From 125c17684daa078b9de2c5c30ecc87bac00d5df3 Mon Sep 17 00:00:00 2001
From: Donat Zenichev <dzenichev@sipwise.com>
Date: Thu, 13 Apr 2023 11:40:18 +0200
Subject: [PATCH] MT#55897 Convert `rtpengine.pod` / `rtpengine-recording.pod`
 to MD

For a compatibility reasons (with other doc files and with RTD)
the `rtpengine.pod` and `rtpengine-recording.pod` file get
converted to the Markdown syntax.

The compilation of MD syntax synopsis files will work
using `ronn` application, which converts them to man pages.

Change-Id: I75b54a712786a0a237c51c702ed1a2cc09e3a033
---
 daemon/Makefile                             |    4 +-
 daemon/rtpengine.8.ronn                     |    1 +
 daemon/rtpengine.pod                        |    1 -
 debian/control                              |    2 +
 docs/rtpengine-recording.md                 |  340 +++++
 docs/rtpengine-recording.pod                |  369 -----
 docs/rtpengine.md                           | 1329 ++++++++++++++++++
 docs/rtpengine.pod                          | 1365 -------------------
 docs/usage.md                               |    2 +-
 lib/common.Makefile                         |    9 +-
 recording-daemon/Makefile                   |    4 +-
 recording-daemon/rtpengine-recording.8.ronn |    1 +
 recording-daemon/rtpengine-recording.pod    |    1 -
 13 files changed, 1682 insertions(+), 1746 deletions(-)
 create mode 120000 daemon/rtpengine.8.ronn
 delete mode 120000 daemon/rtpengine.pod
 create mode 100644 docs/rtpengine-recording.md
 delete mode 100644 docs/rtpengine-recording.pod
 create mode 100644 docs/rtpengine.md
 delete mode 100644 docs/rtpengine.pod
 create mode 120000 recording-daemon/rtpengine-recording.8.ronn
 delete mode 120000 recording-daemon/rtpengine-recording.pod

diff --git a/daemon/Makefile b/daemon/Makefile
index 6dceebb32..9b468cf4a 100644
--- a/daemon/Makefile
+++ b/daemon/Makefile
@@ -90,8 +90,8 @@ LIBASM=		mvr2s_x64_avx2.S mvr2s_x64_avx512.S
 endif
 OBJS=		$(SRCS:.c=.o) $(LIBSRCS:.c=.o) $(LIBASM:.S=.o)
 
-PODS=		rtpengine.pod
-MANS=		$(PODS:.pod=.8)
+MDS=		rtpengine.ronn
+MANS=		$(MDS:.ronn=.8)
 
 include ../lib/common.Makefile
 
diff --git a/daemon/rtpengine.8.ronn b/daemon/rtpengine.8.ronn
new file mode 120000
index 000000000..281c18e44
--- /dev/null
+++ b/daemon/rtpengine.8.ronn
@@ -0,0 +1 @@
+../docs/rtpengine.md
\ No newline at end of file
diff --git a/daemon/rtpengine.pod b/daemon/rtpengine.pod
deleted file mode 120000
index 2c53748da..000000000
--- a/daemon/rtpengine.pod
+++ /dev/null
@@ -1 +0,0 @@
-../docs/rtpengine.pod
\ No newline at end of file
diff --git a/debian/control b/debian/control
index 0d822959e..50ce11c1e 100644
--- a/debian/control
+++ b/debian/control
@@ -46,6 +46,8 @@ Build-Depends:
  markdown,
  python3,
  python3-websockets,
+ ronn,
+ ruby-ronn,
  zlib1g-dev,
 Testsuite: autopkgtest-pkg-dkms
 
