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rfc:rfc4733

Network Working Group H. Schulzrinne Request for Comments: 4733 Columbia U. Obsoletes: 2833 T. Taylor Category: Standards Track Nortel

                                                         December 2006
RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals

Status of This Memo

 This document specifies an Internet standards track protocol for the
 Internet community, and requests discussion and suggestions for
 improvements.  Please refer to the current edition of the "Internet
 Official Protocol Standards" (STD 1) for the standardization state
 and status of this protocol.  Distribution of this memo is unlimited.

Copyright Notice

 Copyright (C) The IETF Trust (2006).

Abstract

 This memo describes how to carry dual-tone multifrequency (DTMF)
 signalling, other tone signals, and telephony events in RTP packets.
 It obsoletes RFC 2833.
 This memo captures and expands upon the basic framework defined in
 RFC 2833, but retains only the most basic event codes.  It sets up an
 IANA registry to which other event code assignments may be added.
 Companion documents add event codes to this registry relating to
 modem, fax, text telephony, and channel-associated signalling events.
 The remainder of the event codes defined in RFC 2833 are
 conditionally reserved in case other documents revive their use.
 This document provides a number of clarifications to the original
 document.  However, it specifically differs from RFC 2833 by removing
 the requirement that all compliant implementations support the DTMF
 events.  Instead, compliant implementations taking part in
 out-of-band negotiations of media stream content indicate what events
 they support.  This memo adds three new procedures to the RFC 2833
 framework: subdivision of long events into segments, reporting of
 multiple events in a single packet, and the concept and reporting of
 state events.

Schulzrinne & Taylor Standards Track [Page 1] RFC 4733 Telephony Events and Tones December 2006

Table of Contents

 1. Introduction ....................................................4
    1.1. Terminology ................................................4
    1.2. Overview ...................................................4
    1.3. Potential Applications .....................................5
    1.4. Events, States, Tone Patterns, and Voice-Encoded Tones .....6
 2. RTP Payload Format for Named Telephone Events ...................8
    2.1. Introduction ...............................................8
    2.2. Use of RTP Header Fields ...................................8
         2.2.1. Timestamp ...........................................8
         2.2.2. Marker Bit ..........................................8
    2.3. Payload Format .............................................8
         2.3.1. Event Field .........................................9
         2.3.2. E ("End") Bit .......................................9
         2.3.3. R Bit ...............................................9
         2.3.4. Volume Field ........................................9
         2.3.5. Duration Field ......................................9
    2.4. Optional Media Type Parameters ............................10
         2.4.1. Relationship to SDP ................................10
    2.5. Procedures ................................................11
         2.5.1. Sending Procedures .................................11
         2.5.2. Receiving Procedures ...............................16
    2.6. Congestion and Performance ................................19
         2.6.1. Performance Requirements ...........................20
         2.6.2. Reliability Mechanisms .............................20
         2.6.3. Adjusting to Congestion ............................22
 3. Specification of Event Codes for DTMF Events ...................23
    3.1. DTMF Applications .........................................23
    3.2. DTMF Events ...............................................25
    3.3. Congestion Considerations .................................25
 4. RTP Payload Format for Telephony Tones .........................26
    4.1. Introduction ..............................................26
    4.2. Examples of Common Telephone Tone Signals .................27
    4.3. Use of RTP Header Fields ..................................27
         4.3.1. Timestamp ..........................................27
         4.3.2. Marker Bit .........................................27
         4.3.3. Payload Format .....................................28
         4.3.4. Optional Media Type Parameters .....................29
    4.4. Procedures ................................................29
         4.4.1. Sending Procedures .................................29
         4.4.2. Receiving Procedures ...............................30
         4.4.3. Handling of Congestion .............................30
 5. Examples .......................................................31
 6. Security Considerations ........................................38

Schulzrinne & Taylor Standards Track [Page 2] RFC 4733 Telephony Events and Tones December 2006

 7. IANA Considerations ............................................38
    7.1. Media Type Registrations ..................................40
         7.1.1. Registration of Media Type audio/telephone-event ...40
         7.1.2. Registration of Media Type audio/tone ..............42
 8. Acknowledgements ...............................................43
 9. References .....................................................43
    9.1. Normative References ......................................43
    9.2. Informative References ....................................44
 Appendix A. Summary of Changes from RFC 2833 ......................46

Schulzrinne & Taylor Standards Track [Page 3] RFC 4733 Telephony Events and Tones December 2006

1. Introduction

1.1. Terminology

 In this document, the key words "MUST", "MUST NOT", "REQUIRED",
 "SHALL", "SHALL NOT", "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY",
 and "OPTIONAL" are to be interpreted as described in RFC 2119 [1].
 This document uses the following abbreviations:
 ANSam   Answer tone (amplitude modulated) [24]
 DTMF    Dual-Tone Multifrequency [10]
 IVR     Interactive Voice Response unit
 PBX     Private branch exchange (telephone system)
 PSTN    Public Switched (circuit) Telephone Network
 RTP     Real-time Transport Protocol [5]
 SDP     Session Description Protocol [9]

1.2. Overview

 This memo defines two RTP [5] payload formats, one for carrying
 dual-tone multifrequency (DTMF) digits and other line and trunk
 signals as events (Section 2), and a second one to describe general
 multifrequency tones in terms only of their frequency and cadence
 (Section 4).  Separate RTP payload formats for telephony tone signals
 are desirable since low-rate voice codecs cannot be guaranteed to
 reproduce these tone signals accurately enough for automatic
 recognition.  In addition, tone properties such as the phase
 reversals in the ANSam tone will not survive speech coding.  Defining
 separate payload formats also permits higher redundancy while
 maintaining a low bit rate.  Finally, some telephony events such as
 "on-hook" occur out-of-band and cannot be transmitted as tones.
 The remainder of this section provides the motivation for defining
 the payload types described in this document.  Section 2 defines the
 payload format and associated procedures for use of named events.
 Section 3 describes the events for which event codes are defined in
 this document.  Section 4 describes the payload format and associated
 procedures for tone representations.  Section 5 provides some
 examples of encoded events, tones, and combined payloads.  Section 6
 deals with security considerations.  Section 7 defines the IANA
 requirements for registration of event codes for named telephone

Schulzrinne & Taylor Standards Track [Page 4] RFC 4733 Telephony Events and Tones December 2006

 events, establishes the initial content of that registry, and
 provides the media type registrations for the two payload formats.
 Appendix A describes the changes from RFC 2833 [12] and in particular
 indicates the disposition of the event codes defined in [12].

1.3. Potential Applications

 The payload formats described here may be useful in a number of
 different scenarios.
 On the sending side, there are two basic possibilities: either the
 sending side is an end system that originates the signals itself, or
 it is a gateway with the task of propagating incoming telephone
 signals into the Internet.
 On the receiving side, there are more possibilities.  The first is
 that the receiver must propagate tone signalling accurately into the
 PSTN for machine consumption.  One example of this is a gateway
 passing DTMF tones to an IVR.  In this scenario, frequencies,
 amplitudes, tone durations, and the durations of pauses between tones
 are all significant, and individual tone signals must be delivered
 reliably and in order.
 In a second receiving scenario, the receiver must play out tones for
 human consumption.  Typically, rather than a series of tone signals
 each with its own meaning, the content will consist of a single tone
 played out continuously or a single sequence of tones and possibly
 silence, repeated cyclically for some period of time.  Often the end
 of the tone playout will be triggered by an event fed back in the
 other direction, using either in- or out-of-band means.  Examples of
 this are dial tone or busy tone.
 The relationship between position in the network and the tones to be
 played out is a complicating factor in this scenario.  In the phone
 network, tones are generated at different places, depending on the
 switching technology and the nature of the tone.  This determines,
 for example, whether a person making a call to a foreign country
 hears her local tones she is familiar with or the tones as used in
 the country called.
 For analog lines, dial tone is always generated by the local switch.
 Integrated Services Digital Network (ISDN) terminals may generate
 dial tone locally and then send a Q.931 [22] SETUP message containing
 the dialed digits.  If the terminal just sends a SETUP message
 without any Called Party digits, then the switch does digit
 collection (provided by the terminal as KEYPAD key press digit
 information within Called Party or Keypad Facility Information
 Elements (IEs) of INFORMATION messages), and provides dial tone over

Schulzrinne & Taylor Standards Track [Page 5] RFC 4733 Telephony Events and Tones December 2006

 the B-channel.  The terminal can either use the audio signal on the
 B-channel or use the Q.931 messages to trigger locally generated dial
 tone.
 Ringing tone (also called ringback tone) is generated by the local
 switch at the callee, with a one-way voice path opened up as soon as
 the callee's phone rings.  (This reduces the chance of clipping the
 called party's response just after answer.  It also permits pre-
 answer announcements or in-band call-progress indications to reach
 the caller before or in lieu of a ringing tone.)  Congestion tone and
 special information tones can be generated by any of the switches
 along the way, and may be generated by the caller's switch based on
 ISDN User Part (ISUP) messages received.  Busy tone is generated by
 the caller's switch, triggered by the appropriate ISUP message, for
 analog instruments, or the ISDN terminal.
 In the third scenario, an end system is directly connected to the
 Internet and processes the incoming media stream directly.  There is
 no need to regenerate tone signals, so that time alignment and power
 levels are not relevant.  These systems rely on sending systems to
 generate events in place of tones and do not perform their own audio
 waveform analysis.  An example of such a system is an Internet
 interactive voice response (IVR) system.
 In circumstances where exact timing alignment between the audio
 stream and the DTMF digits or other events is not important and data
 is sent unicast, as in the IVR example, it may be preferable to use a
 reliable control protocol rather than RTP packets.  In those
 circumstances, this payload format would not be used.
 Note that in a number of these cases it is possible that the gateway
 or end system will be both a sender and receiver of telephone
 signals.  Sometimes the same class of signals will be sent as
 received -- in the case of "RTP trunking" or voice-band data, for
 instance.  In other cases, such as that of an end system serving
 analogue lines, the signals sent will be in a different class from
 those received.

