# The *NG* Control Protocol

In order to provide several advanced features in *rtpengine*, a new advanced
control protocol has been devised, which passes the complete SDP body from the
SIP proxy to the *rtpengine* daemon. The SDP body gets rewritten by the daemon
and then passed back to the SIP proxy in order to embed it into the SIP message.

This control protocol is supported over a number of different transports (plain
UDP, plain TCP, HTTP, WebSocket) and loosely follows the same format as used by
the module. Each message passed between the SIP and the media proxy consists
of two parts separated by a single space:

* a unique message cookie
* a dictionary document

The message cookie is used to match requests to responses and to detect retransmissions.
The message cookie in the response must be the same as in the request it's dedicated to.

The dictionary document can be in one of two formats:
* a JSON object
* a dictionary in [bencode](http://en.wikipedia.org/wiki/Bencode) format

The *bencoding* mechanism supports a subset of JSON features, for example:
* dictionaries/hashes
* lists/arrays
* arbitrary byte strings

On the other hand, it offers some benefits over JSON encoding, e.g. simpler and more
efficient encoding, less encoding overhead, deterministic encoding,
faster encoding and decoding.

The disadvantages compared to JSON are that it's not readily a human readable
format and sometimes it might be difficult to support it in programming languages.

Internally *rtpengine* uses the *bencoding* mechanism natively, leading to additional
overhead when JSON is in use as it has to be converted.

The dictionary of each request must contain at least one key called `command`.
The corresponding value must be a string and determines the type of message.
Currently the following commands are defined:

* ping
* offer
* answer
* delete
* query
* start recording
* stop recording
* pause recording
* block DTMF
* unblock DTMF
* block media
* unblock media
* silence media
* unsilence media
* start forwarding
* stop forwarding
* play media
* stop media
* play DTMF
* statistics
* publish
* subscribe request
* subscribe answer
* unsubscribe
* connect

The response dictionary must contain at least one key called `result`.
The value can be either `ok` or `error`.

If the result is `error`, then another key `error-reason` must be given,
containing a string with a human-readable error message. No other keys should
be present in the error case.

If the result is `ok`, the optional key `warning` may be present, containing a
human-readable warning message. This can be used for non-fatal errors.

For the `ping` command, the additional value `pong` is allowed.

For readability all data objects below are represented in a JSON-like format
and without the message cookie. For example, the `ping` message and
its corresponding `pong` reply would be written as:

	{ "command": "ping" }
	{ "result": "pong" }

While the actual bencode encoded messages, including the message cookie,
might look like this:

	5323_1 d7:command4:pinge
	5323_1 d6:result4:ponge

All keys and values are case-sensitive unless specified otherwise.
The bencode standard's requirement that dictionary keys must be presented in the lexicographical order
is currently not honored.

The *NG* protocol that is used by the module utilises the *bencoding* mechanism
and the UDP transport by default, or, alternatively the websocket transport if enabled.

Of course the agent controlling *rtpengine* using the *NG* protocol does not have
to be a SIP proxy (e.g. kamailio). Any process that involves SDP can potentially talk to
*rtpengine* using this protocol.

As mentioned already, each NG-protocol message can include optional flags in order
to cause specific behavior for this particular SDP offer/answer
(e.g. transport, transcoding, preferred encryption parameters etc.)

The parsing of option flags (sometimes also called *rtpp flags*) can be done:
* by remote SIP proxy (e.g. kamailio)
* by rtpengine itself

*NOTE: currently parsing on the daemon side is implemented, but not all control agents
*may support it. As of the time of writing only the kamailio module uses it.*

The difference between two approaches is that in the first case, the parsing of flags
is done with help of module, meanwhile in the second case a list of flags is passed
to rtpengine using bencode string format and is then parsed here.
The benefit of the second approach is that any new flags supported by rtpengine
will automatically be supported without having to worry about support in the control module.

When the flags are passed to rtpengine, they are formated as following:

	{ "rtpp_flags": "replace-origin via-branch=auto-next strict-source label=callee OSRTP-accept transport-protocol=RTP/AVP address-family=IP4" }

Lists and dictionaries are supported in this format using square brackets `[ ]`, for example:

	{ "rtpp_flags": "via-branch=auto-next OSRTP=[accept] codec=[transcode=[PCMA PCMU] accept=[AMR-WB AMR] strip=[EVS]]" }

Regardless whether the flags parsing is done by the module or daemon,
a functional behavior remains the same and has no difference in terms of SDP processing.

## Messages description

## `ping` Message

The request dictionary contains no other keys and the reply dictionary also contains no other keys. The
only valid value for `result` is `pong`.

## `offer` Message

The request dictionary must contain at least the following keys:

* `sdp`

  Contains the complete SDP body as string.

* `call-id`

  The SIP call ID as string.

* `from-tag`

  The SIP `From` tag as string.

Optionally included keys are:

* `all`

	Can be set to the string `none` to disable any extra behaviour (which
	is the default if this key is omitted altogether) or to one of `all`,
	`offer-answer`, `except-offer-answer` or `flows`.  Applicable to
	certain messages only. The behaviour is explained below separately for
	each affected message.

* `address family`

	A string value of either `IP4` or `IP6` to select the primary address family in the substituted SDP
	body. The default is to auto-detect the address family if possible (if the receiving end is known
	already) or otherwise to leave it unchanged.

* `audio player`

    Contains a string value of either `default`, `transcoding`, `off`, or `always`.

    The values `transcoding` and `always` result in the behaviour described
    under the `audio-player` config option in the manual, and override the
    global setting from the config file. The value `off` disables usage of the
    audio player regardless of the global config setting. The option `default`
    results in the behaviour mandated by the global config setting.

* `delay-buffer`

	Takes an integer as value. When set to non-zero, enables the delay
	buffer when setting up codec handlers. The delay buffer delays all
	media by the given number of milliseconds before passing it on. Once
	the delay buffer is configured, it must explicitly be disabled again by
	setting this value to zero. The delay buffer setting is honoured in all
	messages that set up codec handlers, such as `block DTMF`.

* `direction`

	Contains a list of two strings and corresponds to the *rtpproxy* `e` and `i` flags. Each element must
	correspond to one of the named logical interfaces configured on the
	command line (through `--interface`). For example, if there is one logical interface named `pub` and
	another one named `priv`, then if side A (originator of the message) is considered to be
	on the private network and side B (destination of the message) on the public network, then that would
	be rendered within the dictionary as:

		{ ..., "direction": [ "priv", "pub" ], ... }

	This only needs to be done for an initial `offer`; for the `answer` and any subsequent offers (between
	the same endpoints) *rtpengine* will remember the selected network interface.

	As a special case to support legacy usage of this option, if the given interface names are
	`internal` or `external` and if no such interfaces have been configured, then they're understood as
	selectors between IPv4 and IPv6 addresses.
	However, this mechanism for selecting the address family is now obsolete
	and the `address family` dictionary key should be used instead.

	For legacy support, the special direction keyword `round-robin-calls` can be used to invoke the
	round-robin interface selection algorithm described in the section *Interfaces configuration*.
	If this special keyword is used, the round-robin selection will run over all configured
	interfaces, whether or not they are configured using the `BASE:SUFFIX` interface name notation.
	This special keyword is provided only for legacy support and should be considered obsolete.
	It will be removed in future versions.

    For commands that require only one interface (e.g. `publish`), use the
    `interface=...` key. For commands that require two interfaces, as an
    alternative to the `direction=` key, the two interfaces can be listed
    separately, using `from-interface=...` for the first interface and
    `to-interface=...` for the second one.

* `digit` or `code`

	Sets the replacement digit for `DTMF-security=DTMF`.

* `drop-traffic`

	Contains a string, valid values are `start` or `stop`.

	`start` signals to *rtpengine* that all RTP involved in this call is dropped.
	Can be present either in `offer` or `answer`, the behavior is for the entire call.

	`stop` signals to *rtpengine* that all RTP involved in this call is NOT dropped anymore.
	Can be present either in `offer` or `answer`, the behavior is for the entire call.

	`stop` has priority over `start`, if both are present.

* `DTLS`

	Contains a string and influences the behaviour of DTLS-SRTP. Possible values are:

	- `off` or `no` or `disable`

		Prevents *rtpengine* from offering or acceping DTLS-SRTP when otherwise it would. The default
		is to offer DTLS-SRTP when encryption is desired and to favour it over SDES when accepting
		an offer.

	- `passive`

		Instructs *rtpengine* to prefer the passive (i.e. server) role for the DTLS
		handshake. The default is to take the active (client) role if possible. This is useful in cases
		where the SRTP endpoint isn't able to receive or process the DTLS handshake packets, for example
		when it's behind NAT or needs to finish ICE processing first.

	- `active`

		Reverts the `passive` setting. Only useful if the `dtls-passive` config option is set.

* `DTLS-reverse`

	Contains a string and influences the behaviour of DTLS-SRTP. Unlike the regular `DTLS` flag, this one
	is used to control behaviour towards DTLS that was offered to *rtpengine*. In particular, if `passive`
	mode is used, it prevents *rtpengine* from prematurely sending active DTLS connection attempts.
	Possible values are:

	- `passive`

		Instructs *rtpengine* to prefer the passive (i.e. server) role for the DTLS
		handshake. The default is to take the active (client) role if possible. This is useful in cases
		where the SRTP endpoint isn't able to receive or process the DTLS handshake packets, for example
		when it's behind NAT or needs to finish ICE processing first.

	- `active`

		Reverts the `passive` setting. Only useful if the `dtls-passive` config option is set.

* `DTLS-fingerprint`

	Contains a string and is used to select the hashing function to generate the DTLS fingerprint
	from the certificate. The default is SHA-256, or the same hashing function as was used by the
	peer. Available are `SHA-1`, `SHA-224`, `SHA-256`, `SHA-384`, and `SHA-512`.

* `DTMF-security`

	Used in the `block DTMF` message to select the DTMF blocking mode. The
	default mode is `drop` which simply drops DTMF event packets. The other
	supported modes are: `silence` which replaces DTMF events with silence
	audio; `tone` which replaces DTMF events with a single sine wave tone;
	`random` which replaces DTMF events with random other DTMF events (both
	in-band DTMF audio tones and RFC event packets); `zero` which is
	similar to `random` except that a zero event is always used; `DTMF`
	which is similar to `zero` except that a different DTMF digit can be
	specified; `off` to disable DTMF blocking.

* `DTMF-security-trigger`

	Blocking mode to enable when the DTMF `trigger` (see below) is detected.

* `DTMF-security-trigger-end`

	Blocking mode to enable when the DTMF `end trigger` (see below) is
	detected.

* `DTMF-delay`

	Time in milliseconds to delay DTMF events (both RFC event packets and
	DTMF tones) for. With this option enabled (set to non-zero), DTMF
	events are initially replaced by silence and then subsequently
	reproduced after the given delay. DTMF blocking modes are honoured at
	the time when the DTMF events are reproduced.

* `DTMF-log-dest`

    Contains a destination address and port for the DTMF logging feature. This
    overrides the global destination from the `dtmf-log-dest` config option on
    a per-call basis. Even if the global config option is unset, setting the
    destination address/port via this option enables DTMF logging for this
    call.