diff --git a/docs/rtpengine-recording.md b/docs/rtpengine-recording.md
new file mode 100644
index 000000000..43d29687c
--- /dev/null
+++ b/docs/rtpengine-recording.md
@@ -0,0 +1,340 @@
+rtpengine-recording(8) - media recording daemon for Sipwise rtpengine
+==========================================
+
+## SYNOPSIS
+
+**rtpengine-recording** \[_option_...\]
+
+## DESCRIPTION
+
+The Sipwise rtpengine media proxy has support for exporting media (RTP) packets
+that it forwards. The rtpengine-recording daemon collects these exported
+packets and decodes them into an audio format that can be listened to.
+
+## OPTIONS
+
+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.
+
+If no options are given, then default values are assumed, which should be
+sufficient for a standard installation of rtpengine.
+
+- **--help**
+
+    Print the usage information.
+
+- **-v**, **--version**
+
+    If called with this option, the **rtpengine-recording** daemon will simply print
+    its version number 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. **output-mixed**), 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.
+
+    As a special value, **none** can be passed here to suppress loading of the
+    default config file.
+
+- **--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-recording**.
+    A config file section is started in the config file using square brackets
+    (e.g. **\[rtpengine-recording\]**).
+
+- **-L**, **--log-level=**_INT_
+
+    Takes an integer as argument and controls the highest log level which
+    will be sent to syslog.
+    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.
+
+- **--log-facilty=****daemon**\|**local0**\|...\|**local7**\|...
+
+    The syslog facilty to use when sending log messages to the syslog daemon.
+    Defaults to **daemon**.
+
+- **-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.
+
+- **--no-log-timestamps**
+
+    Don't add timestamps to log lines written to stderr.
+    Only useful in combination with **--log-stderr**.
+
+- **--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.
+
+- **-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.
+
+- **-t**, **--table=**_INT_
+
+    Takes an integer argument. The value must match the **table** option given to
+    the **rtpengine** media proxy to use for in-kernel packet forwarding.
+    Defaults to **0** if not specified.
+
+- **--spool-dir=**_PATH_
+
+    The path given here must match the **recording-dir** path given to the
+    **rtpengine** media proxy. Defaults to `/var/spool/rtpengine`. The path must
+    reside on a file system that supports the **inotify** mechanism.
+
+- **--num-threads=**_INT_
+
+    How many worker threads to launch. Defaults to the number of CPU cores
+    available, or **8** if there are fewer than that or if the number is not
+    known.
+
+- **--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.
+
+- **--output-storage=****file**\|**db**\|**both**
+
+    Where to store media files. By default, media files are written directly to the
+    file system (see **output-dir**). They can also be stored as a **BLOB** in a
+    MySQL database, either instead of, or in addition to, being written to the file
+    system.
+
+- **--output-dir=**_PATH_
+
+    Path for media files to be written to if file output is enabled. Defaults to
+    `/var/lib/rtpengine-recording`. The path must not be the same as used for the
+    **spool-dir**.
+
+- **--output-pattern=**_STRING_
+
+    File name pattern to be used for recording files. The pattern can reference
+    sub-directories. Parent directories will be created on demand. The default
+    setting is **%c-%t**.
+
+    The pattern must include **printf**-style format sequences. Supported format
+    sequences are:
+
+    - **%%**
+
+        A literal percent sign.
+
+    - **%c**
+
+        The call ID. It is mandatory for the output pattern to include this format
+        sequence.
+
+    - **%t**
+
+        The stream type. For **single** streams this is the SSRC written as hexadecimal;
+        for **mix** stream this is the string **mix**. It is mandatory for the output
+        pattern to include this format sequence.
+
+    - **%l**
+
+        The label for the participating party as communicated from the controlling
+        daemon.
+
+    - **%Y**
+    - **%m**
+    - **%d**
+    - **%H**
+    - **%M**
+    - **%S**
+
+        These format sequence reference the current system time (when the output file
+        was created) and are the same as the format sequences supported by [date(1)](http://man.he.net/man1/date)
+        or [strftime(3)](http://man.he.net/man3/strftime) (year, month, day, hours, minutes, and seconds,
+        respectively).
+
+    - **%u**
+
+        Microseconds, expanded to 6 digits (**000000** through **999999**).
+
+    - **%**_INT_
+
+        References a prefix from the call ID of the given length. If this format
+        sequence is present more than once, then the prefixes are cumulative. For
+        example, if the call ID is **abcdefgh** and the output pattern is configured as
+        **%2/%3/%c**, then the resulting output file name would be **ab/cde/abcdefgh**.
+
+- **--output-format=****wav**\|**mp3**\|**none**
+
+    File format to be used for media files that are produced. Defaults to PCM WAV
+    (RIFF) files. Applicable for both files stored on the file system and in a
+    database. If **none** is selected then file output is disabled.
+
+- **--resample-to=**_INT_
+
+    Resample all audio to the given sample rate (e.g. **48000**). Resampling is
+    disabled by default, meaning that files will be written with the same sample
+    rate as the source media.
+
+- **--mp3-bitrate=**_INT_
+
+    If MP3 output is selected, use the given bitrate for the MP3 encoder (e.g.
+    **64000**). There is no default value, so this option must be given if MP3
+    output is selected. Note that not all bitrates are valid in combinations with
+    all sample rates. For MP3 output it's therefore recommended to also set
+    **resample-to**.
+
+- **--output-mixed**
+- **--output-single**
+
+    Whether to produce **mixed** audio files, or **single** audio files, or both. If
+    neither option is given, then by default both are enabled. If no file output is
+    desired, set **output-format** to **none**.
+
+    A **single** audio file contains the audio for a single RTP SSRC, which usually
+    means an unidirectional audio stream. These are decoded directly from an RTP
+    stream and do not take timestamping into account, meaning that gaps or pauses
+    in the RTP stream are not reflected in the output audio file.
+
+    A **mixed** audio file consists of the first four RTP SSRC seen, mixed together
+    into a single output file, which usually means that a bidirectional audio
+    stream is produced. Audio mixing takes RTP timestamping into account, so gaps
+    and pauses in the RTP media are reflected in the output audio to keep the
+    multiple audio sources in sync.
+
+- **--mix-method=****direct**\|**channels**
+
+    Selects a method to mix multiple audio inputs into a single output file for
+    **mixed** output. The default is **direct** which directly mixes all audio inputs
+    together, producing a mixed output file with the same format as an audio file
+    from a single input (**output-single**) would be.
+
+    The **channels** mixing method puts each audio input into its own audio channel
+    in the output file, therefore producing a multi-channel output file. Up to four
+    separate RTP SSRCs are supported for a mixed output, which means that if each
+    input is mono audio, then the mixed output file would contain 4 audio channels.
+    This mixing method requires an output file format which supports these kinds of
+    multi-channel audio formats (e.g. **wav**).
+
+- **--mix-num-inputs=**_INT_
+
+    Change the number of recording channel in the output file. The value is between 1 to 4 (e.g. **4**, which is also the default value).
+
+- **--output-chmod=**_INT_
+
+    Change the file permissions of recording files to the given mode. Must be given
+    as an octal integer, for example **0660**.
+
+- **--output-chmod-dir=**_INT_
+
+    Change the file permissions of recording files to the given mode. Must be given
+    as an octal integer, for example **0700** (which is also the default value).
+
+- **--output-chown=**_USER_\|_UID_
+- **--output-chgrp=**_GROUP_\|_GID_
+
+    Change the ownership of recording files. Either user/group names or numeric IDs
+    are supported. If the value is blank or given as **-1** then the user/group is
+    left unchanged.
+
+- **--mysql-host=**_HOST_\|_IP_
+- **--mysql-port=**_INT_
+- **--mysql-user=**_USERNAME_
+- **--mysql-pass=**_PASSWORD_
+- **--mysql-db=**_STRING_
+
+    Configuration for a MySQL storage backend. Details about calls and media files
+    that are produced are stored into the database. Optionally the media files
+    themselves can be stored as well (see **output-storage**).
+
+- **--forward-to=**_PATH_
+
+    Forward raw RTP packets to a Unix socket. Disabled by default.
+
+- **--tls-send-to=**_IP_**:**_PORT_
+- **--tls-resample=**_INT_
+
+    Send decoded audio over a TCP TLS connection to the specified destination.
+    Audio is sent as raw mono 16-bit PCM in the given sample rate.