1.4. Events, States, Tone Patterns, and Voice-Encoded Tones

 This document provides the means for in-band transport over the
 Internet of two broad classes of signalling information: in-band
 tones or tone sequences, and signals sent out-of-band in the PSTN.
 Tone signals can be carried using any of the three methods listed
 below.  Depending on the application, it may be desirable to carry
 the signalling information in more than one form at once.

Schulzrinne & Taylor Standards Track [Page 6] RFC 4733 Telephony Events and Tones December 2006

 1.  The gateway or end system can change to a higher-bandwidth codec
     such as G.711 [19] when tone signals are to be conveyed.  See new
     ITU-T Recommendation V.152 [26] for a formal treatment of this
     approach.  Alternatively, for fax, text, or modem signals
     respectively, a specialized transport such as T.38 [23], RFC 4103
     [15], or V.150.1 modem relay [25] may be used.  Finally, 64
     kbit/s channels may be carried transparently using the RFC 4040
     Clearmode payload type [14].  These methods are out of scope of
     the present document, but may be used along with the payload
     types defined here.
 2.  The sending gateway can simply measure the frequency components
     of the voice-band signals and transmit this information to the
     RTP receiver using the tone representation defined in this
     document (Section 4).  In this mode, the gateway makes no attempt
     to discern the meaning of the tones, but simply distinguishes
     tones from speech signals.  An end system may use the same
     approach using configured rather than measured frequencies.
     All tone signals in use in the PSTN and meant for human
     consumption are sequences of simple combinations of sine waves,
     either added or modulated.  (However, some modem signals such as
     the ANSam tone [24] or systems dependent on phase shift keying
     cannot be conveyed so simply.)
 3.  As a third option, a sending gateway can recognize tones such as
     ringing or busy tone or DTMF digit '0', and transmit a code that
     identifies them using the telephone-event payload defined in this
     document (Section 2).  The receiver then produces a tone signal
     or other indication appropriate to the signal.  Generally, since
     the recognition of signals at the sender often depends on their
     on/off pattern or the sequence of several tones, this recognition
     can take several seconds.  On the other hand, the gateway may
     have access to the actual signalling information that generates
     the tones and thus can generate the RTP packet immediately,
     without the detour through acoustic signals.
 The third option (use of named events) is the only feasible method
 for transmitting out-of-band PSTN signals as content within RTP
 sessions.

Schulzrinne & Taylor Standards Track [Page 7] RFC 4733 Telephony Events and Tones December 2006

2. RTP Payload Format for Named Telephone Events

2.1. Introduction

 The RTP payload format for named telephone events is designated as
 "telephone-event", the media type as "audio/telephone-event".  In
 accordance with current practice, this payload format does not have a
 static payload type number, but uses an RTP payload type number
 established dynamically and out-of-band.  The default clock frequency
 is 8000 Hz, but the clock frequency can be redefined when assigning
 the dynamic payload type.
 Named telephone events are carried as part of the audio stream and
 MUST use the same sequence number and timestamp base as the regular
 audio channel to simplify the generation of audio waveforms at a
 gateway.  The named telephone-event payload type can be considered to
 be a very highly-compressed audio codec and is treated the same as
 other codecs.

2.2. Use of RTP Header Fields

2.2.1. Timestamp

 The event duration described in Section 2.5 begins at the time given
 by the RTP timestamp.  For events that span multiple RTP packets, the
 RTP timestamp identifies the beginning of the event, i.e., several
 RTP packets may carry the same timestamp.  For long-lasting events
 that have to be split into segments (see below, Section 2.5.1.3), the
 timestamp indicates the beginning of the segment.

2.2.2. Marker Bit

 The RTP marker bit indicates the beginning of a new event.  For long-
 lasting events that have to be split into segments (see below,
 Section 2.5.1.3), only the first segment will have the marker bit
 set.

2.3. Payload Format

 The payload format for named telephone events is shown in Figure 1.
  0                   1                   2                   3
  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |     event     |E|R| volume    |          duration             |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
               Figure 1: Payload Format for Named Events

Schulzrinne & Taylor Standards Track [Page 8] RFC 4733 Telephony Events and Tones December 2006

2.3.1. Event Field

 The event field is a number between 0 and 255 identifying a specific
 telephony event.  An IANA registry of event codes for this field has
 been established (see IANA Considerations, Section 7).  The initial
 content of this registry consists of the events defined in Section 3.

2.3.2. E ("End") Bit

 If set to a value of one, the "end" bit indicates that this packet
 contains the end of the event.  For long-lasting events that have to
 be split into segments (see below, Section 2.5.1.3), only the final
 packet for the final segment will have the E bit set.

2.3.3. R Bit

 This field is reserved for future use.  The sender MUST set it to
 zero, and the receiver MUST ignore it.

2.3.4. Volume Field

 For DTMF digits and other events representable as tones, this field
 describes the power level of the tone, expressed in dBm0 after
 dropping the sign.  Power levels range from 0 to -63 dBm0.  Thus,
 larger values denote lower volume.  This value is defined only for
 events for which the documentation indicates that volume is
 applicable.  For other events, the sender MUST set volume to zero and
 the receiver MUST ignore the value.

2.3.5. Duration Field

 The duration field indicates the duration of the event or segment
 being reported, in timestamp units, expressed as an unsigned integer
 in network byte order.  For a non-zero value, the event or segment
 began at the instant identified by the RTP timestamp and has so far
 lasted as long as indicated by this parameter.  The event may or may
 not have ended.  If the event duration exceeds the maximum
 representable by the duration field, the event is split into several
 contiguous segments as described below (Section 2.5.1.3).
 The special duration value of zero is reserved to indicate that the
 event lasts "forever", i.e., is a state and is considered to be
 effective until updated.  A sender MUST NOT transmit a zero duration
 for events other than those defined as states.  The receiver SHOULD
 ignore an event report with zero duration if the event is not a
 state.

Schulzrinne & Taylor Standards Track [Page 9] RFC 4733 Telephony Events and Tones December 2006

 Events defined as states MAY contain a non-zero duration, indicating
 that the sender intends to refresh the state before the time duration
 has elapsed ("soft state").
    For a sampling rate of 8000 Hz, the duration field is sufficient
    to express event durations of up to approximately 8 seconds.

2.4. Optional Media Type Parameters

 As indicated in the media type registration for named events in
 Section 7.1.1, the telephone-event media type supports two optional
 parameters: the "events" parameter and the "rate" parameter.
 The "events" parameter lists the events supported by the
 implementation.  Events are listed as one or more comma-separated
 elements.  Each element can be either a single integer providing the
 value of an event code or an integer followed by a hyphen and a
 larger integer, presenting a range of consecutive event code values.
 The list does not have to be sorted.  No white space is allowed in
 the argument.  The union of all of the individual event codes and
 event code ranges designates the complete set of event numbers
 supported by the implementation.
 The "rate" parameter describes the sampling rate, in Hertz, and hence
 the units for the RTP timestamp and event duration fields.  The
 number is written as an integer.  If omitted, the default value is
 8000 Hz.

2.4.1. Relationship to SDP

 The recommended mapping of media type optional parameters to SDP is
 given in Section 3 of RFC 3555 [6].  The "rate" media type parameter
 for the named event payload type follows this convention: it is
 expressed as usual as the <clock rate> component of the a=rtpmap:
 attribute line.
 The "events" media type parameter deviates from the convention
 suggested in RFC 3555 because it omits the string "events=" before
 the list of supported events.
    a=fmtp:<format> <list of values>
 The list of values has the format and meaning described above.

Schulzrinne & Taylor Standards Track [Page 10] RFC 4733 Telephony Events and Tones December 2006

 For example, if the payload format uses the payload type number 100,
 and the implementation can handle the DTMF tones (events 0 through
 15) and the dial and ringing tones (assuming as an example that these
 were defined as events with codes 66 and 70, respectively), it would
 include the following description in its SDP message:
    m=audio 12346 RTP/AVP 100
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-15,66,70
 The following sample media type definition corresponds to the SDP
 example above:
    audio/telephone-event;events="0-15,66,70";rate="8000"

2.5. Procedures

 This section defines the procedures associated with the named event
 payload type.  Additional procedures may be specified in the
 documentation associated with specific event codes.