* `endpoint-learning`

	Contains one of the strings `off`, `immediate`, `delayed` or `heuristic`. This tells rtpengine which endpoint learning algorithm to use and overrides the `endpoint-learning` configuration option. This option can also be put into the `flags` list using a prefix of `endpoint-learning-`.

* `from-interface`

    Contains a string identifying the network interface pertaining to the
    "received from" direction of this message. Identical to setting the first
    `direction=` value.

* `frequency` or `frequencies`

    Sets the tone frequency or frequencies for `DTMF-security=tone` in Hertz.
    The default is a single frequency of 400 Hz. A list of frequencies can be
    given either as a list object, or as a string containing a comma-separated
    list of integers. The given frequencies will be picked from the list in
    order, one for each DTMF event detected, and will be repeated once the end
    of the list is reached.

* `from-tags`

	Contains a list of strings used to selected multiple existing call
	participants (e.g. for the `subscribe request` message). An alternative
	way to list multiple tags is by putting them into the `flags` list,
	each prefixed with `from-tags-`.

* `generate RTCP`

	Contains a string, either `on` or `off`. If enabled for a call,
	received RTCP packets will not simply be passed through as usual, but
	instead will be consumed, and instead *rtpengine* will generate its own
	RTCP packets to send to the RTP peers. This flag will be effective for
	both sides of a call.

* `ICE`

	Contains a string which must be one of the following values:

	With `remove`, any ICE attributes are stripped from the SDP body. Also
	see the flag `reject ICE` to effect an early removal of ICE support
	during an `offer`.

	With `force`, ICE attributes are first stripped, then new attributes are
	generated and inserted, which leaves the media proxy as the only ICE candidate.

	With `default`, the behaviour will be the same as with `force` if the incoming SDP already
	had ICE attributes listed. If the incoming SDP did not contain ICE attributes, then no
	ICE attributes are added.

	With `force-relay`, existing ICE candidates are left in place except `relay`
	type candidates, and *rtpengine* inserts itself as a `relay` candidate. It will also leave SDP
	c= and m= lines unchanged.

	With `optional`, if no ICE attributes are present, a new set is generated and the
	media proxy lists itself as ICE candidate; otherwise, the media proxy inserts itself as a
	low-priority candidate. This used to be the default behaviour in previous versions of
	*rtpengine*.

	The default behaviour (no `ICE` key present at all) is the same as `default`.

	This flag operates independently of the `replace` flags.

	Note that if config parameter `save-interface-ports = true`, ICE will be broken, because
	rtpengine will bind ports only on the first local interface of desired family of logical interface.

* `ICE-lite`

	Contains a string which must be one of the following values:

	- `forward` to enable "ICE lite" mode towards the peer that this offer is sent to.

	- `backward` to enable "ICE lite" mode towards the peer that has sent this offer.

	- `both` to enable "ICE lite" towards both peers.

	- `off` to disable "ICE lite" towards both peers and revert to full ICE support.

	The default (keyword not present at all) is to use full ICE support, or to leave the previously
	set "ICE lite" mode unchanged. This keyword is valid in `offer` messages only.

* `interface`

	Contains a single string naming one of the configured interfaces, just like `direction` does. The
	`interface` option is used instead of `direction` where only one interface is required (e.g. outside
	of an offer/answer scenario), for example in the `publish` or `subscribe request` commands.

* `label` or `from-label`

	A custom free-form string which *rtpengine* remembers for this participating endpoint and reports
	back in logs and statistics output. For some commands (e.g. `block media`) the given label is not
	used to set the label of the call participant, but rather to select an existing call participant.

* `media address`

	This can be used to override both the addresses present in the SDP body
	and the `received from` address. Contains either an IPv4 or an IPv6 address, expressed as a simple
	string. The format must be dotted-quad notation for IPv4 or RFC 5952 notation for IPv6.
	It's up to the RTP proxy to determine the address family type.

* `media echo` or `media-echo`

	Contains a string to enable a special media echo mode. Recognised values are:

	- `blackhole` or `sinkhole`

		Media arriving from either side of the call is simply discarded
		and not forwarded.

	- `forward`

		Enables media echo towards the receiver of this message (e.g.
		the called party if the message is an `offer` from the caller).
		Media arriving from that side is echoed back to its sender
		(with a new SSRC if it's RTP). Media arriving from the opposite
		side is discarded.

	- `backwards`

		Enables media echo towards the sender of this message (i.e. the
		opposite of `forward`). Media arriving from the other side is
		discarded.

	- `both`

		Enables media echo towards both the sender and the receiver of
		this message.

* `metadata`

	This is a generic metadata string. The metadata will be written to the bottom of
	metadata files within `/path/to/recording_dir/metadata/` or to
	`recording_metakeys` table.  In the latter case, `metadata` string must
	contain a list of `key:val` pairs separated by `|` character.  `metadata` can be used to
	record additional information about recorded calls. `metadata` values passed in
	through subsequent messages will overwrite previous metadata values.

	See the `--recording-dir` option above.

* `OSRTP`

	Similar to `SDES` but controls OSRTP behaviour. Default behaviour is to pass through
	OSRTP negotiations. Supported options:

	- `offer` or `offer-RFC`

        When processing a non-OSRTP offer, convert it to an OSRTP offer. Will
        result in RTP/SRTP transcoding if the OSRTP offer is accepted. The
        transport protocol should be a non-SRTP (plain RTP) protocol such as
        RTP/AVP.

    - `offer-legacy`

        Convert a regular offer to a legacy, non-RFC "best effort" SRTP offer,
        which involves duplicating each SDP media section in the output,
        advertised once as plain RTP and once as SRTP. The transport protocol
        should be set to an SRTP protocol such as RTP/SAVP. To enable full
        interoperability with endpoints which support this usage, the flag
        `accept-legacy` (see below) should also be given in all signalling
        exchanges.

	- `accept-RFC`

		When processing a non-OSRTP answer in response to an OSRTP offer, accept the
		OSRTP offer anyway. Results in RTP/SRTP transcoding.

    - `accept-legacy`

        Enables support for legacy, non-RFC "best effort" SRTP offers, which
        consist of media sections being advertised twice, once as plain RTP and
        once as SRTP. With this option set, *rtpengine* will treat such SDPs as
        SRTP SDPs, removing the duplicated media sections. This flag must be
        given for both offer and answer messages.

    - `accept`

        Short for both `accept-RFC` and `accept-legacy`. Can be used
        unconditionally in all signalling if so desired.

* `output-destination`

    See `start recording` below.

* `ptime`

	Contains an integer. If set, changes the `a=ptime` attribute's value in the outgoing
	SDP to the provided value. It also engages the transcoding engine for supported codecs
	to provide repacketization functionality, even if no additional codec has actually
	been requested for transcoding. Note that not all codecs support all packetization
	intervals.

	The selected ptime (which represents the duration of a single media packet in milliseconds)
	will be used towards the endpoint receiving this offer, even if the matching answer
	prefers a different ptime.

	This option is ignored in `answer` messages. See below for the reverse.

* `ptime-reverse`

	This is the reciprocal to `ptime`. It sets the ptime to be used towards the endpoint
	who has sent the offer. It will be inserted in the `answer` SDP. This option is also
	ignored in `answer` messages.

* `received from`

	Contains a list of exactly two elements. The first element denotes the address family and the second
	element is the SIP message's source address itself. The address family can be one of `IP4` or `IP6`.
	Used if SDP addresses are neither trusted (through `SIP source address` or `--sip-source`) nor the
	`media address` key is present.

* `record call`

	Contains one of the strings `yes`, `no`, `on` or `off`. This tells rtpengine
	whether or not to record the call to PCAP files. If the call is recorded, it
	will generate PCAP files for each stream and a metadata file for each call.
	Note that rtpengine *will not* force itself into the media path, and other
	flags like `ICE=force` may be necessary to ensure the call is recorded.

	See the `--recording-dir` option above.

	Enabling call recording via this option has the same effect as doing it separately
	via the `start recording` message, except that this option guarantees that the
	entirety of the call gets recorded, including all details such as SDP bodies
	passing through *rtpengine*.

* `rtcp-mux`

	A list of strings controlling the behaviour regarding rtcp-mux (multiplexing RTP and RTCP on a single
	port, RFC 5761). The default behaviour is to go along with the client's preference. The list can contain
	zero of more of the following strings. Note that some of them are mutually exclusive.

	- `offer`

		Instructs *rtpengine* to always offer rtcp-mux, even if the client itself doesn't offer it.

	- `require`

		Similar to `offer` but pretends that the receiving client has already accepted rtcp-mux.
		The effect is that no separate RTCP ports will be advertised, even in an initial offer
		(which is against RFC 5761). This option is provided to talk to WebRTC clients.

	- `demux`

		If the client is offering rtcp-mux, don't offer it to the other side, but accept it back to
		the offering client.

	- `accept`

		Instructs *rtpengine* to accept rtcp-mux and also offer it to the other side if it has been
		offered.

	- `reject`

		Reject rtcp-mux if it has been offered. Can be used together with `offer` to achieve the opposite
		effect of `demux`.

* `SIP message type`

    Contains a string indicating whether the SIP message that triggered this
    signalling message was either a `SIP request` or a `SIP response`.

* `SIP code`

    Contains an integer corresponding to the SIP response code (e.g. 180 or
    200) if this signalling message was triggered by a SIP response.

* `template`

    Contains the name of a signalling template to be used for this particular
    control message. See documentation for *SIGNALLING TEMPLATES* in the man
    page.

* `via-branch`

	The SIP `Via` branch as string. Used to additionally refine the matching logic between media streams
	and calls and call branches.

* `set-label`

	Some commands (e.g. `block media`) use the given `label` to select
	an existing call participant. For these commands, `set-label` instead
	of `label` can be used to set the label at the same time, either for
	the selected call participant (if selected via `from-tag`) or for the
	newly created participant (e.g. for `subscribe request`).

* `SDES`

	A list of strings controlling the behaviour regarding SDES. The default is to offer SDES without any
	session parameters when encryption is desired, and to accept it when DTLS-SRTP is unavailable. If two
	SDES endpoints are connected to each other, then the default is to offer SDES with the same options
	as were received from the other endpoint. Additionally, all other supported SDES crypto suites are
	added to the outgoing offer by default.

	These options can also be put into the `flags` list using a prefix of `SDES-`. All options controlling
	SDES session parameters can be used either in all lower case or in all upper case.

	- `off` or `no` or `disable`

		Prevents *rtpengine* from offering SDES, leaving DTLS-SRTP as the other option.

	- `unencrypted_srtp`, `unencrypted_srtcp` and `unauthenticated_srtp`

		Enables the respective SDES session parameter (see section 6.3 or RFC 4568). The default is to
		copy these options from the offering client, or not to have them enabled if SDES wasn't offered.

	- `encrypted_srtp`, `encrypted_srtcp` and `authenticated_srtp`

		Negates the respective option. This is useful if one of the session parameters was offered by
		an SDES endpoint, but it should not be offered on the far side if this endpoint also speaks SDES.