+
+- **--notify-uri=**_URI_
+
+    Enable HTTP notification about finished recordings to the specified URI, which
+    must be an HTTP or HTTPS URI. Information about the finished recording is
+    provided via custom HTTP headers, all of which use a prefix of **X-Recording-**.
+
+- **--notify-post**
+
+    Use HTTP POST instead of GET for the HTTP notification requests. The request
+    body is empty even if POST is used.
+
+- **--notify-no-verify**
+
+    Disable TLS peer certificate verification for HTTPS requests.
+
+- **--notify-concurrency=**_INT_
+
+    The maximum number of HTTP requests to perform simultaneously.
+
+- **--notify-retries=**_INT_
+
+    How many times to retry a failed HTTP notification before giving up. An
+    exponential falloff time is used for each subsequent attempt, starting with 5
+    seconds.
+
+- **--notify-record**
+
+    Attach recorded file to HTTP notification request. If enabled, notification
+    request behaves as HTTP POST (ignoring **--notify-post**). Note that this option
+    is incompatible with DB-only storage as no recording file exists on storage
+    (see **output-storage**).
+
+## EXIT STATUS
+
+- **0**
+
+    Successful termination.
+
+- **1**
+
+    An error occurred.
+
+## FILES
+
+- `/etc/rtpengine/rtpengine-recording.conf`
+
+    Configuration file.
+
+## SEE ALSO
+
+[rtpengine(8)](http://man.he.net/man8/rtpengine).
diff --git a/docs/rtpengine-recording.pod b/docs/rtpengine-recording.pod
deleted file mode 100644
index ea23e5c1c..000000000
--- a/docs/rtpengine-recording.pod
+++ /dev/null
@@ -1,369 +0,0 @@
-=head1 NAME
-
-rtpengine-recording - media recording daemon for Sipwise rtpengine
-
-=head1 SYNOPSIS
-
-B<rtpengine-recording> [I<option>...]
-
-=head1 DESCRIPTION
-
-The Sipwise rtpengine media proxy has support for exporting media (RTP) packets
-that it forwards. The rtpengine-recording daemon collects these exported
-packets and decodes them into an audio format that can be listened to.
-
-=head1 OPTIONS
-
-All options can (and should) be provided in a config file instead of
-at the command line. See the B<--config-file> option below for details.
-
-If no options are given, then default values are assumed, which should be
-sufficient for a standard installation of rtpengine.
-
-=over 4
-
-=item B<--help>
-
-Print the usage information.
-
-=item B<-v>, B<--version>
-
-If called with this option, the B<rtpengine-recording> daemon will simply print
-its version number and exit.
-
-=item B<--config-file=>I<FILE>
-
-Specifies the location of a config file to be used. The config file is an
-I<.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. B<table> instead of just B<t>) must be
-used in the config file.
-For boolean options that are either present or not (e.g. B<output-mixed>), a
-boolean value (either B<true> or B<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.
-
-As a special value, B<none> can be passed here to suppress loading of the
-default config file.
-
-=item B<--config-section=>I<STRING>
-
-Specifies the I<.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 B<rtpengine-recording>.
-A config file section is started in the config file using square brackets
-(e.g. B<[rtpengine-recording]>).
-
-=item B<-L>, B<--log-level=>I<INT>
-
-Takes an integer as argument and controls the highest log level which
-will be sent to syslog.
-The log levels correspond to the ones found in the L<syslog(3)> man page.
-The default value is B<6>, equivalent to LOG_INFO.
-The highest possible value is B<7> (LOG_DEBUG) which will log everything.
-
-=item B<--log-facilty=>B<daemon>|B<local0>|...|B<local7>|...
-
-The syslog facilty to use when sending log messages to the syslog daemon.
-Defaults to B<daemon>.
-
-=item B<-E>, B<--log-stderr>
-
-Log to stderr instead of syslog.
-Only useful in combination with B<--foreground>.
-
-=item B<--split-logs>
-
-Split multi-line log messages into individual log messages so that each
-line receives its own log line prefix.
-
-=item B<--no-log-timestamps>
-
-Don't add timestamps to log lines written to stderr.
-Only useful in combination with B<--log-stderr>.
-
-=item B<--log-mark-prefix=>I<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.
-
-=item B<--log-mark-suffix=>I<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.
-
-=item B<-p>, B<--pidfile=>I<FILE>
-
-Specifies a path and file name to write the daemon's PID number to.
-
-=item B<-f>, B<--foreground>
-
-If given, prevents the daemon from daemonizing, meaning it will stay in
-the foreground.
-Useful for debugging.
-
-=item B<-t>, B<--table=>I<INT>
-
-Takes an integer argument. The value must match the B<table> option given to
-the B<rtpengine> media proxy to use for in-kernel packet forwarding.
-Defaults to B<0> if not specified.
-
-=item B<--spool-dir=>I<PATH>
-
-The path given here must match the B<recording-dir> path given to the
-B<rtpengine> media proxy. Defaults to F</var/spool/rtpengine>. The path must
-reside on a file system that supports the B<inotify> mechanism.
-
-=item B<--num-threads=>I<INT>
-
-How many worker threads to launch. Defaults to the number of CPU cores
-available, or B<8> if there are fewer than that or if the number is not
-known.
-
-=item B<--thread-stack=>I<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.
-
-=item B<--evs-lib-path=>I<FILE>
-
-Points to the shared object file (B<.so>) containing the reference
-implementation for the EVS codec. See the F<README> for more details.
-
-=item B<--output-storage=>B<file>|B<db>|B<both>
-
-Where to store media files. By default, media files are written directly to the
-file system (see B<output-dir>). They can also be stored as a B<BLOB> in a
-MySQL database, either instead of, or in addition to, being written to the file
-system.
-
-=item B<--output-dir=>I<PATH>
-
-Path for media files to be written to if file output is enabled. Defaults to
-F</var/lib/rtpengine-recording>. The path must not be the same as used for the
-B<spool-dir>.
-
-=item B<--output-pattern=>I<STRING>
-
-File name pattern to be used for recording files. The pattern can reference
-sub-directories. Parent directories will be created on demand. The default
-setting is B<%c-%t>.
-
-The pattern must include B<printf>-style format sequences. Supported format
-sequences are:
-
-=over
-
-=item B<%%>
-
-A literal percent sign.
-
-=item B<%c>
-
-The call ID. It is mandatory for the output pattern to include this format
-sequence.
-
-=item B<%t>
-
-The stream type. For B<single> streams this is the SSRC written as hexadecimal;
-for B<mix> stream this is the string B<mix>. It is mandatory for the output
-pattern to include this format sequence.
-
-=item B<%l>
-
-The label for the participating party as communicated from the controlling
-daemon.
-
-=item B<%Y>
-
-=item B<%m>
-
-=item B<%d>
-
-=item B<%H>
-
-=item B<%M>
-
-=item B<%S>
-
-These format sequence reference the current system time (when the output file
-was created) and are the same as the format sequences supported by L<date(1)>
-or L<strftime(3)> (year, month, day, hours, minutes, and seconds,
-respectively).
-
-=item B<%u>
-
-Microseconds, expanded to 6 digits (B<000000> through B<999999>).
-
-=item B<%>I<INT>
-
-References a prefix from the call ID of the given length. If this format
-sequence is present more than once, then the prefixes are cumulative. For
-example, if the call ID is B<abcdefgh> and the output pattern is configured as
-B<%2/%3/%c>, then the resulting output file name would be B<ab/cde/abcdefgh>.
-
-=back
-
-=item B<--output-format=>B<wav>|B<mp3>|B<none>
-
-File format to be used for media files that are produced. Defaults to PCM WAV
-(RIFF) files. Applicable for both files stored on the file system and in a
-database. If B<none> is selected then file output is disabled.
-
-=item B<--resample-to=>I<INT>
-
-Resample all audio to the given sample rate (e.g. B<48000>). Resampling is
-disabled by default, meaning that files will be written with the same sample
-rate as the source media.
-
-=item B<--mp3-bitrate=>I<INT>
-
-If MP3 output is selected, use the given bitrate for the MP3 encoder (e.g.
-B<64000>). There is no default value, so this option must be given if MP3
-output is selected. Note that not all bitrates are valid in combinations with
-all sample rates. For MP3 output it's therefore recommended to also set
-B<resample-to>.
-
-=item B<--output-mixed>
-
-=item B<--output-single>
-
-Whether to produce B<mixed> audio files, or B<single> audio files, or both. If
-neither option is given, then by default both are enabled. If no file output is
-desired, set B<output-format> to B<none>.
-
-A B<single> audio file contains the audio for a single RTP SSRC, which usually
-means an unidirectional audio stream. These are decoded directly from an RTP
-stream and do not take timestamping into account, meaning that gaps or pauses
-in the RTP stream are not reflected in the output audio file.
-
-A B<mixed> audio file consists of the first four RTP SSRC seen, mixed together
-into a single output file, which usually means that a bidirectional audio
-stream is produced. Audio mixing takes RTP timestamping into account, so gaps
-and pauses in the RTP media are reflected in the output audio to keep the
-multiple audio sources in sync.
-
-=item B<--mix-method=>B<direct>|B<channels>