2.5.1. Sending Procedures

2.5.1.1. Negotiation of Payloads

 Events are usually sent in combination with or alternating with other
 payload types.  Payload negotiation may specify separate event and
 other payload streams, or it may specify a combined stream that mixes
 other payload types with events using RFC 2198 [2] redundancy
 headers.  The purpose of using a combined stream may be for debugging
 or to ease the transition between general audio and events.
 Negotiation of payloads between sender and receiver is achieved by
 out-of-band means, using SDP, for example.
 The sender SHOULD indicate what events it supports, using the
 optional "events" parameter associated with the telephone-event media
 type.  If the sender receives an "events" parameter from the
 receiver, it MUST restrict the set of events it sends to those listed
 in the received "events" parameter.  For backward compatibility, if
 no "events" parameter is received, the sender SHOULD assume support
 for the DTMF events 0-15 but for no other events.
 Events MAY be sent in combination with older events using RFC 2198
 [2] redundancy.  Section 2.5.1.4 describes how this can be used to
 avoid packet and RTP header overheads when retransmitting final event
 reports.  Section 2.6 discusses the use of additional levels of RFC
 2198 redundancy to increase the probability that at least one copy of

Schulzrinne & Taylor Standards Track [Page 11] RFC 4733 Telephony Events and Tones December 2006

 the report of the end of an event reaches the receiver.  The
 following SDP shows an example of such usage, where G.711 audio
 appears in a separate stream, and the primary component of the
 redundant payload is events.
    m=audio 12344 RTP/AVP 99
    a=rtpmap:99 pcmu/8000
    m=audio 12346 RTP/AVP 100 101
    a=rtpmap:100 red/8000/1
    a=fmtp:100 101/101/101
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
 When used in accordance with the offer-answer model (RFC 3264 [4]),
 the SDP a=ptime: attribute indicates the packetization period that
 the author of the session description expects when receiving media.
 This value does not have to be the same in both directions.  The
 appropriate period may vary with the application, since increased
 packetization periods imply increased end-to-end response times in
 instances where one end responds to events reported from the other.
 Negotiation of telephone-events sessions using SDP MAY specify such
 differences by separating events corresponding to different
 applications into different streams.  In the example below, events
 0-15 are DTMF events, which have a fairly wide tolerance on timing.
 Events 32-49 and 52-60 are events related to data transmission and
 are subject to end-to-end response time considerations.  As a result,
 they are assigned a smaller packetization period than the DTMF
 events.
    m=audio 12344 RTP/AVP 99
    a=rtpmap:99 telephone-event/8000
    a=fmtp:99 0-15
    a=ptime:50
    m=audio 12346 RTP/AVP 100
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 32-49,52-60
    a=ptime:30
 For further discussion of packetization periods see Section 2.6.3.

2.5.1.2. Transmission of Event Packets

 DTMF digits and other named telephone events are carried as part of
 the audio stream, and they MUST use the same sequence number and
 timestamp base as the regular audio channel to simplify the
 generation of audio waveforms at a gateway.

Schulzrinne & Taylor Standards Track [Page 12] RFC 4733 Telephony Events and Tones December 2006

 An audio source SHOULD start transmitting event packets as soon as it
 recognizes an event and continue to send updates until the event has
 ended.  The update packets MUST have the same RTP timestamp value as
 the initial packet for the event, but the duration MUST be increased
 to reflect the total cumulative duration since the beginning of the
 event.
 The first packet for an event MUST have the M bit set.  The final
 packet for an event MUST have the E bit set, but setting of the "E"
 bit MAY be deferred until the final packet is retransmitted (see
 Section 2.5.1.4).  Intermediate packets for an event MUST NOT have
 either the M bit or the E bit set.
 Sending of a packet with the E bit set is OPTIONAL if the packet
 reports two events that are defined as mutually exclusive states, or
 if the final packet for one state is immediately followed by a packet
 reporting a mutually exclusive state.  (For events defined as states,
 the appearance of a mutually exclusive state implies the end of the
 previous state.)
 A source has wide latitude as to how often it sends event updates.  A
 natural interval is the spacing between non-event audio packets.
 (Recall that a single RTP packet can contain multiple audio frames
 for frame-based codecs and that the packet interval can vary during a
 session.)  Alternatively, a source MAY decide to use a different
 spacing for event updates, with a value of 50 ms RECOMMENDED.
 Timing information is contained in the RTP timestamp, allowing
 precise recovery of inter-event times.  Thus, the sender does not in
 theory need to maintain precise or consistent time intervals between
 event packets.  However, the sender SHOULD minimize the need for
 buffering at the receiving end by sending event reports at constant
 intervals.
    DTMF digits and other tone events are sent incrementally to avoid
    having the receiver wait for the completion of the event.  In some
    cases (for example, data session startup protocols), waiting until
    the end of a tone before reporting it will cause the session to
    fail.  In other cases, it will simply cause undesirable delays in
    playout at the receiving end.
 For robustness, the sender SHOULD retransmit "state" events
 periodically.

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2.5.1.3. Long-Duration Events

 If an event persists beyond the maximum duration expressible in the
 duration field (0xFFFF), the sender MUST send a packet reporting this
 maximum duration but MUST NOT set the E bit in this packet.  The
 sender MUST then begin reporting a new "segment" with the RTP
 timestamp set to the time at which the previous segment ended and the
 duration set to the cumulative duration of the new segment.  The M
 bit of the first packet reporting the new segment MUST NOT be set.
 The sender MUST repeat this procedure as required until the end of
 the complete event has been reached.  The final packet for the
 complete event MUST have the E bit set (either on initial
 transmission or on retransmission as described below).

2.5.1.3.1. Exceptional Procedure for Combined Payloads

 If events are combined as a redundant payload with another payload
 type using RFC 2198 [2] redundancy, the above procedure SHALL be
 applied, but using a maximum duration that ensures that the timestamp
 offset of the oldest generation of events in an RFC 2198 packet never
 exceeds 0x3FFF.  If the sender is using a constant packetization
 period, the maximum segment duration can be calculated from the
 following formula:
    maximum duration = 0x3FFF - (R-1)*(packetization period in
    timestamp units)
 where R is the highest redundant layer number consisting of event
 payload.
    The RFC 2198 redundancy header timestamp offset value is only 14
    bits, compared with the 16 bits in the event payload duration
    field.  Since with other payloads the RTP timestamp typically
    increments for each new sample, the timestamp offset value becomes
    limiting on reported event duration.  The limit becomes more
    constraining when older generations of events are also included in
    the combined payload.

2.5.1.4. Retransmission of Final Packet

 The final packet for each event and for each segment SHOULD be sent a
 total of three times at the interval used by the source for updates.
 This ensures that the duration of the event or segment can be
 recognized correctly even if an instance of the last packet is lost.
 A sender MAY use RFC 2198 [2] with up to two levels of redundancy to
 combine retransmissions with reports of new events, thus saving on
 header overheads.  In this usage, the primary payload is new event

Schulzrinne & Taylor Standards Track [Page 14] RFC 4733 Telephony Events and Tones December 2006

 reports, while the first and (if necessary) second levels of
 redundancy report first and second retransmissions of final event
 reports.  Within a session negotiated to allow such usage, packets
 containing the RFC 2198 payload SHOULD NOT be sent except when both
 primary and retransmitted reports are to be included.  All other
 packets of the session SHOULD contain only the simple, non-redundant
 telephone-event payload.  Note that the expected proportion of simple
 versus redundant packets affects the order in which they should be
 specified on an SDP m= line.
    There is little point in sending initial or interim event reports
    redundantly because each succeeding packet describes the event
    fully (except for typically irrelevant variations in volume).
 A sender MAY delay setting the E bit until retransmitting the last
 packet for a tone, rather than setting the bit on its first
 transmission.  This avoids having to wait to detect whether the tone
 has indeed ended.  Once the sender has set the E bit for a packet, it
 MUST continue to set the E bit for any further retransmissions of
 that packet.

2.5.1.5. Packing Multiple Events into One Packet

 Multiple named events can be packed into a single RTP packet if and
 only if the events are consecutive and contiguous, i.e., occur
 without overlap and without pause between them, and if the last event
 packed into a packet occurs quickly enough to avoid excessive delays
 at the receiver.
 This approach is similar to having multiple frames of frame-based
 audio in one RTP packet.
 The constraint that packed events not overlap implies that events
 designated as states can be followed in a packet only by other state
 events that are mutually exclusive to them.  The constraint itself is
 needed so that the beginning time of each event can be calculated at
 the receiver.
 In a packet containing events packed in this way, the RTP timestamp
 MUST identify the beginning of the first event or segment in the
 packet.  The M bit MUST be set if the packet records the beginning of
 at least one event.  (This will be true except when the packet
 carries the end of one segment and the beginning of the next segment
 of the same long-lasting event.)  The E bit and duration for each
 event in the packet MUST be set using the same rules as if that event
 were the only event contained in the packet.

Schulzrinne & Taylor Standards Track [Page 15] RFC 4733 Telephony Events and Tones December 2006

2.5.1.6. RTP Sequence Number

 The RTP sequence number MUST be incremented by one in each successive
 RTP packet sent.  Incrementing applies to retransmitted as well as
 initial instances of event reports, to permit the receiver to detect
 lost packets for RTP Control Protocol (RTCP) receiver reports.