	- `no-`*SUITE*

		Exclude individual crypto suites from being included in the offer. For example,
		`no-NULL_HMAC_SHA1_32` would exclude the crypto suite `NULL_HMAC_SHA1_32` from
		the offer. This has two effects: if a given crypto suite was present in a received
		offer, it will be removed and will be missing in the outgoing offer; and if a given crypto
		suite was not present in the received offer, it will not be added to it.

		*Remark: if after applying the policies to the processed offer, there are no crypto suites left,*
		*which can be used later in the answer towards the offerer, then rtpengine will intentionally*
		*leave the top most one offered, for the answer towards the originator.*
		*However it will be not used for the recipient.*

	- `only-`*SUITE*

		Add only these individual crypto suites and none of the others. For example,
		`only-NULL_HMAC_SHA1_32` would only accept the crypto suite `NULL_HMAC_SHA1_32` for
		the offer being generated. This takes precedence over the `SDES-no-` flag(s),
		if used together, so the `SDES-no` will be not taken into account.
		This has two effects: if a given crypto suite was present in a received offer, it will
		be kept, so will be present in the outgoing offer; and if a given crypto suite was not
		present in the received offer, it will be added to it. The rest, which is not mentioned,
		will be dropped/not added.

		*Remark: if after applying the policies to the processed offer, there are no crypto suites left,*
		*which can be used later in the answer towards the offerer, then rtpengine will intentionally*
		*leave the top most one offered, for the answer towards the originator.*
		*However it will be not used for the recipient.*

	- `nonew`

		Don't add any new crypto suites into the offer. This means, offered SDES crypto suites
		will be accepted, meanwhile no new are going to be generated by rtpengine.
		It takes precedence over the `SDES-no` and `SDES-only` flags, if used in combination.

	- `order:`*SUITES LIST*

		The order, in which crypto suites are being added to the SDP.
		Example: `SDES-order:AES_256_CM_HMAC_SHA;AES_256_CM_HMAC_SHA1_32;AES_192_CM_HMAC_SHA1_80;`,
		this means — those listed SDES crypto suites will be added into the generated SDP body at the top
		of crypto suites list, in the given order. But, each of them is added, only if it is
		about to be added/generated. In other words, the `SDES-order:` flag itself doesn't add crypto suites,
		it just affects the order of those suites to be added.

		And the rest of non-mentioned suites (not mentioned in the `SDES-order:` list),
		which are also to be added, will be appended after those given, in the free manner of ordering.

		Important thing to remember - it doesn't change the crypto suite tag
		for the recipient, even though changing the order of them.

		This flag does not contradict with `SDES-nonew`, `SDES-only-` and `SDES-no-` flags.
		It just orders the list of crypto suites already prepared to be sent out.

	- `offerer_pref:`*SUITES LIST*

		The list of preferred crypto suites to be selected for the offerer.

		It provides a possibility to select specific crypto suite(s) for the offerer from
		the given list of crypto suites received in the offer.
    
		This will be used later on, when processing an answer from
		the recipient and generating an answer to be sent out towards offerer.

		Furthermore, this is being decided not when the answer is processed,
		but already when the offer is processed.

		Flag usage example:
		`SDES-offerer_pref:AES_256_CM_HMAC_SHA;AES_256_CM_HMAC_SHA1_32;`

	- `pad`

		RFC 4568 (section 6.1) is somewhat ambiguous regarding the base64 encoding format of
		`a=crypto` parameters added to an SDP body. The default interpretation is that trailing
		`=` characters used for padding should be omitted. With this flag set, these padding
		characters will be left in place.

	- `lifetime`

		Add the key lifetime parameter `2^31` to each crypto key.

	- `static`

		Instructs *rtpengine* to skip the full SDES negotiation routine during a re-invite
		(e.g. pick the first support crypto suite, look for possible SRTP passthrough)
		and instead leave the previously negotiated crypto suite in place. Only useful in
		subsequent `answer` messages and ignored in `offer` messages.

	- `prefer`

        If an `offer` or `publish` contain both DTLS and SDES options, by
        default *rtpengine* prefers DTLS over SDES and would end up accepting
        DTLS. With this option set, in this scenario SDES would be preferred
        and accepted, while DTLS would be rejected. Useful in combination with
        `DTLS=off`.

* `supports`

	Contains a list of strings. Each string indicates support for an additional feature
	that the controlling SIP proxy supports. Currently defined values are:

	* `load limit`

		Indicates support for an extension to the *ng* protocol to facilitate certain load
		balancing mechanisms. If *rtpengine* is configured with certain session or load
		limit options enabled (such as the `max-sessions` option), then normally *rtpengine*
		would reply with an error to an `offer` if one of the limits is exceeded. If support
		for the `load limit` extension is indicated, then instead of replying with an error,
		*rtpengine* responds with the string `load limit` in the `result` key of the response
		dictionary. The response dictionary may also contain the optional key `message` with
		an explanatory string. No other key is required in the response dictionary.

* `to-interface`

    Contains a string identifying the network interface pertaining to the
    "going to" direction of this message. Identical to setting the second
    `direction=` value.

* `to-label`

	Commands that allow selection of two call participants (e.g. `block
	media`) can use `label` instead of `from-tag` to select the first call
	participant. The `to-label` can then be used instead of `to-tag` to
	select the other call participant.

	For `subscribe request` the `to-label` is synonymous with `set-label`.

* `TOS`

	Contains an integer. If present, changes the TOS value for the entire call, i.e. the TOS value used
	in outgoing RTP packets of all RTP streams in all directions. If a negative value is used, the previously
	used TOS value is left unchanged. If this key is not present or its value is too large (256 or more), then
	the TOS value is reverted to the default (as per `--tos` command line).

* `transport protocol`

	The transport protocol specified in the SDP body is to be rewritten to the string value given here.
	The media
	proxy will expect to receive this protocol on the allocated ports, and will talk this protocol when
	sending packets out. Translation between different transport protocols will happen as necessary.

	Valid values are: `RTP/AVP`, `RTP/AVPF`, `RTP/SAVP`, `RTP/SAVPF`.

	Additionally the string `accept` can be given in `answer` messages to allow a special case: By
	default (when no `transport-protocol` override is given) in answer messages, *rtpengine* will
	use the transport protocol that was originally offered. However, an answering client may answer
	with a different protocol than what was offered (e.g. offer was for `RTP/AVP` and answer comes
	with `RTP/AVPF`). The default behaviour for *rtpengine* is to ignore this protocol change and
	still proceed with the protocol that was originally offered. Using the `accept` option here
	tells *rtpengine* to go along with this protocol change and pass it to the original offerer.

* `trigger`

	A string of DTMF digits that enable a DTMF blocking mode when detected.

* `trigger-end` or `end trigger`

	A string of DTMF digits that disable DTMF blocking or enable a
	different DTMF blocking mode when detected, but only after the initial
	enabling `trigger` has been detected.

* `trigger-end-time`

	Time in milliseconds that a DTMF blocking mode enabled by the `trigger`
	should remain active the most. After the time has expired, the blocking
	mode is switched to the `trigger-end` mode.

* `trigger-end-digits`

	Number of DTMF digits that a DTMF blocking mode enabled by the
	`trigger` should remain active the most. After this number of DTMF
	digits has been detected, the blocking mode is switched to the
	`trigger-end` mode.

* `T.38`

	Contains a list of strings. Each string is a flag that controls the behaviour regarding
	T.38 transcoding. These flags are ignored if the message is not an `offer`.
	Recognised flags are:

	- `decode`

		If the received SDP contains a media section with an `image` type, `UDPTL`
		transport, and `t38` format string, this flag instructs *rtpengine* to convert
		this media section into an `audio` type using RTP as transport protocol.
		Other transport protocols (such as SRTP) can be selected using `transport protocol`
		as described above.

		The default audio codecs to be offered are `PCMU` and `PCMA`. Other audio codecs
		can be specified using the `transcode=` flag described above, in which case the
		default codecs will not be offered automatically.

	- `force`

		If the received SDP contains an audio media section using RTP transport, this flag
		instructs *rtpengine* to convert it to an `image` type media section using the UDPTL
		protocol. The first supported audio codec that was offered will be used to transport
		T.30. Default options for T.38 are used for the generated SDP.

	- `stop`

		Stops a currently active T.38 gateway that was previously engaged using the `decode`
		or `force` flags. This is useful to handle a rejected T.38 offer and revert the
		session back to media passthrough.

	- `no-ECM`

		Disable support for ECM. Support is enabled by default.

	- `no-V.17`

		Disable support for V.17. Support is enabled by default.

	- `no-V.27ter`

		Disable support for V.27ter. Support is enabled by default.

	- `no-V.29`

		Disable support for V.29. Support is enabled by default.

	- `no-V.34`

		Disable support for V.34. Support is enabled by default.

	- `no-IAF`

		Disable support for IAF. Support is enabled by default.

	- `FEC`

		Use UDPTL FEC instead of redundancy. Only useful with `T.38=force` as
		it's a negotiated parameter.

* `T.38 version`

    Sets the T.38 version number to use for the T.38 gateway. The default is
    version zero, or to go along with what has been advertised in the SDP if
    responding to a received T.38 offer. Overriding the version to zero
    regardless of what has been advertised in the SDP can solve T.38 gateway
    problems against certain endpoints.

* `volume`

	Sets the tone volume for `DTMF-security` modes `tone`, `zero, `DTMF`,
	and `random` in negative dB. The default is -10 dB. The highest
	possible volume is 0 dB and the lowest possible volume is -63 dB.

* `xmlrpc-callback`

	Contains a string that encodes an IP address (either IPv4 or IPv6) in printable format.
	If specified, then this address will be used as destination address for the XMLRPC timeout
	callback (see `b2b-url` option).

**Optionally included flags are:**

The value of the `flags` key is a list. The list contains zero or more of the following strings.
Spaces in each string may be replaced by hyphens.

* `all`

	Synonymous to `all=all` (see below).

* `allow asymmetric codecs`

    Normally *rtpengine* expects codecs that were offered during an SDP `offer`
    to match the ones that are accepted in the corresponding SDP `answer`. This
    expectation includes the RTP payload type number. In particular this is
    relevant to codecs using dynamic RTP payload type numbering (generally 96
    and above). For example if the SDP `offer` included AMR-WB with payload
    type number 98, then the answering client is expected to also use payload
    type number 98 if it wanted to accept this codec.

    With this option set, mismatched payload type numbers are accepted and
    honoured. If an answering client accepts a codec that was not offered (with
    that payload type number), then a lookup is performed in attempt to find a
    matching and compatible codec from the offer with a different payload type
    number. If a match is found then the codec is considered as accepted.

    Note that payload type number translation will not be performed in this
    situation.

* `allow no codec media`

    Enables special handling for SDP media sections (`m=` lines) that are left
    without any codecs after codec manipulation operations (in particular codec
    stripping) have been performed. By default without this option set, a media
    section without any codecs would be considered a usage error, and the
    original list of codecs would be restored so that media flow can be
    established. With this option set, a media section without any codecs would
    be considered intentionally so, and would be converted to a rejected or
    removed media section, that is a media section with a zero port, a dummy
    format list, and further attributes.