-
-Selects a method to mix multiple audio inputs into a single output file for
-B<mixed> output. The default is B<direct> which directly mixes all audio inputs
-together, producing a mixed output file with the same format as an audio file
-from a single input (B<output-single>) would be.
-
-The B<channels> mixing method puts each audio input into its own audio channel
-in the output file, therefore producing a multi-channel output file. Up to four
-separate RTP SSRCs are supported for a mixed output, which means that if each
-input is mono audio, then the mixed output file would contain 4 audio channels.
-This mixing method requires an output file format which supports these kinds of
-multi-channel audio formats (e.g. B<wav>).
-
-=item B<--mix-num-inputs=>I<INT>
-
-Change the number of recording channel in the output file. The value is between 1 to 4 (e.g. B<4>, which is also the default value).
-
-=item B<--output-chmod=>I<INT>
-
-Change the file permissions of recording files to the given mode. Must be given
-as an octal integer, for example B<0660>.
-
-=item B<--output-chmod-dir=>I<INT>
-
-Change the file permissions of recording files to the given mode. Must be given
-as an octal integer, for example B<0700> (which is also the default value).
-
-=item B<--output-chown=>I<USER>|I<UID>
-
-=item B<--output-chgrp=>I<GROUP>|I<GID>
-
-Change the ownership of recording files. Either user/group names or numeric IDs
-are supported. If the value is blank or given as B<-1> then the user/group is
-left unchanged.
-
-=item B<--mysql-host=>I<HOST>|I<IP>
-
-=item B<--mysql-port=>I<INT>
-
-=item B<--mysql-user=>I<USERNAME>
-
-=item B<--mysql-pass=>I<PASSWORD>
-
-=item B<--mysql-db=>I<STRING>
-
-Configuration for a MySQL storage backend. Details about calls and media files
-that are produced are stored into the database. Optionally the media files
-themselves can be stored as well (see B<output-storage>).
-
-=item B<--forward-to=>I<PATH>
-
-Forward raw RTP packets to a Unix socket. Disabled by default.
-
-=item B<--tls-send-to=>I<IP>B<:>I<PORT>
-
-=item B<--tls-resample=>I<INT>
-
-Send decoded audio over a TCP TLS connection to the specified destination.
-Audio is sent as raw mono 16-bit PCM in the given sample rate.
-
-=item B<--notify-uri=>I<URI>
-
-Enable HTTP notification about finished recordings to the specified URI, which
-must be an HTTP or HTTPS URI. Information about the finished recording is
-provided via custom HTTP headers, all of which use a prefix of B<X-Recording->.
-
-=item B<--notify-post>
-
-Use HTTP POST instead of GET for the HTTP notification requests. The request
-body is empty even if POST is used.
-
-=item B<--notify-no-verify>
-
-Disable TLS peer certificate verification for HTTPS requests.
-
-=item B<--notify-concurrency=>I<INT>
-
-The maximum number of HTTP requests to perform simultaneously.
-
-=item B<--notify-retries=>I<INT>
-
-How many times to retry a failed HTTP notification before giving up. An
-exponential falloff time is used for each subsequent attempt, starting with 5
-seconds.
-
-=item B<--notify-record>
-
-Attach recorded file to HTTP notification request. If enabled, notification
-request behaves as HTTP POST (ignoring B<--notify-post>). Note that this option
-is incompatible with DB-only storage as no recording file exists on storage
-(see B<output-storage>).
-
-=back
-
-=head1 EXIT STATUS
-
-=over
-
-=item B<0>
-
-Successful termination.
-
-=item B<1>
-
-An error occurred.
-
-=back
-
-=head1 FILES
-
-=over
-
-=item F</etc/rtpengine/rtpengine-recording.conf>
-
-Configuration file.
-
-=back
-
-=head1 SEE ALSO
-
-L<rtpengine(8)>.
diff --git a/docs/rtpengine.md b/docs/rtpengine.md
new file mode 100644
index 000000000..d5f8c0f1f
--- /dev/null
+++ b/docs/rtpengine.md
@@ -0,0 +1,1329 @@
+rtpengine(8) - 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.
+
+- **-c**, **--listen-cli=**\[_IP46_:\]_PORT_
+
+    TCP ip and port to listen for the CLI (command line interface).
+
+- **-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.
+
+- **--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).
diff --git a/docs/rtpengine.pod b/docs/rtpengine.pod
deleted file mode 100644
index 5dc098fb0..000000000
--- a/docs/rtpengine.pod
+++ /dev/null
@@ -1,1365 +0,0 @@
-=head1 NAME
-
-rtpengine - NGCP proxy for RTP and other UDP based media traffic
-
-=head1 SYNOPSIS
-
-B<rtpengine> B<--interface>=I<addr>... B<--listen-tcp>|B<--listen-udp>|B<--listen-ng>|B<--listen-tcp-ng>|B<--listen-http>|B<--listen-https>=I<addr>... [I<option>...]
-
-=head1 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.
-
-=head1 OPTIONS
-
-Most of these options are indeed optional, with two exceptions. It's
-mandatory to specify at least one local IP address through B<--interface>,
-and at least one of the B<--listen->I<...> options must be given.
-
-All options can (and should) be provided in a config file instead of
-at the command line. See the B<--config-file> option below for details.
-
-=over 4
-
-=item B<--help>
-
-Print the usage information.
-
-=item B<-v>, B<--version>
-
-If called with this option, the B<rtpengine> daemon will simply print its
-version number and exit.
-
-=item B<--codecs>
-
-Print a list of supported codecs and exit.
-
-=item B<--config-file=>I<FILE>
-
-Specifies the location of a config file to be used. The config file is an
-I<.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. B<table> instead of just B<t>) must be
-used in the config file.
-For boolean options that are either present or not (e.g. B<no-fallback>), a
-boolean value (either B<true> or B<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 L<INTERFACES> below for an example).
-
-As a special value, B<none> can be passed here to suppress loading of the
-default config file F</etc/rtpengine/rtpengine.conf>.
-
-=item B<--config-section=>I<STRING>
-
-Specifies the I<.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 B<rtpengine>.
-A config file section is started in the config file using square brackets
-(e.g. B<[rtpengine]>).
-
-=item B<-t>, B<--table=>I<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 F<README.md> for more detail.
-Optional and defaults to zero.
-If in-kernel operation is not desired, a negative number can be specified.
-
-=item B<-F>, B<--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.
-
-=item B<-S>, B<--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.
-
-=item B<-i>, B<--interface=>[I<NAME>B</>]I<IP>[B<!>I<IP>]
-
-Specifies a local network interface for RTP.
-At least one must be given, but multiple can be specified.
-See the section L<INTERFACES> just below for details.
-
-=item B<-l>, B<--listen-tcp=>[I<IP>B<:>]I<PORT>
-
-=item B<-u>, B<--listen-udp=>[I<IP46>B<:>]I<PORT>
-
-=item B<-n>, B<--listen-ng=>[I<IP46>B<:>]I<PORT>
-
-=item B<-n>, B<--listen-tcp-ng=>[I<IP46>B<:>]I<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 I<address:port>
-pair, separated by colon.
-At least one of these 3 options must be given.
-
-The B<tcp> protocol is obsolete.
-It was used by old versions of B<OpenSER> and its B<mediaproxy> module.
-It is provided for backwards compatibility.
-
-The B<udp> protocol is used by B<Kamailio>'s B<rtpproxy> module.
-In this mode, B<rtpengine> can be used as a drop-in replacement for any
-other compatible RTP proxy.
-
-The B<ng> protocol is an advanced control protocol and can be used with
-B<Kamailio>'s B<rtpengine> module.
-With this protocol, the complete SDP body is passed to B<rtpengine>,
-rewritten and passed back to B<Kamailio>.
-Several additional features are available with this protocol, such as
-ICE handling, SRTP bridging, etc.
-
-The B<tcp-ng> protocol is in fact the B<ng> protocol but transported over TCP.
-
-It is recommended to specify not only a local port number, but also
-B<127.0.0.1> as interface to bind to.
-
-=item B<-c>, B<--listen-cli=>[I<IP46>:]I<PORT>
-
-TCP ip and port to listen for the CLI (command line interface).
-
-=item B<-g>, B<--graphite=>I<IP46>:I<PORT>
-
-Address of the graphite statistics server.
-
-=item B<-w>, B<--graphite-interval=>I<INT>
-
-Interval of the time when information is sent to the graphite server.
-
-=item B<--graphite-prefix=>I<STRING>
-
-Add a prefix for every graphite line.
-
-=item B<--graphite-timeout=>I<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.
-
-=item B<-t>, B<--tos=>I<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 (B<Expedited Forwarding>).
-
-=item B<--control-tos=>I<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 B<rtpengine-ctl>.
-
-=item B<--control-pmtu=>B<want>|B<dont>
-
-Forces a specific PMTU discovery behaviour on IPv4 UDP control sockets,
-overriding the system-wide default. If set to B<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 B<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 B<dont> is useful in broken IPv4 environments without
-functioning PMTU discovery, for example in networks which unconditionally block
-all ICMP.
-
-=item B<-o>, B<--timeout=>I<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 B<rtpengine>'s internal state table.
-Defaults to 60 seconds.
-
-=item B<-s>, B<--silent-timeout=>I<SECS>
-
-Ditto as the B<--timeout> option, but applies to muted or inactive media
-streams.
-Defaults to 3600 (one hour).
-
-=item B<-a>, B<--final-timeout=>I<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.