2.5.2. Receiving Procedures

2.5.2.1. Indication of Receiver Capabilities Using SDP

 Receivers can indicate which named events they can handle, for
 example, by using the Session Description Protocol (RFC 4566 [9]).
 SDP descriptions using the event payload MUST contain an fmtp format
 attribute that lists the event values that the receiver can process.

2.5.2.2. Playout of Tone Events

 In the gateway scenario, an Internet telephony gateway connecting a
 packet voice network to the PSTN re-creates the DTMF or other tones
 and injects them into the PSTN.  Since, for example, DTMF digit
 recognition takes several tens of milliseconds, the first few
 milliseconds of a digit will arrive as regular audio packets.  Thus,
 careful time and power (volume) alignment between the audio samples
 and the events is needed to avoid generating spurious digits at the
 receiver.  The receiver may also choose to delay playout of the tones
 by some small interval after playout of the preceding audio has
 ended, to ensure that downstream equipment can discriminate the tones
 properly.
 Some implementations send events and encoded audio packets (e.g.,
 PCMU or the codec used for speech signals) for the same time instant
 for the duration of the event.  It is RECOMMENDED that gateways
 render only the telephone-event payload once it is received, since
 the audio may contain spurious tones introduced by the audio
 compression algorithm.  However, it is anticipated that these extra
 tones in general should not interfere with recognition at the far
 end.
 Receiver implementations MAY use different algorithms to create
 tones, including the two described here.  (Note that not all
 implementations have the need to re-create a tone; some may only care
 about recognizing the events.)  With either algorithm, a receiver may
 impose a playout delay to provide robustness against packet loss or
 delay.  The tradeoff between playout delay and other factors is
 discussed further in Section 2.6.3.

Schulzrinne & Taylor Standards Track [Page 16] RFC 4733 Telephony Events and Tones December 2006

 In the first algorithm, the receiver simply places a tone of the
 given duration in the audio playout buffer at the location indicated
 by the timestamp.  As additional packets are received that extend the
 same tone, the waveform in the playout buffer is extended
 accordingly.  (Care has to be taken if audio is mixed, i.e., summed,
 in the playout buffer rather than simply copied.)  Thus, if a packet
 in a tone lasting longer than the packet interarrival time gets lost
 and the playout delay is short, a gap in the tone may occur.
 Alternatively, the receiver can start a tone and play it until one of
 the following occurs:
 o  it receives a packet with the E bit set;
 o  it receives the next tone, distinguished by a different timestamp
    value (noting that new segments of long-duration events also
    appear with a new timestamp value);
 o  it receives an alternative non-event media stream (assuming none
    was being received while the event stream was active); or
 o  a given time period elapses.
 This is more robust against packet loss, but may extend the tone
 beyond its original duration if all retransmissions of the last
 packet in an event are lost.  Limiting the time period of extending
 the tone is necessary to avoid that a tone "gets stuck".  This
 algorithm is not a license for senders to set the duration field to
 zero; it MUST be set to the current duration as described, since this
 is needed to create accurate events if the first event packet is
 lost, among other reasons.
 Regardless of the algorithm used, the tone SHOULD NOT be extended by
 more than three packet interarrival times.  A slight extension of
 tone durations and shortening of pauses is generally harmless.
 A receiver SHOULD NOT restart a tone once playout has stopped.  It
 MAY do so if the tone is of a type meant for human consumption or is
 one for which interruptions will not cause confusion at the receiving
 device.
 If a receiver receives an event packet for an event that it is not
 currently playing out and the packet does not have the M bit set,
 earlier packets for that event have evidently been lost.  This can be
 confirmed by gaps in the RTP sequence number.  The receiver MAY
 determine on the basis of retained history and the timestamp and

Schulzrinne & Taylor Standards Track [Page 17] RFC 4733 Telephony Events and Tones December 2006

 event code of the current packet that it corresponds to an event
 already played out and lapsed.  In that case, further reports for the
 event MUST be ignored, as indicated in the previous paragraph.
 If, on the other hand, the event has not been played out at all, the
 receiver MAY attempt to play the event out to the complete duration
 indicated in the event report.  The appropriate behavior will depend
 on the event type, and requires consideration of the relationship of
 the event to audio media flows and whether correct event duration is
 essential to the correct operation of the media session.
 A receiver SHOULD NOT rely on a particular event packet spacing, but
 instead MUST use the event timestamps and durations to determine
 timing and duration of playout.
 The receiver MUST calculate jitter for RTCP receiver reports based on
 all packets with a given timestamp.  Note: The jitter value should
 primarily be used as a means for comparing the reception quality
 between two users or two time periods, not as an absolute measure.
 If a zero volume is indicated for an event for which the volume field
 is defined, then the receiver MAY reconstruct the volume from the
 volume of non-event audio or MAY use the nominal value specified by
 the ITU Recommendation or other document defining the tone.  This
 ensures backwards compatibility with RFC 2833 [12], where the volume
 field was defined only for DTMF events.

2.5.2.3. Long-Duration Events

 If an event report is received with duration equal to the maximum
 duration expressible in the duration field (0xFFFF) and the E bit for
 the report is not set, the event report may mark the end of a segment
 generated according to the procedures of Section 2.5.1.3.  If another
 report for the same event type is received, the receiver MUST compare
 the RTP timestamp for the new event with the sum of the RTP timestamp
 of the previous report plus the duration (0xFFFF).  The receiver uses
 the absence of a gap between the events to detect that it is
 receiving a single long-duration event.
 The total duration of a long-duration event is (obviously) the sum of
 the durations of the segments used to report it.  This is equal to
 the duration of the final segment (as indicated in the final packet
 for that segment), plus 0xFFFF multiplied by the number of segments
 preceding the final segment.

Schulzrinne & Taylor Standards Track [Page 18] RFC 4733 Telephony Events and Tones December 2006

2.5.2.3.1. Exceptional Procedure for Combined Payloads

 If events are combined as a redundant payload with another payload
 type using RFC 2198 [2] redundancy, segments are generated at
 intervals of 0x3FFF or less, rather than 0xFFFF, as required by the
 procedures of Section 2.5.1.3.1 in this case.  If a receiver is using
 the events component of the payload, event duration may be only an
 approximate indicator of division into segments, but the lack of an E
 bit and the adjacency of two reports with the same event code are
 strong indicators in themselves.

2.5.2.4. Multiple Events in a Packet

 The procedures of Section 2.5.1.5 require that if multiple events are
 reported in the same packet, they are contiguous and non-overlapping.
 As a result, it is not strictly necessary for the receiver to know
 the start times of the events following the first one in order to
 play them out -- it needs only to respect the duration reported for
 each event.  Nevertheless, if knowledge of the start time for a given
 event after the first one is required, it is equal to the sum of the
 start time of the preceding event plus the duration of the preceding
 event.

2.5.2.5. Soft States

 If the duration of a soft state event expires, the receiver SHOULD
 consider the value of the state to be "unknown" unless otherwise
 indicated in the event documentation.

2.6. Congestion and Performance

 Packet transmission through the Internet is marked by occasional
 periods of congestion lasting on the order of second, during which
 network delay, jitter, and packet loss are all much higher than they
 are in between these periods.  Reference [28] characterizes this
 phenomenon.  Well-behaved applications are expected, preferably, to
 reduce their demands on the network during such periods of
 congestion.  At the least, they should not increase their demands.
 This section explores both application performance and the
 possibilities for good behavior in the face of congestion.

Schulzrinne & Taylor Standards Track [Page 19] RFC 4733 Telephony Events and Tones December 2006

2.6.1. Performance Requirements

 Typically, an implementation of the telephone-event payload will aim
 to limit the rate at which each of the following impairments occurs:
 a.  an event encoded at the sender fails to be played out at the
     receiver, either because the event report is lost or because it
     arrives after playout of later content has started;
 b.  the start of playout of an event at the receiver is delayed
     relative to other events or other media operating on the same
     timestamp base;
 c.  the duration of playout of a given event differs from the correct
     duration as detected at the sender by more than a given amount;
 d.  gaps occur in playout of a given event;
 e.  end-to-end delay for the media stream exceeds a given value.
 The relative importance of these constraints varies between
 applications.