* `allow transcoding`

	This flag is only useful in commands that provide an explicit answer SDP to *rtpengine*
	(e.g. `subscribe answer`). For these commands, if the answer SDP does not accept all
	codecs that were offered, the default behaviour is to reject the answer. With this flag
	given, the answer will be accepted even if some codecs were rejected, and codecs will be
	transcoded as required.

* `force transcoding`

	This flag will force transcoding between channels. This provides a loss measurement
	between the A-leg and B-leg. If this flag appears in the NG protocol, it will always be 
	transcoded (for example, between codecs G722 and G722). 

* `always transcode`

	Legacy flag, synonymous to `codec-accept=all`.

* `asymmetric`

	Corresponds to the *rtpproxy* `a` flag. Advertises an RTP endpoint which uses asymmetric
	RTP, which disables learning of endpoint addresses (see below).

* `block DTMF`

    Useful in `offer` or `answer` messages to immdiately enable DTMF blocking
    (or other DTMF security mechanism) for the relevant call party, identical
    to using a `block DTMF` message for the call party immediately after.

* `block egress`

    Instructs *rtpengine* to suppress and block other egress media to a remote
    client while media playback towards that client is ongoing. Useful for
    `play media` messages, as well as `offer` and `answer` in combination with
    `recording announcement`.

* `block short` or `block short packets`

    Enables blocking of short RTP packets for the applicable call participant.
    Short RTP packets are packets shorter than the expected minimum length,
    which is determined empirically based on what is observed on the wire.
    Short packets are simply discarded. This is supported only for codecs for
    which a fixed packet size is expected (e.g. G.711).

* `debug` or `debugging`

	Enabled full debug logging for this call, regardless of global log level settings.

* `detect DTMF`

	When present in a message that sets up codec handlers, enables
	the DSP to detect in-band DTMF audio tones even when it
	wouldn't otherwise be necessary.

* `discard recording`

    When file recording is in use, instructs the recording daemon to discard
    (delete) the recording files, as well as the database entries if present.

* `exclude recording`

    Instructs *rtpengine* to exclude this call participant's media from being
    recorded. When used within an offer/answer exchange, applies to both call
    parties involved.

* `skip-recording-db`

    Suppress writing the information about the call recording to the configured
    metadata DB.

* `early media`

    Used in conjunction with the audio player. If set, audio playback is
    started immediately when processing an `offer` message. The default
    behaviour is to start the audio player only after the `answer` has been
    processed, or when any audio to be played back has actually been received
    (either from another party to the call, or via the `play media` command).

* `full rtcp attribute`

	Include the full version of the `a=rtcp` line (complete with network address) instead of
	the short version with just the port number.

* `generate RTCP`

	Identical to setting `generate RTCP = on`.

* `generate mid`

	Add `a=mid` attributes to the outgoing SDP if they were not already present.

* `inactive`

    Useful for `subscribe request` messages to produce an SDP which is marked
    as inactive, instead of `sendonly` which is the default. This can be used
    to pause media sent to a subscription.

* `inject DTMF`

	Signals to *rtpengine* that the audio streams involved in this `offer` or `answer`
	(the flag should be present in both of them) are to be made available for DTMF
	injection via the `play DTMF` control message. See `play DTMF` below for additional
	information.

* `loop protect`

	Inserts a custom attribute (`a=rtpengine:...`) into the outgoing SDP to prevent *rtpengine*
	processing and rewriting the same SDP multiple times. This is useful if your setup
	involves signalling loops and need to make sure that *rtpengine* doesn't start looping
	media packets back to itself. When this flag is present and *rtpengine* sees a matching
	attribute already present in the SDP, it will leave the SDP untouched and not process
	the message.

* `media handover`

	Similar to the `strict source` option, but instead of dropping packets when the source address
	or port don't match, the endpoint address will be re-learned and moved to the new address. This
	allows endpoint addresses to change on the fly without going through signalling again. Note that
	this opens a security hole and potentially allows RTP streams to be hijacked, either partly or
	in whole.

* `NAT-wait`

	Prevents forwarding media packets to the respective endpoint
	until at least one media packet has been received from that
	endpoint. This is to allow a NAT binding to open in the ingress
	direction before sending packets out, which could result in an
	automated firewall block.

* `new branch`

    If *rtpengine* receives an answer from a to-tag that hasn't previously seen
    and no corresponding call party is known (created from a previous offer),
    previously it would treat this is a separate new call branch, create a
    brand new internal call party, and dissociate the previous one. This may
    lead to unexpected results as this new call party has been created without
    the same initialisation as was done for the original one, and so may be
    left with incorrect or incomplete data (e.g.  SRTP keys, codec information,
    interface bindings, etc).

    Improve this by treating an unexpected and unseen to-tag as an alias to the
    already existing to-tag. Going forward both tags can then be used
    interchangeably to refer to the same monologue.

    This flag suppresses this new behaviour, in case some situation is made
    worse by it.

* `no port latching`

	Port latching is enabled by default for endpoints which speak
	ICE. With this option preset, a remote port change will result
	in a local port change even for endpoints which speak ICE,
	which will imply an ICE restart.

* `no rtcp attribute`

	Omit the `a=rtcp` line from the outgoing SDP.

* `original sendrecv`

	With this flag present, *rtpengine* will leave the media direction attributes
	(`sendrecv`, `recvonly`, `sendonly`, and `inactive`) from the received SDP body
	unchanged. Normally *rtpengine* would consume these attributes and insert its
	own version of them based on other media parameters (e.g. a media section with
	a zero IP address would come out as `sendonly` or `inactive`).

* `pad crypto`

	Legacy alias to SDES=pad.

* `pierce NAT`

	Sends empty UDP packets to the remote RTP peer as soon as an
	endpoint address is available from a received SDP, for as long
	as no incoming packets have been received. Useful to create an
	initial NAT mapping. Not needed when ICE is in use.

* `port latching`

	Forces *rtpengine* to retain its local ports during a signalling exchange even when the
	remote endpoint changes its port.

* `provisional`

    Disables special behaviour when operating on a message that was triggered
    by a SIP response with a non-provisional (>= 200) status code.
    Specifically, setting this flag allows for changed or updated to-tag even
    after a final SIP response has been received.

* `record call`

	Identical to setting `record call` to `on` (see below).

* `recording announcement`

    Enable playback of an announcement message when call recording is started.
    One of the flags identifying a media file (such as `file=`, same as for the
    `play media` message) must also be given, and generally usage of `block
    egress` is recommended.

    Announcement messages are enabled directionally, meaning this flag enables
    it for the call party relevant to the current message (e.g the call
    originator for an initial `invite`) but not for other. In other words this
    flag must be set for all call parties which are meant to hear the
    announcement.

* `recrypt`

    If set, forces decryption and re-encryption of all SRTP, even if
    passthrough is possible.

    Without this flag set, in situations where SRTP is in use on both sides of
    a call, and if the SRTP keys are the same on both sides (commonly the case
    with SDES), and if no other media manipulations are required, *rtpengine*
    would simply pass through all media packets directly, without getting
    involved in the encryption. This saves CPU time as there's no point in
    decypting each packet, only to encrypt it again using the same SRTP key.
    The side effect is that even non-SRTP packets, as well as SRTP packets with
    an invalid encryption, are passed through, as no verification of the SRTP
    encryption takes place. This can have security implications, as then any
    and all media packets are considered for endpoint learning purposes,
    regardless of whether they have an intact SRTP authentication tag.

    Setting this flag forces decryption and re-encryption of all SRTP packets,
    validating the authentication tag in the process, and discarding packets
    without a valid tag.

* `reject ICE`

	Useful for `offer` messages that advertise support for ICE.
	Instructs *rtpengine* to reject the offered ICE. This is
	similar to using `ICE=remove` in the respective `answer`.

* `reset`

	This causes *rtpengine* to un-learn certain aspects of the RTP endpoints involved, such as
	support for ICE or support for SRTP. For example, if `ICE=force` is given, then *rtpengine*
	will initially offer ICE to the remote endpoint. However, if a subsequent answer from that
	same endpoint indicates that it doesn't support ICE, then no more ICE offers will be made
	towards that endpoint, even if `ICE=force` is still specified. With the `reset` flag given,
	this aspect will be un-learned and *rtpengine* will again offer ICE to this endpoint.
	This flag is valid only in an `offer` message and is useful when the call has been
	transferred to a new endpoint without change of `From` or `To` tags.

* `reuse codecs` or `no codec renegotiation`

	Instructs *rtpengine* to prevent endpoints from switching codecs during call run-time
	if possible. Codecs that were listed as preferred in the past will be kept as preferred
	even if the re-offer lists other codecs as preferred, or in a different order. Recommended
	to be combined with `single codec`.

* `RTCP mirror`

	Useful only for `subscribe request` message. Instructs
	*rtpengine* to not only create a one-way subscription for both
	RTP and RTCP from the source to the sink, but also create a
	reverse subscription for RTCP only from the sink back to the
	source.  This makes it possible for the media source to receive
	feedback from all media receivers (sinks).

* `single codec`

	Using this flag in an `answer` message will leave only the first listed codec in place
	and will remove all others from the list. Useful for RTP clients which get confused if
	more than one codec is listed in an answer.

* `static codecs`

    Useful in an `offer` message to suppress any change in codecs towards the
    answer side, instead of passing along the list of offered codecs from the
    offer side as it normally would.

* `SIP source address`

	Ignore any IP addresses given in the SDP body and use the source address of the received
	SIP message (given in `received from`) as default endpoint address. This was the default
	behaviour of older versions of *rtpengine* and can still be made the default behaviour
	through the `--sip-source` CLI switch.
	Can be overridden through the `media address` key.

* `symmetric`

	Corresponds to the *rtpproxy* `w` flag. Not used by *rtpengine* as this is the default,
	unless `asymmetric` is specified.

* `trust address`

	The opposite of `SIP source address`. This is the default behaviour unless the CLI switch
	`--sip-source` is active. Corresponds to the *rtpproxy* `r` flag.
	Can be overridden through the `media address` key.

* `strip extmap`

	Remove `a=rtpmap` attributes from the outgoing SDP.

* `strict source`

	Normally, *rtpengine* attempts to learn the correct endpoint address for every stream during
	the first few seconds after signalling by observing the source address and port of incoming
	packets (unless `asymmetric` is specified). Afterwards, source address and port of incoming
	packets are normally ignored and packets are forwarded regardless of where they're coming from.
	With the `strict source` option set, *rtpengine* will continue to inspect the source address
	and port of incoming packets after the learning phase and compare them with the endpoint
	address that has been learned before. If there's a mismatch, the packet will be dropped and
	not forwarded.

* `trickle ICE`

	Useful for `offer` messages when ICE is advertised to also advertise
	support for trickle ICE.

* `unidirectional`

	When this flag is present, kernelize also one-way rtp media.

* `WebRTC`

    Shortcut alias for several other flags that must be set when talking to a
    WebRTC client. Currently an alias for (subject to change):
    `transport-protocol=UDP/TLS/RTP/SAVPF` `ICE=force` `tricke-ICE`
    `rtcp-mux-offer` `rtcp-mux-require` `no-rtcp-attribute` `SDES-off`
    `generate-mid`

**Optionally included replace-flags are:**

Similar to the usual `flags` list, but this one controls which parts of the SDP body should be rewritten.
Contains zero or more of:

* `force-increment-sdp-ver`

	Force increasing the SDP version, even if the SDP hasn't been changed.

