.76.2 is the actual local
address on the server, but outgoing SDP bodies should advertise 192.0.2.4 as the address
that endpoints should talk to. Note that you may have to escape the exclamation point
from your shell when using command-line options, e.g. using \!.
Giving an interface a name (separated from the address by a slash) is optional; if
omitted, the name default is used. Names are useful to create logical interfaces which
consist of one or more local addresses. It is then possible to instruct rtpengine to use
particular interfaces when processing an SDP message, to use different local addresses
when talking to different endpoints. The most common use case for this is to bridge
between one or more private IP networks and the public internet.
For example, if clients coming from a private IP network must communicate their RTP with
the local address 10.35.2.75, while clients coming from the public internet must
communicate with your other local address 192.0.2.67, you could create one logical
interface pub and a second one priv by using --interface=pub/192.0.2.67
--interface=priv/10.35.2.75. You can then use the direction option to tell rtpengine
which local address to use for which endpoints (either pub or priv).
If multiple logical interfaces are configured, but the direction option is not given in a
particular call, then the first interface given on the command line will be used.
It is possible to specify multiple addresses for the same logical interface (the same
name). Most commonly this would be one IPv4 addrsess and one IPv6 address, for example:
--interface=192.168.63.1 --interface=fe80::800:27ff:fe00:0. In this example, no interface
name is given, therefore both addresses will be added to a logical interface named
default. You would use the address family option to tell rtpengine which address to use
in a particular case.
It is also possible to have multiple addresses of the same family in a logical network
interface. In this case, the first address (of a particular family) given for an
interface will be the primary address used by rtpengine for most purposes. Any additional
addresses will be advertised as additional ICE candidates with increasingly lower
priority. This is useful on multi-homed systems and allows endpoints to choose the best
possible path to reach the RTP proxy. If ICE is not being used, then additional addresses
will go unused, even though ports would still get allocated on those interfaces.
Another option is to give interface names in the format BASE:SUFFIX. This allows
interfaces to be used in a round-robin fashion, useful for load-balancing the port ranges
of multiple interfaces. For example, consider the following configuration:
--interface=pub:1/192.0.2.67 --interface=pub:2/10.35.2.75. These two interfaces can still
be referenced directly by name (e.g. direction=pub:1), but it is now also possible to
reference only the base name (i.e. direction=pub). If the base name is used, one of the
two interfaces is selected in a round-robin fashion, and only if the interface actually
has enough open ports available. This makes it possible to effectively increase the
number of available media ports across multiple IP addresses. There is no limit on how
many interfaces can share the same base name.
It is possible to combine the BASE:SUFFIX notation with specifying multiple addresses for
the same interface name. An advanced example could be (using config file notation, and
omitting actual network addresses):
interface = pub:1/IPv4 pub:1/IPv4 pub:1/IPv6 pub:2/IPv4 pub:2/IPv6 pub:3/IPv6 pub:4/IPv4
In this example, when direction=pub is IPv4 is needed as a primary address, either pub:1,
pub:2, or pub:4 might be selected. When pub:1 is selected, one IPv4 and one IPv6 address
will be used as additional ICE alternatives. For pub:2, only one IPv6 is used as ICE
alternative, and for pub:4 no alternatives would be used. When IPv6 is needed as a
primary address, either pub:1, pub:2, or pub:3 might be selected. If at any given time
not enough ports are available on any interface, it will not be selected by the round-
robin algorithm.
It is possible to use the round-robin algorithm even if the direction is not given. If
the first given interface has the BASE:SUFFIX format then the round-robin algorithm is
used and will select interfaces with the same BASE name.
If you are not using the NG protocol but rather the legacy UDP protocol used by the
rtpproxy module, the interfaces must be named internal and external corresponding to the i
and e flags if you wish to use network bridging in this mode.
EXIT STATUS
• 0
Successful termination.
• 1
An error occurred.
ENVIRONMENT
• RTPENGINE*REDIS*AUTH*PW
Redis server password for persistent state storage.
• RTPENGINE*REDIS*WRITE*AUTH*PW
Redis server password for write operations, if --redis has been specified, in which case
the one specified in --redis will be used for read operations only.
FILES
• /etc/rtpengine/rtpengine.conf
Configuration file.
EXAMPLES
A typical command line (enabling both UDP and NG protocols) may look like:
rtpengine --table=0 --interface=10.64.73.31 --interface=2001:db8::4f3:3d \
--listen-udp=127.0.0.1:22222 --listen-ng=127.0.0.1:2223 --tos=184 \
--pidfile=/run/rtpengine.pid
SEE ALSO
kamailio(8) (http://man.he.net/man8/kamailio).