2.6.2. Reliability Mechanisms

 To improve reliability, all payload types including telephone-events
 can use a jitter buffer, i.e., impose a playout delay, at the
 receiving end.  This mechanism addresses the first four requirements
 listed above, but at the expense of the last one.
 The named event procedures provide two complementary redundancy
 mechanisms to deal with lost packets:
 a.  Intra-event updates:
     Events that last longer than one packetization period (e.g., 50
     ms) are updated periodically, so that the receiver can
     reconstruct the event and its duration if it receives any of the
     update packets, albeit with delay.
     During an event, the RTP event payload format provides
     incremental updates on the event.  The error resiliency afforded
     by this mechanism depends on whether the first or second
     algorithm in Section 2.5.2.2 is used and on the playout delay at
     the receiver.  For example, if the receiver uses the first
     algorithm and only places the current duration of tone signal in
     the playout buffer, for a playout delay of 120 ms and a

Schulzrinne & Taylor Standards Track [Page 20] RFC 4733 Telephony Events and Tones December 2006

     packetization interval of 50 ms, two packets in a row can get
     lost without causing a premature end of the tone generated.
 b.  Repeat last event packet:
     As described in Section 2.5.1.4, the last report for an event is
     transmitted a total of three times.  This mechanism adds
     robustness to the reporting of the end of an event.
     It may be necessary to extend the level of redundancy to achieve
     requirement a) (in Section 2.6.1) in a specific network
     environment.  Taking the 25-30% loss rate during congestion
     periods illustrated in [28] as typical, and setting an objective
     that at least 99% of end-of-event reports will eventually get
     through to the receiver under these conditions, simple
     probability calculations indicate that each event completion has
     to be reported four times.  This is one more level of redundancy
     than required by the basic "Repeat last event packet" algorithm.
     Of course, the objective is probably unrealistically stringent;
     it was chosen to make a point.
     Where Section 2.5.1.4 indicates that it is appropriate to use the
     RFC 2198 [2] audio redundancy mechanism to carry retransmissions
     of final event reports, this mechanism MAY also be used to extend
     the number of final report retransmissions.  This is done by
     using more than two levels of redundancy when necessary.  The use
     of RFC 2198 helps to mitigate the extra bandwidth demands that
     would be imposed simply by retransmitting final event packets
     more than three times.
 These two redundancy mechanisms clearly address requirement a) in the
 previous section.  They also help meet requirement c), to the extent
 that the redundant packets arrive before playout of the events they
 report is due to expire.  They are not helpful in meeting the other
 requirements, although they do not directly cause impairments
 themselves in the way that a large jitter buffer increases end-to-end
 delay.
 The playout algorithm is an additional mechanism for meeting the
 performance requirements.  In particular, using the second algorithm
 in Section 2.5.2.2 will meet requirement d) of the previous section
 by preventing gaps in playout, but at the potential cost of increases
 in duration (requirement c)).
 Finally, there is an interaction between the packetization period
 used by a sender, the playout delay used by the receiver, and the
 vulnerability of an event flow to packet losses.  Assuming packet
 losses are independent, a shorter packetization interval means that

Schulzrinne & Taylor Standards Track [Page 21] RFC 4733 Telephony Events and Tones December 2006

 the receiver can use a smaller playout delay to recover from a given
 number of consecutive packet losses, at any stage of event playout.
 This improves end-to-end delays in applications where that matters.
 In view of the tradeoffs between the different reliability
 mechanisms, documentation of specific events SHOULD include a
 discussion of the appropriate design decisions for the applications
 of those events.  This mandate is repeated in the section on IANA
 considerations.

2.6.3. Adjusting to Congestion

 So far, the discussion has been about meeting performance
 requirements.  However, there is also the question of whether
 applications of events can adapt to congestion to the point that they
 reduce their demands on the networks during congestion.  In theory
 this can be done for events by increasing the packetization interval,
 so that fewer packets are sent per second.  This has to be
 accompanied by an increased playout delay at the receiving end.
 Coordination between the two ends for this purpose is an interesting
 issue in itself.  If it is done, however, such an action implies a
 one-time gap or extended playout of an event when the packetization
 interval is first extended, as well as increased end-to-end delay
 during the whole period of increased playout delay.
 The benefit from such a measure varies primarily depending on the
 average duration of the events being handled.  In the worst case, as
 a first example shows, the reduction in aggregate bandwidth usage due
 to an increased packetization interval may be quite modest.  Suppose
 the average event duration is 3.33 ms (V.21 bits, for instance).
 Suppose further that four transmissions in total are required for a
 given event report to meet the loss objective.  Table 1 shows the
 impact of varying packetization intervals on the aggregate bit rate
 of the media stream.
 +--------------------+-----------+---------------+------------------+
 | Packetization      | Packets/s |     IP Packet |     Total IP Bit |
 | Interval (ms)      |           |   Size (bits) |    Rate (bits/s) |
 +--------------------+-----------+---------------+------------------+
 | 50                 |        20 |          2440 |            48800 |
 | 33.3               |        30 |          1800 |            54000 |
 | 25                 |        40 |          1480 |            59200 |
 | 20                 |        50 |          1288 |            64400 |
 +--------------------+-----------+---------------+------------------+
   Table 1: Data Rate at the IP Level versus Packetization Interval
              (three retransmissions, 3.33 ms per event)

Schulzrinne & Taylor Standards Track [Page 22] RFC 4733 Telephony Events and Tones December 2006

 As can be seen, a doubling of the interval (from 25 to 50 ms) drops
 aggregate bit rate by about 20% while increasing end-to-end delay by
 25 ms and causing a one-time gap of the same amount.  (Extending the
 playout of a specific V.21 tone event is out of the question, so the
 first algorithm of Section 2.5.2.2 must be used in this application.)
 The reduction in number of packets per second with longer
 packetization periods is countered by the increase in packet size due
 to the increase in number of events per packet.
 For events of longer duration, the reduction in bandwidth is more
 proportional to the increase in packetization interval.  The loss of
 final event reports may also be less critical, so that lower
 redundancy levels are acceptable.  Table 2 shows similar data to
 Table 1, but assuming 70-ms events separated by 50 ms of silence (as
 in an idealized DTMF-based text messaging session) with only the
 basic two retransmissions for event completions.
 +--------------------+-----------+---------------+------------------+
 | Packetization      | Packets/s |     IP Packet |     Total IP Bit |
 | Interval (ms)      |           |   Size (bits) |    Rate (bits/s) |
 +--------------------+-----------+---------------+------------------+
 | 50                 |        20 |       448/520 |            10040 |
 | 33.3               |        30 |       448/520 |            14280 |
 | 25                 |        40 |       448/520 |            18520 |
 | 20                 |        50 |           448 |            22400 |
 +--------------------+-----------+---------------+------------------+
   Table 2: Data Rate at the IP Level versus Packetization Interval
     (two retransmissions, 70 ms per event, 50 ms between events)
 In the third column of the table, the packet size is 448 bits when
 only one event is being reported and 520 bits when the previous event
 is also included.  No more than one level of redundancy is needed up
 to a packetization interval of 50 ms, although at that point most
 packets are reporting two events.  Longer intervals require a second
 level of redundancy in at least some packets.

3. Specification of Event Codes for DTMF Events

 This document defines one class of named events: DTMF tones.

3.1. DTMF Applications

 DTMF signalling [10] is typically generated by a telephone set or
 possibly by a PBX (Private branch telephone exchange).  DTMF digits
 may be consumed by entities such as gateways or application servers
 in the IP network, or by entities such as telephone switches or IVRs
 in the circuit switched network.

Schulzrinne & Taylor Standards Track [Page 23] RFC 4733 Telephony Events and Tones December 2006

 The DTMF events support two possible applications at the sending end:
 1.  The Internet telephony gateway detects DTMF on the incoming
     circuits and sends the RTP payload described here instead of
     regular audio packets.  The gateway likely has the necessary
     digital signal processors and algorithms, as it often needs to
     detect DTMF, e.g., for two-stage dialing.  Having the gateway
     detect tones relieves the receiving Internet end system from
     having to do this work and also avoids having low bit-rate codecs
     like G.723.1 [20] render DTMF tones unintelligible.
 2.  An Internet end system such as an "Internet phone" can emulate
     DTMF functionality without concerning itself with generating
     precise tone pairs and without imposing the burden of tone
     recognition on the receiver.
 A similar distinction occurs at the receiving end.
 1.  In the gateway scenario, an Internet telephony gateway connecting
     a packet voice network to the PSTN re-creates the DTMF tones or
     other telephony events and injects them into the PSTN.
 2.  In the end system scenario, the DTMF events are consumed by the
     receiving entity itself.
 In the most common application, DTMF tones are sent in one direction
 only, typically from the calling end.  The consuming device is most
 commonly an IVR.  DTMF may alternate with voice from either end.  In
 most cases, the only constraint on tone duration is that it exceed a
 minimum value.  However, in some cases a long-duration tone (in
 excess of 1-2 seconds) has special significance.
    ITU-T Recommendation Q.24 [11], Table A-1, indicates that the
    legacy switching equipment in the countries surveyed expects a
    minimum recognizable signal duration of 40 ms, a minimum pause
    between signals of 40 ms, and a maximum signalling rate of 8 to 10
    digits per second depending on the country.  Human-generated DTMF
    signals, of course, are generally longer with larger pauses
    between them.
 DTMF tones may also be used for text telephony.  This application is
 documented in ITU-T Recommendation V.18 [27] Annex B.  In this case,
 DTMF is sent alternately from either end (half-duplex mode), with a
 minimum 300-ms turn-around time.  The only constraints on tone
 durations in this application are that they and the pauses between
 them must exceed specified minimum values.  It is RECOMMENDED that a
 gateway at the sending end be capable of detecting DTMF signals as
 specified by V.18 Annex B (tones and pauses >=40 ms), but should send

Schulzrinne & Taylor Standards Track [Page 24] RFC 4733 Telephony Events and Tones December 2006

 event durations corresponding to those of a V.18 DTMF sender (tones
 >=70 ms, pauses >=50 ms).  This may occasionally imply some degree of
 buffering of outgoing events, but if the source terminal conforms to
 V.18 Annex B, this should not get out of hand.
 Since minor increases in tone duration are harmless for all
 applications of DTMF, but unintended breaks in playout of a DTMF
 digit can confuse the receiving endpoint by creating the appearance
 of extra digits, receiving applications that are converting DTMF
 events back to tones SHOULD use the second playout algorithm rather
 than the first one in Section 2.5.2.2.  This provides some robustness
 against packet loss or congestion.