* `origin`

	Replace the address found in the *origin* (o=) line of the SDP body.

* `origin-full`

	Replace whole *origin* (o=) line of the SDP body, so that all origin fields
	in the `o=` line always remain the same in all SDPs going to a
	particular RTP endpoint. A behavior in relation to the address field is the same
	as by the `origin` option flag.

* `session name` or `session-name`

	Same as `username` but for the entire contents of the `s=` line.

* `SDP version` or `SDP-version`

	Take control of the version field in the SDP and make sure it's increased every
	time the SDP changes, and left unchanged if the SDP is the same.

* `username`

	Take control of the origin username field in the SDP. With this
	option in use, *rtpengine* will make sure the username field in
	the `o=` line always remains the same in all SDPs going to a
	particular RTP endpoint.

* `zero address`

	Using a zero endpoint address is an obsolete way to signal a
	muted or sendonly stream. Streams with zero addresses are
	normally flagged as sendonly and the zero address in the SDP is
	passed through. With this option set, the zero address is
	replaced with a real address.

**Optionally included codec manipulations:**

`codec` contains a dictionary controlling various aspects of codecs (or RTP payload types).

These options can also be put into the `flags` list using a prefix of `codec-`. For example,
to set the codec options for two variants of Opus when they're implicitly accepted, (see
the example under `set`), one would put the following into the `flags` list:
`codec-set-opus/48000/1/16000` `codec-set-opus/48000/2/32000`

The following keys are understood:

* `accept`

	Similar to `mask` and `consume` but doesn't remove the codec from the list of
	offered codecs. This means that a codec listed under `accept` will still be offered
	to the remote peer, but if the remote peer rejects it, it will still be accepted
	towards the original offerer and then used for transcoding. It is a more selective
	version of what the `always transcode` flag does.

	The special string `any` can be used for the `publish` message.
	See below for more details.

* `consume`

	Identical to `mask` but enables the transcoding engine even if no other transcoding
	related options are given.

* `except`

	Contains a list of strings. Each string is the name of a codec that should be
	included in the list of codecs offered. This is primarily useful to block all
	codecs (`strip -> all` or `mask -> all`) except the ones given in the `except`
	whitelist.  Codecs that were not present in the original list of codecs
	offered by the client will be ignored.

	This list also supports codec format parameters as per above.

* `ignore`

	Similar to the `strip` option below, but affects only codecs listed in the
	incoming received SDP. Codecs listed here are treated as if they were never
	offered, and so will not be used for media towards the offerer. Note that
	codecs listed here would still be used in the outgoing rewritten offer SDP,
	unless the same codecs are also listed under `strip`. This means that if a
	codec is only ignored but not stripped, and if that codec is then accepted
	by the answerer, transcoding will necessarily be enabled.

* `mask`

	Similar to `strip` except that codecs listed here will still be accepted and
	used for transcoding on the offering side. Useful only in combination with
	`transcode`. For example, if an offer advertises Opus and the options
	`mask=opus, transcode=G723` are given, then the rewritten outgoing offer
	will contain only G.723 as offered codec, and transcoding will happen
	between Opus and G.723. In contrast, if only `transcode=G723` were given, then
	the rewritten outgoing offer would contain both Opus and G.723. On the other
	hand, if `strip=opus, transcode=G723` were given, then Opus would be unavailable
	for transcoding.

	As with the `strip` option, the special keywords `all`  and `full` can be used
	to mask all codecs that have been offered.

	This option is only processed in `offer` messages and ignored otherwise.

* `offer`

	This is identical to `except` but additionally allows the codec order to be
	changed. So the first codec listed in `offer` will be the primary (preferred)
	codec in the output SDP, even if it wasn't originally so.

* `set`

	Contains a list of strings. This list makes it possible to set codec options
	(bitrate in particular) for codecs that are implicitly accepted for transcoding.
	For example, if `AMR` was offered, `transcode=PCMU` was given, and the remote
	ended up accepting `PCMU`, then this option can be used to set the bitrate used
	for the AMR transcoding process.

	Each string must be a full codec specification as per above, including clock rate
	and number of channels. Using the example above, `set=AMR/8000/1/7400` can be used
	to transcode to AMR with 7.4 kbit/s.

	Codec options (bitrate) are only applied to codecs that match the given parameters
	(clock rate, channels), and multiple options can be given for the same coded with
	different parameters. For example, to specify different bitrates for Opus for both
	mono and stereo output, one could use `set=[opus/48000/1/16000,opus/48000/2/32000]`.

	This option is only processed in `offer` messages and ignored otherwise.

* `strip`

	Contains a list of strings. Each string is the name of a codec or RTP payload
	type that should be removed from the SDP. Codec names are case sensitive, and
	can be either from the list of codecs explicitly defined by the SDP through
	an `a=rtpmap` attribute, or can be from the list of RFC-defined codecs. Examples
	are `PCMU`, `opus`, or `telephone-event`. Codecs stripped using this option
	are only removed from the outgoing rewritten SDP and don't affect the list
	of codecs that was offered by the source SDP. See the `ignore` option above
	for a similar mechanism that affects the offer codecs.

	It is possible to specify codec format parameters alongside with the codec name
	in the same format as they're written in SDP for codecs that support them,
	for example `opus/48000` to
	specify Opus with 48 kHz sampling rate and one channel (mono), or
	`opus/48000/2` for stereo Opus. If any format parameters are specified,
	the codec will only be stripped if all of the format parameters match, and other
	instances of the same codec with different format parameters will be left
	untouched.

	As a special keyword, `all` can be used to remove all codecs, except the ones
	that should explicitly offered (see below). Note that it is an error to strip
	all codecs and leave none that could be offered. In this case, the original
	list of codecs will be left unchanged.

	The keyword `full` can also be used, which behaves the same as `all` with the
	exception listed under `transcode` below.

* `transcode`

	Similar to `offer` but allows codecs to be added to the list of offered codecs
	even if they were not present in the original list of codecs. In this case,
	the transcoding engine will be engaged. Only codecs that are supported for both
	decoding and encoding can be added in this manner. This also has the side effect
	of automatically stripping all unsupported codecs from the list of offered codecs,
	as *rtpengine* must expect to receive or even send in any codec that is present
	in the list.

	Note that using this option does not necessarily always engage the transcoding
	engine. If all codecs given in the `transcode` list were present in the original
	list of offered codecs, then no transcoding will be done. Also note that if
	transcoding takes place, in-kernel forwarding is disabled for this media stream
	and all processing happens in userspace.

	If no codec format parameters are specified in this list (e.g. just `opus`
	instead of `opus/48000/2`), default values will be chosen for them.

	For codecs that support different bitrates, it can be specified by appending
	another slash followed by the bitrate in bits per second,
	e.g. `opus/48000/2/32000`. In this case, all format parameters (clock rate,
	channels) must also be specified.

	Additional options that can be appended to the codec string with additional slashes
	are ptime, the `fmtp` string, and additional codec-specific options. For example
	`iLBC/8000/1///mode=30` to use `mode=30` as `fmtp` string.

	For Opus, the string of codec-specific options is passed
	directly to ffmpeg, so all ffmpeg codec options can be set. Use
	space, colon, semicolon, or comma to separate individual
	options. For example to set the encoding complexity (also known
	as compression level by ffmpeg):
	`opus/48000/2////compression_level=2`

	If a literal `=` cannot be used due to parsing constraints (i.e. being wrongly
	interpreted as a key-value pair), it can be escaped by using two dashes instead,
	e.g. `iLBC/8000/1///mode--30`.

	As a special case, if the `strip=all` or `mask=all` option has been used and
	the `transcode` option is used on a codec that was originally present in the offer,
	then *rtpengine* will treat this codec the same as if it had been used with the
	`offer` option, i.e. it will simply restore it from the list of stripped codecs and
	won't actually engage transcoding for this codec. On the other hand, if a codec has
	been stripped explicitly by name using the `strip` or `mask` option and then used again
	with the `transcode` option, then the codec will not simply be restored from the
	list of stripped codecs, but instead a new transcoded instance of the codec will
	be inserted into the offer. (This special exception does not apply to `mask=full`
	or `strip=full`.)

	This option is only processed in `offer` messages and ignored otherwise.

**Optionally included SDP attributes manipulations:**

`sdp-attr` contains a dictionary controlling various aspects of attribute lines (or `a=` lines).

An intention of these option flags is to control session (global) and media level
attributes, with help of which it's possible to do the following manipulations:
- addition
- removal
- substitution (replacement)

This does affect an outgoing SDP offer. So it's meant to manipulate body attributes,
which rtpengine generates during the offer processing. In other words, it manipulates
what has been already prepared by rtpengine on its own, taking into account received offer.

Furthermore, it's quite important to remember, that the changes, which have been
applied to SDP body attributes, will be not taken into account by rtpengine itself,
so these changes are rather formal (textual).
This means, it's not the same, as if they would be originally given by the session originator.

That's why this kind of flags must be used with a full carefulness, because if not,
this can potentially lead to the unexpected result.

Usage syntax:

		"sdp-attr" :
		{
			"<media-type>":
			{
				"<command>": ["<value>", "<value>"],
				"<command>": ["<value>", "<value>"]
			},
			"<media-type>":
			{
				"<command>": ["<value>", "<value>"],
				"<command>": ["<value>", "<value>"]
			}
		}

Description:

* `<media-type>`

	Defines a level of command application. One media type can be given only once per command.
	`<media-type>` can have one of the following values:

	- `none` or `global`

		Applies to the session level (global) attributes, but not to any of the
		media session specific attributes.

	- `audio`

		Applies to all currently present media sessions of audio type.

	- `video`

		Applies to all currently present media sessions of audio type.

* `<command>`

	The command to be applied to the targeted attribute line(s).
	Each command can be used multiple times within one media session/global scope.

	- `add`

		Adds a new `a=` line with a given value to the concerned attributes list.
		If the attribute with such value already exists within this scope of media session,
		then no duplication is to be added, therefore the older one remains untouched
		and nothing extra is being added.

		Can take multiple values (so multiple attributes can added per one command).

	- `remove`

		Removes a specified `a=` line from the concerned attributes list.
		If such line hasn't been found, then the attributes list remains untouched.

        The matching can be done using just the attribute name, or the
        attribute name plus a tag value, or the full attribute line (value and
        the following attribute parameters, if given). For example, the
        attribute `a=foo:bar baz quuz` would match any of `foo`, foo:bar`, or
        `foo:bar baz quuz`.

		Can take multiple values (so multiple attributes can removed per one command).

	- `substitute`

		Substitutes a specified `a=` line taken from the concerned media attributes list.
		If such line hasn't been found, then the attributes list remains untouched.