-
-=item B<--offer-timeout=>I<SECS>
-
-This timeout (in seconds) is applied to calls which only had an B<offer>
-but no B<answer>.
-Defaults to 3600 (one hour).
-
-=item B<-p>, B<--pidfile=>I<FILE>
-
-Specifies a path and file name to write the daemon's PID number to.
-
-=item B<-f>, B<--foreground>
-
-If given, prevents the daemon from daemonizing, meaning it will stay in
-the foreground.
-Useful for debugging.
-
-=item B<-m>, B<--port-min=>I<INT>
-
-=item B<-M>, B<--port-max=>I<INT>
-
-Both take an integer as argument and together define the local port range
-from which B<rtpengine> will allocate UDP ports for media traffic relay.
-Default to 30000 and 40000 respectively.
-
-=item B<-L>, B<--log-level=>I<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 L<syslog(3)> man page.
-The default value is B<6>, equivalent to LOG_INFO.
-The highest possible value is B<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.
-
-=item B<--log-level->I<subsystem>B<=>I<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 B<log-level> setting described above, with the exception of the
-B<internals> subsystem which by default has all logging disabled.
-
-The full list of logging subsystems can be viewed by pulling up the B<--help>
-online help. Some (if not all) subsystems are: B<core>, B<spandsp> (messages
-generated by SpanDSP itself), B<ffmpeg> (messages generated by ffmpeg libraries
-themselves), B<transcoding> (messages related to RTP/media transcoding),
-B<codec> (messages related to codec negotiation), B<rtcp>, B<ice>, B<crypto>
-(messages related to crypto/SRTP/SDES/DTLS negotiation), B<srtp> (messages
-related to RTP/SRTP en/decryption), B<internals> (disabled by default), B<http>
-(includes WebSocket), B<control> (messages related to control protocols,
-including SDP exchanges), B<dtx>.
-
-=item B<--log-facilty=>B<daemon>|B<local0>|...|B<local7>|...
-
-The syslog facilty to use when sending log messages to the syslog daemon.
-Defaults to B<daemon>.
-
-=item B<--log-facilty-cdr=>B<daemon>|B<local0>|...|B<local7>|...
-
-Same as B<--log-facility> with the difference that only CDRs are written
-to this log facility.
-
-=item B<--log-facilty-rtcp=>B<daemon>|B<local0>|...|B<local7>|...
-
-Same as B<--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.
-
-=item B<--log-facilty-dtmf=>B<daemon>|B<local0>|...|B<local7>|...
-
-Same as B<--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.
-
-=item B<--log-format=>B<default>|B<parsable>
-
-Selects between multiple log output styles.
-The default is to prefix log lines with a description of the relevant
-entity, such as B<[CALLID]> or B<[CALLID port 12345]>.
-The B<parsable> output style is similar, but makes the ID easier to
-parse by enclosing it in quotes, such as B<[ID="CALLID"]>
-or B<[ID="CALLID" port="12345"]>.
-
-=item B<--dtmf-log-dest=>I<IP46>:I<PORT>
-
-Configures a target address for logging detected DTMF event. Similar
-to the feature enabled by B<--log-facilty-dtmf>, but instead of writing
-detected DTMF events to syslog, this sends the JSON payload to the
-given address as UDP packets.
-
-=item B<--dtmf-log-ng-tcp>
-
-If B<--listen-tcp-ng> is enabled, this will send DTMF events to all
-connected clients encoded in bencode format.
-
-=item B<--dtmf-no-log-injects>
-If B<--dtmf-no-log-injects> is enabled, DTMF events resulting from a
-call to inject-DTMF won't be sent to B<--dtmf-log-dest=> or B<--listen-tcp-ng>
-
-=item B<--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, B<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.
-
-=item B<--log-srtp-keys>
-
-Write SRTP keys to error log instead of debug log.
-
-=item B<-E>, B<--log-stderr>
-
-Log to stderr instead of syslog.
-Only useful in combination with B<--foreground>.
-
-=item B<--split-logs>
-
-Split multi-line log messages into individual log messages so that each
-line receives its own log line prefix.
-
-=item B<--max-log-line-length=>I<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.
-
-=item B<--no-log-timestamps>
-
-Don't add timestamps to log lines written to stderr.
-Only useful in combination with B<--log-stderr>.
-
-=item B<--log-name=>I<STRING>
-
-Set the id to be printed in syslog.
-Defaults to B<rtpengine>.
-
-=item B<--log-mark-prefix=>I<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.
-
-=item B<--log-mark-suffix=>I<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.
-
-=item B<--num-threads=>I<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.
-
-=item B<--media-num-threads=>I<INT>
-
-Number of threads to launch for media playback. Defaults to the same
-number as B<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.
-
-=item B<--thread-stack=>I<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.
-
-=item B<--evs-lib-path=>I<FILE>
-
-Points to the shared object file (B<.so>) containing the reference
-implementation for the EVS codec. See the F<README> for more details.
-
-=item B<--sip-source>
-
-The original B<rtpproxy> as well as older version of B<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 B<rtpengine> reverse this behaviour and honour the
-addresses given in the SDP body by default. This option restores the
-old behaviour.
-
-=item B<--dtls-passive>
-
-Enables the B<DTLS=passive> flag for all calls unconditionally.
-
-=item B<-d>, B<--delete-delay=>I<INT>
-
-Delete the call after the specified delay from memory.
-Can be set to zero for immediate call deletion.
-
-=item B<-r>, B<--redis=>[I<PW>B<@>]I<IP>B<:>I<PORT>B</>I<INT>
-
-Connect to specified Redis database (with the given database number) and
-use it for persistence storage.
-The format of this option is I<ADDRESS>:I<PORT>/I<DBNUM>, for example
-I<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 B<@> symbol, for example I<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 B<RTPENGINE_REDIS_AUTH_PW>.
-
-On startup, B<rtpengine> will read the contents of this database and
-restore all calls stored therein.
-During runtime operation, B<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, B<rtpengine> will delay startup until the
-Redis database adopts the master role (but see below).
-
-=item B<-w>, B<--redis-write=>[I<PW>B<@>]I<IP>B<:>I<PORT>B</>I<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 B<RTPENGINE_REDIS_WRITE_AUTH_PW>.
-
-When both options are given, B<rtpengine> will start and use the Redis
-database regardless of the database's role (master or slave).
-
-=item B<-k>, B<--subscribe-keyspace=>I<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 B<rtpengine-ctl ksadd/ksrm>.
-This may lead to enabling/disabling of the redis keyspace notification feature.
-
-=item B<--redis-num-threads=>I<INT>
-
-How many redis restore threads to create.
-The default is 4.
-
-=item B<--redis-expires=>I<INT>
-
-Expire time in seconds for redis keys.
-Default is 86400.
-
-=item B<--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 B<rtpengine>
-take control of that call. Without this option, an explicit command is required
-for B<rtpengine> to take (or relinquish) control of a call.
-
-=item B<-q>, B<--no-redis-required>
-
-When this parameter is present or B<NO_REDIS_REQUIRED='yes'> or B<'1'> in
-the config file, B<rtpengine> starts even if there is no initial connection
-to redis databases (either to B<-r> or to B<-w> or to both redis).
-
-Be aware that if the B<-r> redis cannot be initially connected, sessions
-are not reloaded upon rtpengine startup, even though rtpengine still starts.
-
-=item B<--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 B<rtpengine-ctl>.
-
-=item B<--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 B<rtpengine-ctl>.
-
-=item B<--redis-cmd-timeout=>I<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 B<rtpengine-ctl>; note that
-setting the parameter to 0 will require a reconnect on all configured
-redis servers.
-
-=item B<--redis-connect-timeout=>I<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 B<PING> command
-is issued before attempting to connect so the B<--redis-cmd-timeout> value
-will also be added to the total waiting time.
-This is useful if using B<--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 B<rtpengine-ctl>.
-
-=item B<-b>, B<--b2b-url=>I<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 B<%%> 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 B<xmlrpc-callback> B<ng> protocol option) will be used.
-
-=item B<-x>, B<--xmlrpc-format=>I<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.
-
-=item B<--max-sessions=>I<INT>
-
-Limit the number of maximum concurrent sessions.
-Set at startup via B<max-sessions> in config file.
-Set at runtime via B<rtpengine-ctl> util.
-Setting the B<rtpengine-ctl set maxsessions 0> can be used in draining
-rtpengine sessions.
-Enable feature: B<max-sessions=1000>
-Enable feature: B<rtpengine-ctl set maxsessions> >= 0
-Disable feature: B<rtpengine-ctl set maxsessions -1>
-By default, the feature is disabled (i.e. maxsessions == -1).
-
-=item B<--max-load=>I<FLOAT>
-
-If the current 1-minute load average exceeds the value given here,
-reject new sessions until the load average drops below the threshold.
-
-=item B<--max-cpu=>I<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 F</proc/stat>.
-
-=item B<--max-bandwidth=>I<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.
-