3.2. DTMF Events

 Table 3 shows the DTMF-related event codes within the telephone-event
 payload format.  The DTMF digits 0-9 and * and # are commonly
 supported.  DTMF digits A through D are less frequently encountered,
 typically in special applications such as military networks.
                  +-------+--------+------+---------+
                  | Event | Code   | Type | Volume? |
                  +-------+--------+------+---------+
                  | 0--9  | 0--9   | tone | yes     |
                  | *     | 10     | tone | yes     |
                  | #     | 11     | tone | yes     |
                  | A--D  | 12--15 | tone | yes     |
                  +-------+--------+------+---------+
                      Table 3: DTMF Named Events

3.3. Congestion Considerations

 The key considerations for the delivery of DTMF events are
 reliability and avoidance of unintended breaks within the playout of
 a given tone.  End-to-end round-trip delay is not a major
 consideration except in the special case where DTMF tones are being
 used for text telephony.  Assuming that, as recommended in
 Section 3.1 above, the second playout algorithm of Section 2.5.2.2 is
 in use, a temporary increase in packetization interval to as much as
 100 ms or double the normal interval, whichever is less, should be
 harmless.

Schulzrinne & Taylor Standards Track [Page 25] RFC 4733 Telephony Events and Tones December 2006

4. RTP Payload Format for Telephony Tones

4.1. Introduction

 As an alternative to describing tones and events by name, as
 described in Section 2, it is sometimes preferable to describe them
 by their waveform properties.  In particular, recognition is faster
 than for naming signals since it does not depend on recognizing
 durations or pauses.
 There is no single international standard for telephone tones such as
 dial tone, ringing (ringback), busy, congestion ("fast-busy"),
 special announcement tones, or some of the other special tones, such
 as payphone recognition, call waiting or record tone.  However, ITU-T
 Recommendation E.180 [18] notes that across all countries, these
 tones share a number of characteristics:
 o  Telephony tones consist of either a single tone, the addition of
    two or three tones or the modulation of two tones.  (Almost all
    tones use two frequencies; only the Hungarian "special dial tone"
    has three.)  Tones that are mixed have the same amplitude and do
    not decay.
 o  In-band tones for telephony events are in the range of 25 Hz
    (ringing tone in Angola) to 2600 Hz (the tone used for line
    signalling in SS No. 5 and R1).  The in-band telephone frequency
    range is limited to 3400 Hz.  R2 defines a 3825 Hz out-of-band
    tone for line signalling on analogue trunks.  (The piano has a
    range from 27.5 to 4186 Hz.)
 o  Modulation frequencies range between 15 (ANSam tone) to 480 Hz
    (Jamaica).  Non-integer frequencies are used only for frequencies
    of 16 2/3 and 33 1/3 Hz.
 o  Tones that are not continuous have durations of less than four
    seconds.
 o  ITU Recommendation E.180 [18] notes that different telephone
    companies require a tone accuracy of between 0.5 and 1.5%.  The
    Recommendation suggests a frequency tolerance of 1%.

Schulzrinne & Taylor Standards Track [Page 26] RFC 4733 Telephony Events and Tones December 2006

4.2. Examples of Common Telephone Tone Signals

 As an aid to the implementor, Table 4 summarizes some common tones.
 The rows labeled "ITU ..." refer to ITU-T Recommendation E.180 [18].
 In these rows, the on and off durations are suggested ranges within
 which local standards would set specific values.  The symbol "+" in
 the table indicates addition of the tones, without modulation, while
 "*" indicates amplitude modulation.
 +-------------------------+-------------------+----------+----------+
 | Tone Name               | Frequency         | On Time  | Off Time |
 |                         |                   | (s)      | (s)      |
 +-------------------------+-------------------+----------+----------+
 | CNG                     | 1100              | 0.5      | 3.0      |
 | V.25 CT                 | 1300              | 0.5      | 2.0      |
 | CED                     | 2100              | 3.3      | --       |
 | ANS                     | 2100              | 3.3      | --       |
 | ANSam                   | 2100*15           | 3.3      | --       |
 | V.21 bit                | 980 or 1180 or    | 0.00333  | --       |
 |                         | 1650 or 1850      |          |          |
 | -------------           | ----------        | -------- | -------- |
 | ITU dial tone           | 425               | --       | --       |
 | U.S. dial tone          | 350+440           | --       | --       |
 | ITU ringing tone        | 425               | 0.67-1.5 | 3-5      |
 | U.S. ringing tone       | 440+480           | 2.0      | 4.0      |
 | ITU busy tone           | 425               | 0.1-0.6  | 0.1-0.7  |
 | U.S. busy tone          | 480+620           | 0.5      | 0.5      |
 | ITU congestion tone     | 425               | 0.1-0.6  | 0.1-0.7  |
 | U.S. congestion tone    | 480+620           | 0.25     | 0.25     |
 +-------------------------+-------------------+----------+----------+
                 Table 4: Examples of Telephony Tones

4.3. Use of RTP Header Fields

4.3.1. Timestamp

 The RTP timestamp reflects the measurement point for the current
 packet.  The event duration described in Section 4.3.3 begins at that
 time.

4.3.2. Marker Bit

 The tone payload type uses the marker bit to distinguish the first
 RTP packet reporting a given instance of a tone from succeeding
 packets for that tone.  The marker bit SHOULD be set to 1 for the
 first packet, and to 0 for all succeeding packets relating to the
 same tone.

Schulzrinne & Taylor Standards Track [Page 27] RFC 4733 Telephony Events and Tones December 2006

4.3.3. Payload Format

 Based on the characteristics described above, this document defines
 an RTP payload format called "tone" that can represent tones
 consisting of one or more frequencies.  (The corresponding media type
 is "audio/tone".)  The default timestamp rate is 8000 Hz, but other
 rates may be defined.  Note that the timestamp rate does not affect
 the interpretation of the frequency, just the durations.
 In accordance with current practice, this payload format does not
 have a static payload type number, but uses an RTP payload type
 number established dynamically and out-of-band.
 The payload format is shown in Figure 2.
      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |    modulation   |T|  volume   |          duration             |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |R R R R|       frequency       |R R R R|       frequency       |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |R R R R|       frequency       |R R R R|       frequency       |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
         ......
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |R R R R|       frequency       |R R R R|      frequency        |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
                  Figure 2: Payload Format for Tones
 The payload contains the following fields:
 modulation:
    The modulation frequency, in Hz.  The field is a 9-bit unsigned
    integer, allowing modulation frequencies up to 511 Hz.  If there
    is no modulation, this field has a value of zero.  Note that the
    amplitude of modulation is not indicated in the payload and must
    be determined by out-of-band means.
 T:
    If the T bit is set (one), the modulation frequency is to be
    divided by three.  Otherwise, the modulation frequency is taken as
    is.
    This bit allows frequencies accurate to 1/3 Hz, since modulation
    frequencies such as 16 2/3 Hz are in practical use.

Schulzrinne & Taylor Standards Track [Page 28] RFC 4733 Telephony Events and Tones December 2006

 volume:
    The power level of the tone, expressed in dBm0 after dropping the
    sign, with range from 0 to -63 dBm0.  (Note: A preferred level
    range for digital tone generators is -8 dBm0 to -3 dBm0.)
 duration:
    The duration of the tone, measured in timestamp units and
    presented in network byte order.  The tone begins at the instant
    identified by the RTP timestamp and lasts for the duration value.
    The value of zero is not permitted, and tones with such a duration
    SHOULD be ignored.
    The definition of duration corresponds to that for sample-based
    codecs, where the timestamp represents the sampling point for the
    first sample.
 frequency:
    The frequencies of the tones to be added, measured in Hz and
    represented as a 12-bit unsigned integer.  The field size is
    sufficient to represent frequencies up to 4095 Hz, which exceeds
    the range of telephone systems.  A value of zero indicates
    silence.  A single tone can contain any number of frequencies.  If
    no frequencies are specified, the packet reports a period of
    silence.
 R:
    This field is reserved for future use.  The sender MUST set it to
    zero, and the receiver MUST ignore it.

4.3.4. Optional Media Type Parameters

 The "rate" parameter describes the sampling rate, in Hertz.  The
 number is written as an integer.  If omitted, the default value is
 8000 Hz.

4.4. Procedures

 This section defines the procedures associated with the tone payload
 type.

4.4.1. Sending Procedures

 The sender MAY send an initial tones packet as soon as a tone is
 recognized, or MAY wait until a pre-negotiated packetization period
 has elapsed.  The first RTP packet for a tone SHOULD have the marker
 bit set to 1.