        The matching can be done using just the attribute name, or the
        attribute name plus a tag value, or the full attribute line (value and
        the following attribute parameters, if given). If the attribute is
        generated by *rtpengine* itself and a tag value is present, then the
        tag value must also be used in the match pattern. For example, the
        attribute `a=foo:bar baz quuz` would match any of `foo`, foo:bar`, or
        `foo:bar baz quuz`, but the self-generated attribute `a=fmtp:10 foobar`
        could only be substituted using either `fmtp:10` or `fmtp:10 foobar`
        but not just `fmtp`.

		Substitutes one attribute at a time, so one attribute into another attribute.
		Read more about that below in the `<value>` section.

* `<value>`

	The `value` has to not include the `a=` (lvalue) part.
	It contains only the value, that is given after the equal sign.

	No wild-cards or regular expressions are accepted.

	It's important to remember that some attributes are allowed to be present multiple times.
	Furthermore rtpengine does not expect given `a=` lines (to be substituted)
	to be unique within concerned media scope (global, audio or video).

	This leads to the next point — `remove` and `substitute` commands can affect just
	a single attribute, as well as multiple attributes, depending on the uniqueness
	of the value in the given command.

	User is supposed to provide full `a=` line value, so that it gives expected behavior.

	Important remark regarding `substitute` command.
	It takes only two values at a time, in other words it substitutes one attribute per command:
	- the first `value`, that matches the full value to be substituted; and
	- the second `value`, that is to be placed instead.
	Therefore, the only allowed syntax for it is (per command):

		"substitute": ["from-this-attribute", "to-that-attribute"]

	All other possible usages will be ignored and only first two values will be taken.
	However, multiple `substitute` commands can be given per time, see examples below.

Examples:

* Add a new (single) attribute line to the session (global) level:

		"sdp-attr" :
		{
			"none" :
			{
				"add" : [ "sendrecv" ]
			}
		}

* Add two new attribute lines to audio session and remove one for video session:

		"sdp-attr" :
		{
			"audio" :
			{
				"add" : [ "ptime:20", "sendrecv" ]
			},
			"video":
			{
				"remove" : [ "rtpmap:101 telephone-event/8000" ]
			}
		}

* Substitute two attributes of the global session and one for audio media section (pay attention, `substitute` uses lists, not dictionaries):

		"sdp-attr" :
		{
			"none" :
			{
				"substitute": [[ "sendrecv" , "sendonly" ], [ "ptime:20" , "ptime:40" ]]
			},
			"audio" :
			{
				"substitute": [["fmtp:101 0-15" , "fmtp:126 0-16" ]]
			},
		}

* As an alternative syntax these can be listed in the `flags` list. An example of such syntax:
- `sdp-attr-remove-audio-ptime:20`
- `sdp-attr-substitude-none-sendrecv>sendonly`.

In such usage equals sign (`=`) can be escaped as double dashes (`--`) and spaces can be escaped as double periods (`..`).

An example of a complete `offer` request dictionary could be (SDP body abbreviated):

	{ "command": "offer", "call-id": "cfBXzDSZqhYNcXM", "from-tag": "mS9rSAn0Cr",
	"sdp": "v=0\r\no=...", "via-branch": "5KiTRPZHH1nL6",
	"flags": [ "trust address" ], "replace": [ "origin" ],
	"address family": "IP6", "received-from": [ "IP4", "10.65.31.43" ],
	"ICE": "force", "transport protocol": "RTP/SAVPF", "media address": "2001:d8::6f24:65b",
	"DTLS": "passive" }

A response message contains only the key `sdp` in addition to `result`, which contains the re-written
SDP body that the SIP proxy should insert into the SIP message.

Example response:

	{ "result": "ok", "sdp": "v=0\r\no=..." }

**Optionally included SDP media manipulations:**

`sdp-media-remove` contains a list pointing, which media types are to be removed from SDP.

This does affect an outgoing SDP offer. So it's meant to manipulate an SDP body,
which rtpengine generates during the offer processing. The removed media type will be then
not taken into consideration during further processing.

When this flag is added, rtpengine will not show concerned media type(s), hence media section(s)
to the recipient's side. Therefore, later on the recipient side will provide only an answer
for those media section(s) shown to it.

Upon processing such an answer coming back to the changed SDP offer,
rtpengine will just add a zeroed media towards the originator's side in order to fulfill RFC
requirements telling to use a zeroed media for those unaccepted media sections.

Usage syntax:

		"sdp-media-remove" : ["<media-type>", "<media-type>", ...]

Examples:

* Remove all occurences of the video media type:

		"sdp-media-remove" : ["video"]

IANA-registered media types are understood, or the special type `other` can be
given to remove all media sections with types that are not understood.

## `answer` Message

The `answer` message is identical to the `offer` message, with the additional requirement that the
dictionary must contain the key `to-tag` containing the SIP `To` tag. It doesn't make sense to include
the `direction` key in the `answer` message.

The reply message is identical as in the `offer` reply.

## `delete` Message

The `delete` message must contain at least the keys `call-id` and `from-tag` and may optionally include
`to-tag` and `via-branch`, as defined above. It may also optionally include a key `flags` containing a list
of zero or more strings. The following flags are defined:

* `fatal`

	Specifies that any non-syntactical error encountered when deleting the stream
	(such as unknown call-ID) shall
	result in an error reply (i.e. `"result": "error"`). The default is to reply with a warning only
	(i.e. `"result": "ok", "warning": ...`).

* `to-tag`

	This flag controls whether the `"To"` tag's value is honoured or ignored when handling
	delete messages. Normally, the `"To"` tag's value is always included when present,
	but can be disregarded for the `"delete"` type of messages.
	So that, including the `"To-tag"` option flag in the `"delete"` message,
	forces to honour it and hence allows to be more selective about monologues within a dialog to be torn down.

Other optional keys are:

* `delete delay`

	Contains an integer and overrides the global command-line option `delete-delay`. Call/branch will be
	deleted immediately if a zero is given. Value must be positive (in seconds) otherwise.

The reply message may contain additional keys with statistics about the deleted call. Those additional keys
are the same as used in the `query` reply.

## `list` Message

The `list` command retrieves the list of currently active call-ids. This list is limited to 32 elements by
default.

* `limit`

	Optional integer value that specifies the maximum number of results (default: 32). Must be > 0. Be
	careful when setting big values, as the response may not fit in a UDP packet, and therefore be invalid.

## `query` Message

The minimum requirement is the presence of the `call-id` key. Keys `from-tag` and/or `to-tag` may optionally
be specified.

The response dictionary contains the following keys:

* `created`

	Contains an integer corresponding to the creation time of this call within the media proxy,
	expressed as seconds since the UNIX epoch.

* `last signal`

	The last time a signalling event (offer, answer, etc) occurred. Also expressed as an integer
	UNIX timestamp.

* `tags`

	Contains a dictionary. The keys of the dictionary are all the SIP tags (From-tag, To-Tag) known
	by *rtpengine* related to this call. One of the keys may be an empty string, which corresponds to
	one side of a dialogue which hasn't signalled its SIP tag yet. Each value of the dictionary is
	another dictionary with the following keys:

	- `created`

		UNIX timestamp of when this SIP tag was first seen by *rtpengine*.

	- `tag`

		Identical to the corresponding key of the `tags` dictionary. Provided to allow for easy
		traversing of the dictionary values without paying attention to the keys.

	- `label`

		The label assigned to this endpoint in the `offer` or `answer` message.

	- `in dialogue with`

		Contains the SIP tag of the other side of this dialogue. May be missing in case of a
		half-established dialogue, in which case the other side is represented by the null-string
		entry of the `tags` dictionary.

	- `medias`

		Contains a list of dictionaries, one for each SDP media stream known to *rtpengine*. The
		dictionaries contain the following keys:

		+ `index`

			Integer, sequentially numbered index of the media, starting with one.

		+ `type`

			Media type as string, usually `audio` or `video`.

		+ `protocol`

			If the protocol is recognized by *rtpengine*, this string contains it.
			Usually `RTP/AVP` or `RTP/SAVPF`.

		+ `flags`

			A list of strings containing various status flags. Contains zero of more
			of: `initialized`, `rtcp-mux`, `DTLS-SRTP`, `SDES`, `passthrough`, `ICE`.

		+ `streams`

			Contains a list of dictionary representing the packet streams associated
			with this SDP media. Usually contains two entries, one for RTP and one for RTCP.
			The keys found in these dictionaries are listed below:

		+ `local port`

			Integer representing the local UDP port. May be missing in case of an inactive stream.

		+ `endpoint`

			Contains a dictionary with the keys `family`, `address` and `port`. Represents the
			endpoint address used for packet forwarding. The `family` may be one of `IPv4` or
			`IPv6`.

		+ `advertised endpoint`

			As above, but representing the endpoint address advertised in the SDP body.

		+ `crypto suite`

			Contains a string such as `AES_CM_128_HMAC_SHA1_80` representing the encryption
			in effect. Missing if no encryption is active.

		+ `last packet`

			UNIX timestamp of when the last UDP packet was received on this port.

		+ `flags`

			A list of strings with various internal flags. Contains zero or more of:
			`RTP`, `RTCP`, `fallback RTCP`, `filled`, `confirmed`, `kernelized,`
			`no kernel support`.

		+ `stats`

			Contains a dictionary with the keys `bytes`, `packets` and `errors`.
			Statistics counters for this packet stream.

* `totals`

	Contains a dictionary with two keys, `RTP` and `RTCP`, each one containing another dictionary
	identical to the `stats` dictionary described above.

A complete response message might look like this (formatted for readability):