-=item B<--homer=>I<IP46>B<:>I<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.
-
-=item B<--homer-protocol=>B<udp>|B<tcp>
-
-Can be either B<udp> or B<tcp> with B<udp> being the default.
-
-=item B<--homer-id=>I<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.
-
-=item B<--recording-dir=>I<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.
-
-B<rtpengine> supports multiple mechanisms for recording calls.
-See B<recording-method> below for a list.
-The default recording method B<pcap> is described in this section.
-
-PCAP files will be stored within a F<pcap> subdirectory and metadata
-within a F<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.
-
-=item B<--recording-method=>B<pcap>|B<proc>|B<all>
-
-Multiple methods of call recording are supported and this option can be
-used to select one.
-Currently supported are the method B<pcap>, B<proc> and B<all>.
-The default method is B<pcap> and is the one described above.
-
-The recording method B<proc> works by writing metadata files directly into
-the B<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 F</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 B<all> will enable both B<pcap> and B<proc>
-at the same time.
-
-=item B<--recording-format=>B<raw>|B<eth>
-
-When recording to pcap file in B<raw> (default) format, there is no
-ethernet header.
-When set to B<eth>, a fake ethernet header is added, making each package
-14 bytes larger.
-
-=item B<--record-egress>
-
-Apply media recording to egress media streams (as they are sent by
-B<rtpengine>) instead of media streams as they are received. This makes it
-possible to include manipulated and generated media (such as from the B<play
-media> command) in the recordings.
-
-=item B<--iptables-chain=>I<STRING>
-
-This option enables explicit management of an iptables chain.
-When enabled, B<rtpengine> takes control of the given iptables chain,
-which must exist already prior to starting the daemon.
-Upon startup, B<rtpengine> will flush the chain, and then add one B<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 B<DROP> policy for
-the entire port range used by B<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 B<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 B<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.
-
-=item B<--scheduling=>B<default>|...
-
-=item B<--priority=>I<INT>
-
-=item B<--idle-scheduling=>B<default>|...
-
-=item B<--idle-priority=>I<INT>
-
-These options control various thread scheduling parameters.
-The B<scheduling> and B<priority> settings are applied to the main
-worker threads, while the B<idle-> versions of these settings are
-applied to various lower priority threads, such as timer runs.
-
-The B<scheduling> settings take the name of one of the supported
-scheduler policies.
-Setting it to B<default> or B<none> is equivalent to not setting the
-option at all and leaves the system default in place.
-The strings B<fifo> and B<rr> refer to realtime scheduling policies.
-B<other> is the Linux default scheduling policy.
-B<batch> is similar to B<other> except for a small wake-up scheduling
-penalty.
-B<idle> is an extremely low priority scheduling policy.
-The Linux-specific B<deadline> policy is not supported by B<rtpengine>.
-Not all systems necessarily supports all scheduling policies; refer to
-your system's sched(7) man page for details.
-
-The B<priority> settings correspond to the scheduling priority for
-realtime (B<fifo> or B<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
-B<priority> settings correspond to the B<nice> value and should be in
-the range of -20 (high) through 19 (low).
-Not all systems support thread-specific B<nice> values; on such a system,
-using these settings might have unexpected results.
-(Linux does support thread-specific B<nice> values.)
-Refer to your system's sched(7) man page.
-
-=item B<--mysql-host=>I<HOST>|I<IP>
-
-=item B<--mysql-port=>I<INT>
-
-=item B<--mysql-user=>I<USERNAME>
-
-=item B<--mysql-pass=>I<PASSWORD>
-
-Configuration for playing back media files that are stored in a
-B<MySQL> (or B<MariaDB>) database. At least B<mysql-host> must be configured
-for this to work. The others are optional and default to their respective
-values from the B<MySQL>/B<MariaDB> client library.
-
-=item B<--mysql-query=>I<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
-B<%llu> and must not contain any other format placeholders. The ID value
-passed to B<rtpengine> in the B<db-id> key of the B<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
-
-=item B<--endpoint-learning=>B<delayed>|B<immediate>|B<off>|B<heuristic>
-
-Chooses one of the available algorithms to learn RTP endpoint addresses. The
-legacy setting is B<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 B<immediate> learns the endpoint address from the
-first incoming packet seen without the 3-second delay. Using B<off> disables
-endpoint learning altogether, likely breaking clients behind NAT. The setting
-B<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.
-
-=item B<--jitter-buffer=>I<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.
-
-=item B<--jb-clock-drift>
-
-Enable clock drift compensation for the jitter buffer.
-
-=item B<--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.
-
-=item B<--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.
-
-=item B<--listen-http=>[I<IP>|I<HOSTNAME>B<:>]I<PORT>
-
-=item B<--listen-https=>[I<IP>|I<HOSTNAME>B<:>]I<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.
-
-=item B<--https-cert=>I<FILE>
-
-=item B<--https-key=>I<FILE>
-
-Provide a server certificate and corresponding private key for the HTTPS/WSS
-listener, in PEM format.
-
-=item B<--http-threads=>I<INT>
-
-Number of worker threads for HTTP/HTTPS/WS/WSS. If not specified, then the same
-number as given under B<num-threads> will be used. If no HTTP listeners are
-enabled, then no threads are created.
-
-=item B<--software-id=>I<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
-B<rtpengine->I<VERSION>. The string is sanitised to replace all
-non-alphanumeric characters with a dash to make it universally usable.
-
-=item B<--dtx-delay=>I<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.
-
-=item B<--max-dtx=>I<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.
-
-=item B<--dtx-buffer=>I<INT>
-
-=item B<--dtx-lag=>I<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 B<dtx-buffer> option
-limits the number of packets held in the DTX buffer, while the B<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 B<dtx-shift> milliseconds.
-
-The defaults are 10 packets and 100 milliseconds.
-
-=item B<--dtx-shift=>I<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.
-
-=item B<--dtx-cn-params=>I<INT>
-
-Specify one comfort noise parameter. This option follows the same format as
-B<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.
-
-=item B<--amr-dtx=>B<native>|B<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 B<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 B<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.
-
-=item B<--silence-detect=>I<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 B<cn-payload> option (see below). Currently this is applicable
-only to RTP peers that have advertised support for the B<CN> RTP payload type,
-in which case the silence audio frames will be replaced by B<CN> RTP frames.
-
-=item B<--cn-payload=>I<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
-B<--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.
-I<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 B<CN> packets generated from the silence
-detection mechanism described above. The configured CN parameters are used
-directly as payload of B<CN> packets sent by B<rtpengine>.
-
-The default values are 32 (-32 dBov) for the noise level and no spectral
-information.
-
-=item B<--player-cache>
-
-Enable caching of encoded media packets for media player. This is applicable
-for media playback initiated through the I<play media> command. When enabled
-B<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 I<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 I<start-pos> for playback with this
-option enabled.
-
-RTP data is cached and retained in memory for the lifetime of the process.
-
-=item B<audio-buffer-length=>I<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 B<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 B<.wav> files this is 256 ms (2048 samples). Therefore 500 ms is the
-recommended value.
-
-=item B<audio-buffer-delay=>I<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.
-
-=item B<audio-player=>B<on-demand>|B<play-media>|B<transcoding>|B<always>
-
-Define when to enable the audio player if not explicitly instructed otherwise.
-The default setting is B<on-demand>.
-
-Enabling the audio player for a party to a call makes B<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 B<play media>
-command.
-
-With this set to B<on-demand>, the audio player is enabled only if explicitly
-requested by the user for a particular call via the B<audio-player=> option
-used in a signalling message.
-
-When set to B<play-media>, the audio player is enabled only while media
-playback via the B<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 B<transcoding> leaves the audio player disabled unless