Schulzrinne & Taylor Standards Track [Page 29] RFC 4733 Telephony Events and Tones December 2006

 In the case of longer-duration tones, the sender SHOULD generate
 multiple RTP packets for the same tone instance.  The RTP timestamp
 MUST be updated for each packet generated (in contrast, for instance,
 to the timestamp for packets carrying telephone events).  Subsequent
 packets for the same tone SHOULD have the marker bit set to 0, and
 the RTP timestamp in each subsequent packet MUST equal the sum of the
 timestamp and the duration in the preceding packet.
 A final RTP packet MAY be generated as soon as the end of the tone is
 detected, without waiting for the latest packetization period to
 elapse.
 The telephone-event payload described in Section 2 is inherently
 redundant, in that later packets for the same event carry all of the
 earlier history of the event except for variations in volume.  In
 contrast, each packet for the tone payload type stands alone; a lost
 packet means a gap in the information available at the receiving end.
 Thus, for increased reliability, the sender SHOULD combine new and
 old tone reports in the same RTP packet using RFC 2198 [2] audio
 redundancy.

4.4.2. Receiving Procedures

 Receiving implementations play out the tones as received, typically
 with a playout delay to allow for lost packets.  When playing out
 successive tone reports for the same tone (marker bit is zero, the
 RTP timestamp is contiguous with that of the previous RTP packet, and
 payload content is identical), the receiving implementation SHOULD
 continue the tone without change or a break.

4.4.3. Handling of Congestion

 If the sender determines that packets are being lost due to
 congestion (e.g., through RTCP receiver reports), it SHOULD increase
 the packetization interval for initial and interim tone reports so as
 to reduce traffic volume to the receiver.  The degree to which this
 is possible without causing damaging consequences at the receiving
 end depends both upon the playout delay used at that end and upon the
 specific application associated with the tones.  Both the maximum
 packetization interval and maximum increase in packetization interval
 at any one time are therefore a matter of configuration or out-of-
 band negotiation.

Schulzrinne & Taylor Standards Track [Page 30] RFC 4733 Telephony Events and Tones December 2006

5. Examples

 Consider a DTMF dialling sequence, where the user dials the digits
 "911" and a sending gateway detects them.  The first digit is 200 ms
 long (1600 timestamp units) and starts at time 0; the second digit
 lasts 250 ms (2000 timestamp units) and starts at time 880 ms (7040
 timestamp units); the third digit is pressed at time 1.4 s (11,200
 timestamp units) and lasts 220 ms (1760 timestamp units).  The frame
 duration is 50 ms.
 Table 5 shows the complete sequence of events assuming that only the
 telephone-event payload type is being reported.  For simplicity: the
 timestamp is assumed to begin at 0, the RTP sequence number at 1, and
 volume settings are omitted.
 +-------+-----------+------+--------+------+--------+--------+------+
 |  Time | Event     |   M  |  Time- |  Seq |  Event |  Dura- |   E  |
 |  (ms) |           |  bit |  stamp |   No |  Code  |   tion |  bit |
 +-------+-----------+------+--------+------+--------+--------+------+
 |     0 | "9"       |      |        |      |        |        |      |
 |       | starts    |      |        |      |        |        |      |
 |    50 | RTP       |  "1" |      0 |    1 |    9   |    400 |  "0" |
 |       | packet 1  |      |        |      |        |        |      |
 |       | sent      |      |        |      |        |        |      |
 |   100 | RTP       |  "0" |      0 |    2 |    9   |    800 |  "0" |
 |       | packet 2  |      |        |      |        |        |      |
 |       | sent      |      |        |      |        |        |      |
 |   150 | RTP       |  "0" |      0 |    3 |    9   |   1200 |  "0" |
 |       | packet 3  |      |        |      |        |        |      |
 |       | sent      |      |        |      |        |        |      |
 |   200 | RTP       |  "0" |      0 |    4 |    9   |   1600 |  "0" |
 |       | packet 4  |      |        |      |        |        |      |
 |       | sent      |      |        |      |        |        |      |
 |   200 | "9" ends  |      |        |      |        |        |      |
 |   250 | RTP       |  "0" |      0 |    5 |    9   |   1600 |  "1" |
 |       | packet 4  |      |        |      |        |        |      |
 |       | first     |      |        |      |        |        |      |
 |       | retrans-  |      |        |      |        |        |      |
 |       | mission   |      |        |      |        |        |      |
 |   300 | RTP       |  "0" |      0 |    6 |    9   |   1600 |  "1" |
 |       | packet 4  |      |        |      |        |        |      |
 |       | second    |      |        |      |        |        |      |
 |       | retrans-  |      |        |      |        |        |      |
 |       | mission   |      |        |      |        |        |      |
 |   880 | First "1" |      |        |      |        |        |      |
 |       | starts    |      |        |      |        |        |      |

Schulzrinne & Taylor Standards Track [Page 31] RFC 4733 Telephony Events and Tones December 2006

 |   930 | RTP       |  "1" |   7040 |    7 |    1   |    400 |  "0" |
 |       | packet 5  |      |        |      |        |        |      |
 |       | sent      |      |        |      |        |        |      |
 |   ... | ...       |  ... |    ... |  ... |   ...  |    ... |  ... |
 |  1130 | RTP       |  "0" |   7040 |   11 |    1   |   2000 |  "0" |
 |       | packet 9  |      |        |      |        |        |      |
 |       | sent      |      |        |      |        |        |      |
 |  1130 | First "1" |      |        |      |        |        |      |
 |       | ends      |      |        |      |        |        |      |
 |  1180 | RTP       |  "0" |   7040 |   12 |    1   |   2000 |  "1" |
 |       | packet 9  |      |        |      |        |        |      |
 |       | first     |      |        |      |        |        |      |
 |       | retrans-  |      |        |      |        |        |      |
 |       | mission   |      |        |      |        |        |      |
 |  1230 | RTP       |  "0" |   7040 |   13 |    1   |   2000 |  "1" |
 |       | packet 9  |      |        |      |        |        |      |
 |       | second    |      |        |      |        |        |      |
 |       | retrans-  |      |        |      |        |        |      |
 |       | mission   |      |        |      |        |        |      |
 |  1400 | Second    |      |        |      |        |        |      |
 |       | "1"       |      |        |      |        |        |      |
 |       | starts    |      |        |      |        |        |      |
 |  1450 | RTP       |  "1" |  11200 |   14 |    1   |    400 |  "0" |
 |       | packet 10 |      |        |      |        |        |      |
 |       | sent      |      |        |      |        |        |      |
 |   ... | ...       |  ... |    ... |  ... |   ...  |    ... |  ... |
 |  1620 | Second    |      |        |      |        |        |      |
 |       | "1" ends  |      |        |      |        |        |      |
 |  1650 | RTP       |  "0" |  11200 |   18 |    1   |   1760 |  "1" |
 |       | packet 14 |      |        |      |        |        |      |
 |       | sent      |      |        |      |        |        |      |
 |  1700 | RTP       |  "0" |  11200 |   19 |    1   |   1760 |  "1" |
 |       | packet 14 |      |        |      |        |        |      |
 |       | first     |      |        |      |        |        |      |
 |       | retrans-  |      |        |      |        |        |      |
 |       | mission   |      |        |      |        |        |      |
 |  1750 | RTP       |  "0" |  11200 |   20 |    1   |   1760 |  "1" |
 |       | packet 14 |      |        |      |        |        |      |
 |       | second    |      |        |      |        |        |      |
 |       | retrans-  |      |        |      |        |        |      |
 |       | mission   |      |        |      |        |        |      |
 +-------+-----------+------+--------+------+--------+--------+------+
                  Table 5: Example of Event Reporting

Schulzrinne & Taylor Standards Track [Page 32] RFC 4733 Telephony Events and Tones December 2006

 Table 6 shows the same sequence assuming that only the tone payload
 type is being reported.  This looks somewhat different.  For
 simplicity: the timestamp is assumed to begin at 0, the sequence
 number at 1.  Volume, the T bit, and the modulation frequency are
 omitted.  The latter two are always 0.
 +-------+-----------+-----+--------+------+--------+-------+--------+
 |  Time | Event     |  M  |  Time- |  Seq | Dura-  | Freq 1| Freq 2 |
 |  (ms) |           | bit |  stamp |   No | tion   | (Hz)  | (Hz)   |
 +-------+-----------+-----+--------+------+--------+-------+--------+
 |     0 | "9"       |     |        |      |        |       |        |
 |       | starts    |     |        |      |        |       |        |
 |    50 | RTP       | "1" |      0 |    1 | 400    | 852   | 1477   |
 |       | packet 1  |     |        |      |        |       |        |
 |       | sent      |     |        |      |        |       |        |
 |   100 | RTP       | "0" |    400 |    2 | 400    | 852   | 1477   |
 |       | packet 2  |     |        |      |        |       |        |
 |       | sent      |     |        |      |        |       |        |
 |   ... | ...       | ... |    ... |  ... | ...    | ...   | ...    |
 |   200 | RTP       | "0" |   1200 |    4 | 400    | 852   | 1477   |
 |       | packet 4  |     |        |      |        |       |        |
 |       | sent      |     |        |      |        |       |        |
 |   200 | "9" ends  |     |        |      |        |       |        |
 |   880 | First "1" |     |        |      |        |       |        |
 |       | starts    |     |        |      |        |       |        |
 |   930 | RTP       | "1" |   7040 |    5 | 400    | 697   | 1209   |
 |       | packet 5  |     |        |      |        |       |        |
 |       | sent      |     |        |      |        |       |        |
 |   980 | RTP       | "0" |   7440 |    6 | 400    | 697   | 1209   |
 |       | packet 6  |     |        |      |        |       |        |
 |       | sent      |     |        |      |        |       |        |
 |   ... | ...       | ... |    ... |  ... | ...    | ...   | ...    |
 |  1130 | First "1" |     |        |      |        |       |        |
 |       | ends      |     |        |      |        |       |        |
 |  1400 | Second    |     |        |      |        |       |        |
 |       | "1"       |     |        |      |        |       |        |
 |       | starts    |     |        |      |        |       |        |
 |  1450 | RTP       | "1" |  11200 |   10 | 400    | 697   | 1209   |
 |       | packet 10 |     |        |      |        |       |        |
 |       | sent      |     |        |      |        |       |        |
 |   ... | ...       | ... |    ... |  ... | ...    | ...   | ...    |
 |  1620 | Second    |     |        |      |        |       |        |
 |       | "1" ends  |     |        |      |        |       |        |
 |  1650 | RTP       | "0" |  12800 |   14 | 160    | 697   | 1209   |
 |       | packet 14 |     |        |      |        |       |        |
 |       | sent      |     |        |      |        |       |        |
 +-------+-----------+-----+--------+------+--------+-------+--------+
                  Table 6: Example of Tone Reporting