          {
            "totals": {
              "RTCP": {
                    "bytes": 2244,
                    "errors": 0,
                    "packets": 22
                  },
              "RTP": {
                   "bytes": 100287,
                   "errors": 0,
                   "packets": 705
                 }
                  },
            "last_signal": 1402064116,
            "tags": {
                  "cs6kn1rloc": {
                  "created": 1402064111,
                  "medias": [
                          {
                      "flags": [
                             "initialized"
                           ],
                      "streams": [
                               {
                           "endpoint": {
                               "port": 57370,
                               "address": "10.xx.xx.xx",
                               "family": "IPv4"
                                   },
                           "flags": [
                                  "RTP",
                                  "filled",
                                  "confirmed",
                                  "kernelized"
                                ],
                           "local port": 30018,
                           "last packet": 1402064124,
                           "stats": {
                                  "packets": 343,
                                  "errors": 0,
                                  "bytes": 56950
                                },
                           "advertised endpoint": {
                                    "family": "IPv4",
                                    "port": 57370,
                                    "address": "10.xx.xx.xx"
                                  }
                               },
                               {
                           "stats": {
                                  "bytes": 164,
                                  "errors": 0,
                                  "packets": 2
                                },
                           "advertised endpoint": {
                                    "family": "IPv4",
                                    "port": 57371,
                                    "address": "10.xx.xx.xx"
                                  },
                           "endpoint": {
                               "address": "10.xx.xx.xx",
                               "port": 57371,
                               "family": "IPv4"
                                   },
                           "last packet": 1402064123,
                           "local port": 30019,
                           "flags": [
                                  "RTCP",
                                  "filled",
                                  "confirmed",
                                  "kernelized",
                                  "no kernel support"
                                ]
                               }
                             ],
                      "protocol": "RTP/AVP",
                      "index": 1,
                      "type": "audio"
                          }
                        ],
                  "in dialogue with": "0f0d2e18",
                  "tag": "cs6kn1rloc"
                      },
                  "0f0d2e18": {
                      "in dialogue with": "cs6kn1rloc",
                      "tag": "0f0d2e18",
                      "medias": [
                        {
                          "protocol": "RTP/SAVPF",
                          "index": 1,
                          "type": "audio",
                          "streams": [
                             {
                               "endpoint": {
                                   "family": "IPv4",
                                   "address": "10.xx.xx.xx",
                                   "port": 58493
                                 },
                               "crypto suite": "AES_CM_128_HMAC_SHA1_80",
                               "local port": 30016,
                               "last packet": 1402064124,
                               "flags": [
                                "RTP",
                                "filled",
                                "confirmed",
                                "kernelized"
                              ],
                               "stats": {
                                "bytes": 43337,
                                "errors": 0,
                                "packets": 362
                              },
                               "advertised endpoint": {
                                  "address": "10.xx.xx.xx",
                                  "port": 58493,
                                  "family": "IPv4"
                                }
                             },
                             {
                               "local port": 30017,
                               "last packet": 1402064124,
                               "flags": [
                                "RTCP",
                                "filled",
                                "confirmed",
                                "kernelized",
                                "no kernel support"
                              ],
                               "endpoint": {
                                   "family": "IPv4",
                                   "port": 60193,
                                   "address": "10.xx.xx.xx"
                                 },
                               "crypto suite": "AES_CM_128_HMAC_SHA1_80",
                               "advertised endpoint": {
                                  "family": "IPv4",
                                  "port": 60193,
                                  "address": "10.xx.xx.xx"
                                },
                               "stats": {
                                "packets": 20,
                                "bytes": 2080,
                                "errors": 0
                              }
                             }
                           ],
                          "flags": [
                           "initialized",
                           "DTLS-SRTP",
                           "ICE"
                         ]
                        }
                      ],
                      "created": 1402064111
                    }
                },
            "created": 1402064111,
            "result": "ok"
          }

## `start recording` Message

The `start recording` message must contain at least the key `call-id` and may optionally include `from-tag`,
`to-tag` and `via-branch`, as defined above. The reply dictionary contains no additional keys.

Enables call recording for the call, either for the entire call or for only the specified call leg. Currently
*rtpengine* always enables recording for the entire call and does not support recording only individual
call legs, therefore all keys other than `call-id` are currently ignored.

If the chosen recording method doesn't support in-kernel packet forwarding, enabling call recording
via this messages will force packet forwarding to happen in userspace only.

If the optional `recording-file` key is set, then its value will be used as an
output file. Note that the value must refer to a complete (absolute) path
including file name, and a file name extension will not be added.

If the optional `recording-dir` key is set, then its value will be used as the
directory path for the output file(s), overriding the `output-dir` config
option of the recording daemon. The value should refer to an existing directory
given as an absolute path. Setting this key does not affect the names of the
files that will be created in the directory.

If the optional `recording-pattern` key is set, then its value will be used as
the pattern to generate the output file name(s), overriding the
`output-pattern` config option of the recording daemon. Note that no validity
checking is performed on the given string, so make sure that the given pattern
does not yield duplicate file names.

The option `recording-file` takes precedence over both `recording-dir` and
`recording-pattern` if multiple options are set.

## `stop recording` Message

The `stop recording` message must contain the key `call-id` as defined above.
The reply dictionary contains no additional keys. See below under `pause
recording` for another alternative usage for this message.

Disables call recording for the call. This can be sent during a call to immediately stop recording it.

## `pause recording` Message

Identical to `stop recording` except that it instructs the recording daemon not
to close the recording file, but instead leave it open so that recording can
later be resumed via another `start recording` message.

Alternatively the `stop recording` message can be used if either the string
`pause` is given in the `flags` list, or if the dictionary contains the key
`pause=yes`.

## `block DTMF` and `unblock DTMF` Messages

These message types must include the key `call-id` in the message. They enable and disable blocking of DTMF
events (RFC 4733 type packets), respectively.

Packets can be blocked for an entire call if only the `call-id` key is present in the message, or can be blocked
directionally for individual participants. Participants can be selected by their SIP tag if the `from-tag` key
is included in the message, they can be selected by their SDP media address if the `address` key is included
in the message, or they can be selected by the user-provided `label` if the `label` key is included in the
message. For an address, it can be an IPv4 or IPv6 address, and any participant that is
found to have a matching address advertised as their SDP media address will have their originating RTP
packets blocked (or unblocked).

Unblocking packets for the entire call (i.e. only `call-id` is given) does not
automatically unblock packets for participants which had their packets blocked
directionally, unless the string `all` (equivalent to setting `all=all`) is
included in the `flags` section of the message.

When DTMF blocking is enabled, DTMF event packets will not be forwarded to the receiving peer.
If DTMF logging is enabled, DTMF events will still be logged to syslog while blocking is enabled. Blocking
of DTMF events can be enabled and disabled at any time during call runtime.

## `block media` and `unblock media` Messages

Analogous to `block DTMF` and `unblock DTMF` but blocks media packets instead of DTMF packets. DTMF packets
can still pass through when media blocking is enabled. Media packets can be blocked for an entire call, or
directionally for individual participants. See `block DTMF` above for details.

In addition to blocking media for just one call participant, it's possible to
block media for just a single media flow. This is relevant to scenarios that
involve forked media that were established with one or more `subscribe
request`. To select just one media flow for media blocking, in addition to
selecting a source call participant as above, a destination call participant
must be specified using the `to-tag` or `to-label`key in the message.

Another possibility to block media for individual media flows is to use one of
the special `all=` keywords instead of directly specifying a single `to-tag` or
`to-label`. With `all=offer-answer` all media flows from the given `from-tag`
that resulted from an offer/answer negotiation are affected. Respectively with
`all=except-offer-answer` the opposite happens. With `all=flows` all currently
established media flows are affected regardless or how they were created.

## `silence media` and `unsilence media` Messages

Identical to `block media` and `unblock media` except that media packets are
not simply blocked, but rather have their payload replaced with silence audio.
This is only supported for certain trivial audio codecs (i.e. G.711, G.722).

## `start forwarding` and `stop forwarding` Messages

These messages control the recording daemon's mechanism to forward PCM via TCP/TLS. Unlike the call recording
mechanism, forwarding can be enabled for individual participants (directionally) only, therefore these
messages can be used with the same options as the `block` and `unblock` messages above. The PCM forwarding
mechanism is independent of the call recording mechanism, and so forwarding and recording can be started
and stopped independently of each other.

## `play media` Message

Only available if compiled with transcoding support. The message must contain the key `call-id` and one
of the participant selection keys described under the `block DTMF` message (such as `from-tag`,
`address`, or `label`). Alternatively, the `all` flag can be set to play the media to all involved
call parties.

Starts playback of a provided media file to the selected call participant. The format of the media file
can be anything that is supported by *ffmpeg*, for example a `.wav` or `.mp3` file. It will automatically
be resampled and transcoded to the appropriate sampling rate and codec. The selected participant's first
listed (preferred) codec that is supported will be chosen for this purpose.
Encoder parameters such as bit rate can be set via the `codec-set` list
described above.

Media files can be provided through one of these keys:

* `file`

	Contains a string that points to a file on the local file system. File names can be relative
	to the daemon's working direction.

* `blob`

	Contains a binary blob (string) of the contents of a media file. Due to the limitations of the
	*ng* transport protocol, only very short files can be provided this way, and so this is primarily
	useful for testing and debugging.

* `db-id`

	Contains an integer. This requires the daemon to be configured for accessing a *MySQL* (or *MariaDB*)
	database via (at the minimum) the `mysql-host` and `mysql-query` config keys. The daemon will then
	retrieve the media file as a binary blob (not a file name!) from the database via the provided query.

* `repeat-times`

	Contains an integer. How many times to repeat playback of the media. Default is 1.

* `repeat-duration`

	Contains an integer. How much time in milliseconds is a playback of the media to be minimally iterated.
	E.g. if set to 10000ms and the playback's length is 1000ms, then this playback will be iterated
	10 times due to limitation set to 10000ms.
	If used together with `repeat-times` then the following logic takes place:
	if `repeat-duration` hits the trigger earlier, then this playback will be stopped,
	otherwise if the `repeat-duration` is still positive, but the `repeat-times` counter went down to 1,
	then still the playback is to be stopped.
	By default is disabled.

* `start-pos`

	Contains an integer. The start frame position to begin the playback from.

In addition to the `result` key, the response dictionary may contain the key `duration` if the length of
the media file could be determined. The duration is given as in integer representing milliseconds.

## `stop media` Message

Stops the playback previously started by a `play media` message. Media playback stops automatically when
the end of the media file is reached, so this message is only useful for prematurely stopping playback.
The same participant selection keys as for the `play media` message can and must be used. Will return the
last frame played in `last-frame-pos` key.

## `play DTMF` Message

Instructs *rtpengine* to inject a DTMF tone or event into a running audio stream. A call participant must
be selected in the same way as described under the `play media` message above (including the possibility
of using the `all` flag). The selected call participant is the one generating the DTMF event, not the
one receiving it.

The dictionary key `code` (or alternatively `digit`) must be present in the message,
indicating the DTMF event to be generated. It can
be either an integer with values 0-15, or a string containing a single character
(`0` - `9`, `*`, `#`, `A` - `D`). Additional optional dictionary keys are: `duration` indicating the duration
of the event in milliseconds (defaults to 250 ms, with a minimum of 100 and a maximum of 5000);
`volume` indicating the volume in absolute decibels (defaults to -8 dB, with 0 being the maximum volume and
positive integers being interpreted as negative); and `pause` indicating the pause in between consecutive
DTMF events in milliseconds (defaults to 100 ms, with a minimum of 100 and a maximum of 5000).

This message can be used to implement `application/dtmf-relay` or `application/dtmf` payloads carried
in SIP INFO messages. Multiple DTMF events can be queued up by issuing multiple consecutive
`play DTMF` messages.

If the destination participant supports the `telephone-event` RTP payload type, then it will be used to
send the DTMF event. Otherwise a PCM DTMF tone will be inserted into the audio stream. Audio samples
received during a generated DTMF event will be suppressed.

The call must be marked for DTMF injection using the `inject DTMF` flag used in both `offer` and `answer`
messages. Enabling this flag forces all audio to go through the transcoding engine, even if input and output
codecs are the same (similar to DTMF transcoding, see above).

## Music on hold functionality (MoH)

Only available if compiled with transcoding support.

This functionality is available only for the offer/answer model, hence no other scenarios like
publish or subscription related are supported with it.

The concept — is that MoH capabilities get advertised always at the beginning of
the call (original offer/answer exchange) and can be used later on to put the other side on hold.

MoH can be can be added for both sides: offerer and answerer.
Hence, for the offerer one will advertisde MoH support when calling `rtpengine_offer()` (or `rtpengine_manage()`)
and for the answerer side when calling `rtpengine_answer()` (or again `rtpengine_manage()`).