-any sort of transcoding is required for a call.
-
-With a setting of B<always>, the audio player is enabled for all calls, unless
-explicitly disabled via the B<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
-B<audio-player=> option in a signalling message. This option supports the
-values B<transcoding> and B<always>, which result in the behaviour described
-just above, and B<off> which forces the audio player to be disabled regardless
-of this setting.
-
-=item B<--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).
-
-=item B<--dtls-cert-cipher=>B<prime256v1>|B<RSA>
-
-Choose the type of key to use for the signature used by the self-signed
-certificate used for DTLS. The previous default was B<RSA>. The current default
-and the only other option is B<prime256v1> which is a 256-bit elliptic-curve
-key.
-
-=item B<--dtls-signature=>B<SHA-256>|B<SHA-1>
-
-Choose the hash algorithm to use for the signature used by the self-signed
-certificate used for DTLS. The default is B<SHA-256>. Not to be confused with
-the hash algorithm used for the certificate fingerprint inserted into the SDP
-(B<a=fingerprint:>), which is independent of the certificate's signature and
-can be selected during runtime.
-
-=item B<--dtls-rsa-key-size=>I<INT>
-
-Size in bits of the RSA key used by the DTLS certificate, if RSA is in use.
-Default is 2048 bits.
-
-=item B<--dtls-ciphers=>I<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
-B<DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK>
-
-=item B<--dtls-mtu=>I<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.
-
-=item B<--mqtt-host=>I<HOST>|I<IP>
-
-Host or IP address of the Mosquitto broker to connect to. Must be set to enable
-exporting stats to Mosquitto.
-
-=item B<--mqtt-port=>I<INT>
-
-Port of the Mosquitto broker. Defaults to 1883.
-
-=item B<--mqtt-id=>I<STRING>
-
-Client ID to use for Mosquitto. Default is a generated random string.
-
-=item B<--mqtt-keepalive=>I<INT>
-
-Keepalive interval in seconds. Defaults to 30.
-
-=item B<--mqtt-user=>I<USERNAME>
-
-=item B<--mqtt-pass=>I<PASSWORD>
-
-Credentials to connect to Mosquitto broker. At least a username must be given
-to enable authentication.
-
-=item B<--mqtt-cafile=>I<FILE>
-
-=item B<--mqtt-capath=>I<PATH>
-
-=item B<--mqtt-certfile=>I<FILE>
-
-=item B<--mqtt-keyfile=>I<FILE>
-
-=item B<--mqtt-tls-alpn=>I<STRING>
-
-Enable TLS to connect to Mosquitto broker, optionally with client certificate
-authentication. At least B<cafile> or B<capath> must be given to enable TLS. To
-enable client certificate authentication, both B<certfile> and B<keyfile> must
-be set. All files must be in PEM format. Password-proteted files are not
-supported. The B<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.
-
-=item B<--mqtt-publish-qos=>B<0>|B<1>|B<2>
-
-QoS value to use for publishing to Mosquitto. See Mosquitto docs for details.
-
-=item B<--mqtt-publish-topic=>I<STRING>
-
-Topic string to use for publishing to Mosquitto. Must be set to a non-empty
-string.
-
-=item B<--mqtt-publish-interval=>I<MILLISECONDS>
-
-Interval in milliseconds to publish to Mosquitto. Defaults to 5000 (5 seconds).
-
-=item B<--mqtt-publish-scope=>B<global>|B<summary>|B<call>|B<media>
-
-When set to B<summary>, one message will be published to Mosquitto every
-I<interval> milliseconds containing all global stats. A setting of B<global>
-has the same effect as B<summary> but will also contain a list of all running
-calls with stats for each call. When set to B<call>, one message per call will
-be published to Mosquitto with stats for that call every I<interval>
-milliseconds, plus one message every I<interval> milliseconds with global
-stats. When set to B<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 I<interval> milliseconds, plus one message
-every I<interval> milliseconds with global stats.
-
-=item B<--mos=>B<CQ>|B<LQ>
-
-MOS (Mean Opinion Score) calculation formula. Defaults to B<CQ> (conversational
-quality) which takes RTT into account and therefore requires peers to correctly
-send RTCP. If set to B<LQ> (listening quality) RTT is ignored, allowing a MOS to
-be calculated in the absence of RTCP.
-
-=item B<--measure-rtp>
-
-Enable measuring RTP metrics even for plain RTP passthrough scenarios. Without
-that option, RTP metrics are measured only in transcoding scenarios.
-
-=item B<--socket-cpu-affinity=>I<INT>
-
-Enables setting the socket CPU affinity via the B<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 B<8> then the values B<0> through B<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.
-
-=back
-
-=head1 INTERFACES
-
-The command-line options B<-i> or B<--interface>, or equivalently the
-B<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 [I<NAME>B</>]I<IP>[!I<IP>] with I<IP> being
-either an IPv4 address, an IPv6 address, the name of a system network interface
-(such as I<eth0>), a DNS host name (such as I<test.example.com>), or B<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 B<rtpengine> (as described below).
-The I<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 I<IP> token in the syntax above.
-For example, to configure a logical network interface called I<int>
-using all the addresses from the existing system network interface
-I<eth0>, you would use the syntax I<int/eth0>.
-(Unless omitted, the second I<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 B<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 B<-i> or B<--interface> options.
-When using the config file, only use a single B<interface> line,
-but specify multiple values separated by semicolons (e.g.
-I<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 B<rtpengine> behaves as
-if multiple B<--interface> options had been specified.
-For example, if interface I<eth0> exists with both addresses
-I<192.168.1.120> and I<2001:db8:85a3::7334> configured on it, and if
-the option I<--interface=ext/eth0> is given, then B<rtpengine> would
-behave as if both options I<--interface=ext/192.168.1.120> and
-I<--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, I<--interface=10.65.76.2!192.0.2.4> means that I<10.65.76.2>
-is the actual local address on the server, but outgoing SDP bodies should
-advertise I<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 I<\!>.
-
-Giving an interface a name (separated from the address by a slash) is
-optional; if omitted, the name B<default> is used.
-Names are useful to create logical interfaces which consist of one or
-more local addresses.
-It is then possible to instruct B<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 I<10.35.2.75>, while clients coming from
-the public internet must communicate with your other local address
-I<192.0.2.67>, you could create one logical interface I<pub> and a second
-one I<priv> by using I<--interface=pub/192.0.2.67 --interface=priv/10.35.2.75>.
-You can then use the B<direction> option to tell B<rtpengine> which local
-address to use for which endpoints (either I<pub> or I<priv>).
-
-If multiple logical interfaces are configured, but the B<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: I<--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 B<default>.
-You would use the B<address family> option to tell B<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 B<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 I<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:
-I<--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.
-I<direction=pub:1>), but it is now also possible to reference only
-the base name (i.e. I<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 I<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 I<direction=pub> is IPv4 is needed as a primary
-address, either I<pub:1>, I<pub:2>, or I<pub:4> might be selected.
-When I<pub:1> is selected, one IPv4 and one IPv6 address will be used
-as additional ICE alternatives.
-For I<pub:2>, only one IPv6 is used as ICE alternative, and for I<pub:4>
-no alternatives would be used.
-When IPv6 is needed as a primary address, either I<pub:1>, I<pub:2>, or
-I<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 B<direction>
-is not given.
-If the first given interface has the I<BASE:SUFFIX> format then the
-round-robin algorithm is used and will select interfaces with the
-same I<BASE> name.
-
-If you are not using the NG protocol but rather the legacy UDP protocol
-used by the B<rtpproxy> module, the interfaces must be named B<internal>
-and B<external> corresponding to the B<i> and B<e> flags if you wish to
-use network bridging in this mode.
-
-=head1 EXIT STATUS
-
-=over
-
-=item B<0>
-
-Successful termination.
-
-=item B<1>
-
-An error occurred.
-
-=back
-
-=head1 ENVIRONMENT
-
-=over
-
-=item B<RTPENGINE_REDIS_AUTH_PW>
-
-Redis server password for persistent state storage.
-
-=item B<RTPENGINE_REDIS_WRITE_AUTH_PW>
-
-Redis server password for write operations, if B<--redis> has been
-specified, in which case the one specified in B<--redis> will be used for
-read operations only.
-
-=back
-
-=head1 FILES
-
-=over
-
-=item F</etc/rtpengine/rtpengine.conf>
-
-Configuration file.
-
-=back
-
-=head1 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
-
-=head1 SEE ALSO
-
-L<kamailio(8)>.
diff --git a/docs/usage.md b/docs/usage.md
index d32cea907..7eacb60a7 100644
--- a/docs/usage.md
+++ b/docs/usage.md
@@ -4,7 +4,7 @@
 
