Provided by: rtpengine-daemon_12.5.1.22-1_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 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.
• --nftables-chain=CHAIN
Name of the netfilter chain in which to create the custom forwarding rule required for in-kernel packet
forwarding. Defaults to rtpengine. Only used if in-kernel packet forwarding is enabled (table set to
zero or higher).
At startup rtpengine creates a new netfilter chain with this name (in the filter table) if it doesn’t
yet exist, or flushes (empties out) the chain if it already exists. It then creates a single
forwarding rule in this chain to direct media packets into the kernel module for processing.
The rule and the chain are deleted during shutdown.
Explicitly setting this option to an empty string disables managing of a netfilter chain and prevents
creation of the custom forwarding rule.
• --nftables-base-chain=CHAIN
Name of the netfilter base chain to use as entry point for in-kernel packet forwarding. Defaults to
INPUT to match legacy iptables setups. Only applicable if the option nftables-chain is active.
If the chain with this name doesn’t exist during startup, rtpengine will create it as a base chain. It
then adds a single immediate-goto (jump) rule to the chain given by the nftables-chain option. During
shutdown this rule is again deleted.
If this option is explicitly set to an empty string, then rtpengine will directly create the chain
given by nftables-chain as a base chain and skip creating the immediate-goto rule.
If this option is set to the special string none, then rtpengine will create its custom chain and rule
as it normally would, but will skip adding an immediate-goto rule to the custom chain. Doing so
requires the operator to manually create this immediate-goto rule somewhere themselves. Otherwise
in-kernel packet forwarding would be left inoperable.
• --nftables-append
With this option set, the netfilter rule created in the base chain is appended to the list of existing
rules. The default is to prepend it (insert it at the beginning).
• --nftables-family=ip|ip6|ip,ip6
Configure for which netfilter address family to manage tables, chains, and rules. The default is to
manage both IPv4 and IPv6 address families.
• --nftables-start
• --nftables-stop
Instructs rtpengine to execute the actions described under nftables-chain and nftables-base-chain and
then immediately exit. Useful to manually re-create the rule(s) if they have gotten lost during
runtime, and/or to manually manage creation and deletion of these rules from a script (typically in
combination with an empty nftables-chain= in the main config file).
• --nftables-status
Instructs rtpengine to check for the existence of the managed netfilter rules and chains, print the
result of check, and exit. The process will exit with code 0 if the check was successful, and 1
otherwise.
• -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 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) 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.
• --codec-num-threads=INT
Enables asynchroneous transcoding operation using the specified number of worker threads. This is an
experimental feature and probably doesn’t bring any benefits over normal synchroneous transcoding.
• --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.
• --max-recv-iters=INT
This parameter sets maximum continuous reading cycles in UDP poller loop, can help to avoid dropped
packets errors on bursty streams (default 50).
• --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. Also enables sending the control NG traffic to a capturing agent. Payload format does not
apply in this case.
• --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.
• --homer-disable-rtcp-stats
Disables the default behaviour that RTCP stats are sent when homer parameter is set. Sending of RTCP
and NG are as such decoupled.
• --homer-enable-ng
Enables sending control NG packages to a Homer capturing software. The capturing agent part is not
officialy supported OOTB, but it can be achieved with Kamailio by using the config. For this feature
to work one has to set at least the homer parameter.
• --homer-ng-capture-proto=INT
The HEP protocol used by Homer contains a “Capture protocol type” UINT8 used by the capturing agent and
UI to make further processing. Some values are registered, but currently 0x3d values onwards are free.
Default value is 0x3d (61).
• --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.
• --kernel-player=INT
• --kernel-player-media=INT
Enables and configures the kernel-based media player. Disabled by default and only available if the
kernel module is in use, and requires player-cache to also be enabled.
When enabled, media playback will be handled by a set of kernel threads. The option kernel-player
defaults to zero and needs to set to non-zero to enable the feature. The number given to the option is
the maximum number of concurrent kernel media players that can be used.
The option kernel-player-media configures the maximum number of unique media “files” that can be stored
for playback in the kernel module. Media files requested for playback are first decoded by the
player-cache feature, and then given to the kernel module in a pre-encoded format for quick playback.
Defaults to 128.
Both player slots and media slots are shared among all instances of rtpengine (using different kernel
table IDs) running on a system using the same kernel module. Unused slots use minimal resources.
• 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).
• --io-uring
Enable experimental support for io_uring. Requires Linux kernel 6.0 or later.
When enabled, instead of the usual polling mechanism each worker thread will set up its own io_uring
and use it for polling, as well as directly sending and receiving certain network data. In particular
userspace media data is sent and received directly via io_uring.
NOTE: As of the time of writing, worker threads sleeping in an io_uring poll are attributed to the host
system as I/O wait CPU usage, with up to 99% CPU time spent in I/O wait (depending on the number of
worker threads), but without being attributed to any process or thread. This is not actual CPU usage
but rather indicates time spent waiting for a network event, and so should be considered the same as
idle CPU time.
• --io-uring-buffers=INT
Number of io_uring entries in the buffer allocated from the kernel per thread. Defaults to 16384.
Must be large enough so that submission entries and completion entries are always available when
needed.
• --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.
• --rtcp-interval=INT
Delay in milliseconds between RTCP packets when generate-rtcp flag is on. The effective value includes
the random dispersion between 0..1 seconds on top, so the timer execution period is randomized and up
to 1 sec greater than given value in ms. Defaults to 5000 ms (5 seconds).
• --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).
System Network Interfaces
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.
Advertised Address
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 \!.
Interface Names
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.
Multiple Addresses per Interface
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.
Round-Robin Address Selection
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.
Alias Names
Interface alias names can be created using the ALIAS=NAME syntax. The alias must be listed after the
primary interface that it references. For example, to create an actual logical interface pub1 and then
an alias pub for that interface:
interface = pub1/IPv4;pub=pub1
Interface aliases are useful in combination with Redis replication. If an interface is referred to via
an alias name (e.g. direction=pub), then the interface’s actual name (pub1 in this example) is propagated
into the Redis storage and thus to any dependent standby instances. These standby instances can then
have different address configurations for that interface, which makes it possible to facilitate failover
with static addressing (for example behind an IP load balancer).
Legacy Protocols
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).
12.5.1.22-1 2025-01-22 rtpengine(8)