The one who advertises its MoH capabilities at the beginning of the call (first offer/answer exchange)
can later trigger MoH using the `sendonly` state (or alternatively `inactive`) and then unhold the other side
using `sendrecv` state (or alternatively no state advertised, which is equal to the `sendrecv` one).

MoH covers only audio type of media sessions, hence other media types, such as video, aren't accepted.
It should also be taken into account that there is no specific selection of media sections
to be held, thus — if one audio media puts the call on hold, the whole call is held.

During MoH being active, it's not possible to mix the MoH stream (sound file being played) with an audio stream
coming from the one who puts on hold. In other words, all egress media stream coming from the MoH originator
will be ignored until the other side gets unheld.

MoH also cannot be mixed with the `play media` functionality, the last one triggered will override previous one.
Although, in this case the behavior of rtpengine can also get unexpected.

MoH must be unheld to stop the media being sent towards the other side (recipient).
If MoH isn't unheld explicitly, then it will be stopped by the `moh-max-duration` config option.

To control a duration of MoH to be played, see `moh-max-duration` and `moh-max-repeats` configuration options.
By default MoH playback is limited to 1800 seconds (half an hour).

MoH stream will also be dropped in case of full dialog termination.

The MoH SDP session can also be marked with a specific (custom) session level attribute.
For that to enable see the `moh-attr-name` configuration option.

In order to protect against double MoH played, e.g. when inadvertently two rtpengine instances
in the interaction chain both try to trigger MoH, use `moh-prevent-double-hold` configuration option (true/false).
It works in combination with `moh-attr-name` config option, which must be defined in order to give a clue to rtpengine,
what is the session level attribute to be used for a double hold detection.

List of parameters to be given when advertising MoH capabilities:

- a sound source : `file`, `blob` or `db-id`.

	At least this parameter must be given.

	`file` — is a full path to a file including name stored on the local file system.

	`blob` — is a blob data (binary data) directly given (in form of string) via option flags to rtpengine.

	`db-id` - an integer which points to a data stored in DB.
	At least `mysql-host` and `mysql-query` must be configured to let the system read blob data from the DB backend.
	For more information see `play media` functionality and `mysql-*` configuration options.

- `mode` : `sendonly` or `sendrecv` (by default always `sendonly`).

	Which state gets advertised to the other side, which is being put on hold.
	In some cases `sendrecv` can be useful for specific client implementations,
	which don't want to see `sendonly` state in coming SDP.

	`mode` also has an in-dialog-only value called `reflect`.
	This one isn't meant to advertise own capabilities and can only be used with in-dialog SDP offers,
	assuming that at the beginning of the call session MoH capabilities were advertised by the capable side.

	This mode is only supposed to be used in cases when:
	* in-dialog offerer doesn't have own MoH capabilities; and
	* still wants the recipient to hear the MoH music; hence
	* an offerer checks whether a recipient is capable of MoH and launches a player based on the given capabilities;
	* the rest functions the same as with usual MoH hold.

	`mode=reflect` contradicts with `mode=sendrecv`/`mode=sendonly` which in its turn serves another purpose.

- `connection` type : `zero` (for now only one type is available).

  If set, then connection information (`c=` field) in the according media session
  will be set to all zeroes (`0.0.0.0`). So called zeroed-hold.
  Can be useful for some older client implementations using outdated standards.

Usage syntax:

		"moh" :
		{
			"<source>": "<value>",
			"mode": "<value>",
			"connection": "<value>"
		}

Usage example:

		"moh" :
		{
			"db-id": "123456789",
			"mode": "sendrecv",
			"connection": "zero"
		}

Another usage example:

		"moh" :
		{
			"file": "/tmp/music-on-hold.wav"
		}

## `statistics` Message

Returns a set of general statistics metrics with identical content and format as the `list jsonstats` CLI
command. Sample return dictionary:

	{
	  "statistics": {
	    "currentstatistics": {
	      "sessionsown": 0,
	      "sessionsforeign": 0,
	      "sessionstotal": 0,
	      "transcodedmedia": 0,
	      "packetrate": 0,
	      "byterate": 0,
	      "errorrate": 0
	    },
	    "totalstatistics": {
	      "uptime": "18",
	      "managedsessions": 0,
	      "rejectedsessions": 0,
	      "timeoutsessions": 0,
	      "silenttimeoutsessions": 0,
	      "finaltimeoutsessions": 0,
	      "offertimeoutsessions": 0,
	      "regularterminatedsessions": 0,
	      "forcedterminatedsessions": 0,
	      "relayedpackets": 0,
	      "relayedpacketerrors": 0,
	      "zerowaystreams": 0,
	      "onewaystreams": 0,
	      "avgcallduration": "0.000000"
	    },
	    "intervalstatistics": {
	      "totalcallsduration": "0.000000",
	      "minmanagedsessions": 0,
	      "maxmanagedsessions": 0,
	      "minofferdelay": "0.000000",
	      "maxofferdelay": "0.000000",
	      "avgofferdelay": "0.000000",
	      "minanswerdelay": "0.000000",
	      "maxanswerdelay": "0.000000",
	      "avganswerdelay": "0.000000",
	      "mindeletedelay": "0.000000",
	      "maxdeletedelay": "0.000000",
	      "avgdeletedelay": "0.000000",
	      "minofferrequestrate": 0,
	      "maxofferrequestrate": 0,
	      "avgofferrequestrate": 0,
	      "minanswerrequestrate": 0,
	      "maxanswerrequestrate": 0,
	      "avganswerrequestrate": 0,
	      "mindeleterequestrate": 0,
	      "maxdeleterequestrate": 0,
	      "avgdeleterequestrate": 0
	    },
	    "controlstatistics": {
	      "proxies": [
		{
		  "proxy": "127.0.0.1",
		  "pingcount": 0,
		  "offercount": 0,
		  "answercount": 0,
		  "deletecount": 0,
		  "querycount": 0,
		  "listcount": 0,
		  "startreccount": 0,
		  "stopreccount": 0,
		  "startfwdcount": 0,
		  "stopfwdcount": 0,
		  "blkdtmfcount": 0,
		  "unblkdtmfcount": 0,
		  "blkmedia": 0,
		  "unblkmedia": 0,
		  "playmedia": 0,
		  "stopmedia": 0,
		  "playdtmf": 0,
		  "statistics": 0,
		  "errorcount": 0
		}
	      ],
	      "totalpingcount": 0,
	      "totaloffercount": 0,
	      "totalanswercount": 0,
	      "totaldeletecount": 0,
	      "totalquerycount": 0,
	      "totallistcount": 0,
	      "totalstartreccount": 0,
	      "totalstopreccount": 0,
	      "totalstartfwdcount": 0,
	      "totalstopfwdcount": 0,
	      "totalblkdtmfcount": 0,
	      "totalunblkdtmfcount": 0,
	      "totalblkmedia": 0,
	      "totalunblkmedia": 0,
	      "totalplaymedia": 0,
	      "totalstopmedia": 0,
	      "totalplaydtmf": 0,
	      "totalstatistics": 0,
	      "totalerrorcount": 0
	    }
	  },
	  "result": "ok"
	}

## `publish` Message

Similar to an `offer` message except that it is used outside of an offer/answer
scenario. The media described by the SDP is published to *rtpengine* directly,
and other peer can then subscribe to the published media to receive a copy.

The message must include the key `sdp` which should describe `sendonly` media;
and the key `call-id` and `from-tag` to identify the publisher. Most other keys
and options supported by `offer` are also supported for `publish`.

The reply message will contain an answer SDP in `sdp`, but unlike with `offer`
this is not a rewritten version of the received SDP, but rather a `recvonly`
answer SDP generated by *rtpengine* locally. Only one codec for each media
section will be listed, and by default this will be the first supported codec
from the published media. This can be influenced with the `codec` options
described above, in particular the `accept` option.

The list of codecs given in the `accept` option is treated as a list of codec
preferences, with the first codec listed being the most preferred codec to be
accepted, and so on. It is allowable to list codecs that are not supported for
transcoding. If no codecs from the `accept` list are present in the offer, then
the first codec that is supported for transcoding is selected. If no such codec
is present, then the offer is rejected. The special string `any` can be given
in the `accept` list to influence this behaviour: If `any` is listed, then the
first codec from the offer is accepted even if it's not supported for
transcoding.

## `subscribe request` Message

This message is used to request subscription (i.e. receiving a copy of the
media) to one or multiple existing call participants, which must have been
created either through the offer/answer mechanism, or through the publish
mechanism.

A single call participant can be selected in the same way as described under
`block DTMF`. Multiple call participants can be selected either by using the
`all` keyword, in which case all call participants that were created through
the offer/answer mechanism will be selected, or by providing a list of tags
(from-tags) in the `from-tags` list.

This message then creates a new call participant, which corresponds to the
subscription. This new call participant will be identified by a newly generated
unique tag, or by the tag given in the `to-tag` key. If a label is to be set
for the newly created subscription, it can be set through `set-label`.

The reply message will contain a sendonly offer SDP in `sdp` which by default
will mirror the SDP of the call participant being subscribed to. If multiple
call participants are subscribed to at the same time, then this SDP will
contain multiple media sections, combined out of the media sections of all
selected call participants. This offer SDP can be manipulated with the same
flags as used in an `offer` message, including the option to manipulate the
codecs. The reply message will also contain the `from-tags` (corresponding to
the call participants being subscribed to) and the `to-tag` (corresponding to
the subscription, either generated or taken from the received message).

If a `subscribe request` is made for an existing `to-tag` then all existing
subscriptions for that `to-tag` are deleted before the new subscriptions are
created.

## `subscribe answer` Message

This message is expected to be received after responding to a `subscribe
request` message. The message should contain the same `to-tag` as the reply to
the `subscribe request` as well as the answer SDP in `sdp`.

By default, the answer SDP must accept all codecs that were presented in the
offer SDP (given in the reply to `subscribe request`). If not all codecs were
accepted, then the `subscribe answer` will be rejected. This behaviour can be
changed by including the `allow transcoding` flag in the message. If this flag
is present, then the answer SDP will be accepted as long as at least one valid
codec is present, and the media will be transcoded as required. This also holds
true if some codecs were added for transcoding in the `subscribe request`
message, which means that `allow transcoding` must always be included in
`subscribe answer` if any transcoding is to be allowed.

The reply message will simply indicate success or failure. If successful, media
forwarding will start to the endpoint given in the answer SDP.

## `unsubscribe` Message

This message is a counterpart to `subsscribe answer` to stop an established
subscription. The subscription to be stopped is identified by the `to-tag`.

## `connect` Message

This message makes it posible to directly connect the media of two call parties
without the need for a full offer/answer exchange. The required keys are
`call-id` to identify the call, and `from-tag and `to-tag` to identify the two
call parties to connect. The media will be connected in the same way as it
would through an offer/answer exchange. Transcoding will automaticaly be
engaged if needed.

It's also possible to connect two call parties from two different calls
(different call IDs). To do so, the second call ID must be given as
`to-call-id`, with the given `to-tag` then being one of the call parties from
that second call. Internally, both calls will be merged into a single call
object, with both call IDs then corresponding to the same call. This will be
visible in certain statistics (e.g. two call IDs appearing in the list, but
only one call being counted).