 The options are described in detail in the rtpengine(1) man page. If you're
 reading this on Github, you can view the current master's man page
-[here](https://github.com/sipwise/rtpengine/blob/master/daemon/rtpengine.pod).
+[here](https://github.com/sipwise/rtpengine/blob/master/docs/rtpengine.md).
 
 ## In-kernel Packet Forwarding
 
diff --git a/lib/common.Makefile b/lib/common.Makefile
index e0126eca7..04481604b 100644
--- a/lib/common.Makefile
+++ b/lib/common.Makefile
@@ -34,12 +34,11 @@ $(DAEMONSRCS) $(HASHSRCS):	$(patsubst %,../daemon/%,$(DAEMONSRCS)) $(patsubst %,
 		echo '#line 1' && \
 		cat ../daemon/"$@" ) > "$@"
 
-%.8: %.pod
-	pod2man \
-		--center="NGCP rtpengine" \
+%.8: %.8.ronn
+	ronn -r \
+		--organization="NGCP rtpengine" \
 		--date="$(RELEASE_DATE)" \
-		--release="$(RTPENGINE_VERSION)" \
-		$< $@
+		$<
 
 resample.c codeclib.strhash.c mix.c:	fix_frame_channel_layout.h
 
diff --git a/recording-daemon/Makefile b/recording-daemon/Makefile
index 4caabc100..543b2d89c 100644
--- a/recording-daemon/Makefile
+++ b/recording-daemon/Makefile
@@ -38,8 +38,8 @@ LIBSRCS=	loglib.c auxlib.c rtplib.c codeclib.strhash.c resample.c str.c socket.c
 LIBASM=		mvr2s_x64_avx2.S mvr2s_x64_avx512.S
 OBJS=		$(SRCS:.c=.o) $(LIBSRCS:.c=.o) $(LIBASM:.S=.o)
 
-PODS=		rtpengine-recording.pod
-MANS=		$(PODS:.pod=.8)
+MDS=		rtpengine-recording.ronn
+MANS=		$(MDS:.ronn=.8)
 
 include ../lib/common.Makefile
 
diff --git a/recording-daemon/rtpengine-recording.8.ronn b/recording-daemon/rtpengine-recording.8.ronn
new file mode 120000
index 000000000..1dcc4538b
--- /dev/null
+++ b/recording-daemon/rtpengine-recording.8.ronn
@@ -0,0 +1 @@
+../docs/rtpengine-recording.md
\ No newline at end of file
diff --git a/recording-daemon/rtpengine-recording.pod b/recording-daemon/rtpengine-recording.pod
deleted file mode 120000
index 0a3a1b491..000000000
--- a/recording-daemon/rtpengine-recording.pod
+++ /dev/null
@@ -1 +0,0 @@
-../docs/rtpengine-recording.pod
\ No newline at end of file