Schulzrinne & Taylor Standards Track [Page 33] RFC 4733 Telephony Events and Tones December 2006

 Now consider a combined payload, where the tone payload is the
 primary payload type and the event payload is treated as a redundant
 encoding (one level of redundancy).  Because the primary payload is
 tones, the tone payload rules determine the setting of the RTP header
 fields.  This means that the RTP timestamp always advances.  As a
 corollary, the timestamp offset for the events payload in the RFC
 2198 header increases by the same amount.
 One issue that has to be considered in a combined payload is how to
 handle retransmissions of final event reports.  The tone payload
 specification does not recommend retransmissions of final packets, so
 it is unclear what to put in the primary payload fields of the
 combined packet.  In the interests of simplicity, it is suggested
 that the retransmitted packets copy the fields relating to the
 primary payload (including the RTP timestamp) from the original
 packet.  The same principle can be applied if the packet includes
 multiple levels of event payload redundancy.
 The figures below all illustrate "RTP packet 14" in the above tables.
 Figure 3 shows an event-only payload, corresponding to Table 5.
 Figure 4 shows a tone-only payload, corresponding to Table 6.
 Finally, Figure 5 shows a combined payload, with tones primary and
 events as a single redundant layer.  Note that the combined payload
 has the RTP sequence numbers shown in Table 5, because the
 transmitted sequence includes the retransmitted packets.
 Figure 3 assumes that the following SDP specification was used.  This
 session description provides for separate streams of G.729 [21] audio
 and events.  Packets reported within the G.729 stream are not
 considered here.
    m=audio 12344 RTP/AVP 99
    a=rtpmap:99 G729/8000
    a=ptime:20
    m=audio 12346 RTP/AVP 100
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-15
    a=ptime:50

Schulzrinne & Taylor Standards Track [Page 34] RFC 4733 Telephony Events and Tones December 2006

     0                   1                   2                   3
     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |V=2|P|X|  CC   |M|     PT      |       sequence number         |
    | 2 |0|0|   0   |0|    100      |            18                 |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |                           timestamp                           |
    |                             11200                             |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |           synchronization source (SSRC) identifier            |
    |                            0x5234a8                           |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |     event     |E R| volume    |          duration             |
    |       1       |1 0|    20     |             1760              |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
            Figure 3: Example RTP Packet for Event Payload
 Figure 4 assumes that an SDP specification similar to that of the
 previous case was used.
    m=audio 12344 RTP/AVP 99
    a=rtpmap:99 G729/8000
    a=ptime:20
    m=audio 12346 RTP/AVP 101
    a=rtpmap:101 tone/8000
    a=ptime:50
     0                   1                   2                   3
     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |V=2|P|X|  CC   |M|     PT      |       sequence number         |
    | 2 |0|0|   0   |0|    101      |             14                |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |                           timestamp                           |
    |                             12800                             |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |           synchronization source (SSRC) identifier            |
    |                            0x5234a8                           |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |    modulation   |T|  volume   |          duration             |
    |        0        |0|    20     |             160               |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |R R R R|       frequency       |R R R R|       frequency       |
    |0 0 0 0|          697          |0 0 0 0|         1209          |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
             Figure 4: Example RTP Packet for Tone Payload

Schulzrinne & Taylor Standards Track [Page 35] RFC 4733 Telephony Events and Tones December 2006

 Figure 5, for the combined payload, assumes the following SDP session
 description:
    m=audio 12344 RTP/AVP 99
    a=rtpmap:99 G729/8000
    a=ptime:20
    m=audio 12346 RTP/AVP 102 101 100
    a=rtpmap:102 red/8000/1
    a=fmtp:102 101/100
    a=rtpmap:101 tone/8000
    a=rtpmap:100 telephone-event/8000
    a=fmtp:100 0-15
    a=ptime:50
 For ease of presentation, Figure 5 presents the actual payloads as if
 they began on 32-bit boundaries.  In the actual packet, they follow
 immediately after the end of the RFC 2198 header, and thus are
 displaced one octet into successive words.

Schulzrinne & Taylor Standards Track [Page 36] RFC 4733 Telephony Events and Tones December 2006

     0                   1                   2                   3
     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |V=2|P|X|  CC   |M|     PT      |       sequence number         |
    | 2 |0|0|   0   |0|    102      |             18                |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |                           timestamp                           |
    |                             12800                             |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |           synchronization source (SSRC) identifier            |
    |                            0x5234a8                           |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |F|   block PT  |  timestamp offset         |   block length    |
    |1|      100    |       1600                |        4          |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |F|   block PT  |   event payload begins ...                    /
    |0|      101    |                                               \
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
        Event payload
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |     event     |E R| volume    |          duration             |
    |       1       |1 0|    20     |             1760              |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
        Tone payload
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |    modulation   |T|  volume   |          duration             |
    |        0        |0|    20     |             160               |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
    |R R R R|       frequency       |R R R R|       frequency       |
    |0 0 0 0|          697          |0 0 0 0|         1209          |
    +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   Figure 5: Example RTP Packet for Combined Tone and Event Payloads

Schulzrinne & Taylor Standards Track [Page 37] RFC 4733 Telephony Events and Tones December 2006

6. Security Considerations

 RTP packets using the payload formats defined in this specification
 are subject to the security considerations discussed in the RTP
 specification (RFC 3550 [5]), and any appropriate RTP profile (for
 example, RFC 3551 [13]).  The RFC 3550 discussion focuses on
 requirements for confidentiality.  Additional security considerations
 relating to implementation are described in RFC 2198 [2].
 The telephone-event payload defined in this specification is highly
 compressed.  A change in value of just one bit can result in a major
 change in meaning as decoded at the receiver.  Thus, message
 integrity MUST be provided for the telephone-event payload type.
 To meet the need for protection both of confidentiality and
 integrity, compliant implementations SHOULD implement the Secure
 Real-time Transport Protocol (SRTP) [7].
    Note that the appropriate method of key distribution for SRTP may
    vary with the specific application.
    In some deployments, it may be preferable to use other means to
    provide protection equivalent to that provided by SRTP.
 Provided that gateway design includes robust, low-overhead tone
 generation, this payload type does not exhibit any significant non-
 uniformity in the receiver side computational complexity for packet
 processing to cause a potential denial-of-service threat.

7. IANA Considerations

 This document updates the descriptions of two RTP payload formats,
 'telephone-event' and 'tone', and associated Internet media types,
 audio/telephone-event and audio/tone.  It also documents the event
 codes for DTMF tone events.
 Within the audio/telephone-event type, events MUST be registered with
 IANA.  Registrations are subject to the policies "Specification
 Required" and "Expert Review" as defined in RFC 2434 [3].  The IETF-
 appointed expert must ensure that:
 a.  the meaning and application of the proposed events are clearly
     documented;
 b.  the events cannot be represented by existing event codes,
     possibly with some minor modification of event definitions;

Schulzrinne & Taylor Standards Track [Page 38] RFC 4733 Telephony Events and Tones December 2006

 c.  the number of events is the minimum necessary to fulfill the
     purpose of their application(s).
 The expert is further responsible for providing guidance on the
 allocation of event codes to the proposed events.  Specifically, the
 expert must indicate whether the event appears to be the same as one
 defined in RFC 2833 but not specified in any new document.  In this
 case, the event code specified in RFC 2833 for that event SHOULD be
 assigned to the proposed event.  Otherwise, event codes MUST be
 assigned from the set of available event codes listed below.  If this
 set is exhausted, the criterion for assignment from the reserved set
 of event codes is to first assign those that appear to have the
 lowest probability of being revived in their RFC 2833 meaning in a
 new specification.
 The documentation for each event MUST indicate whether the event is a
 state, tone, or other type of event (e.g., an out-of-band electrical
 event such as on-hook or an indication that will not itself be played
 out as tones at the receiving end).  For tone events, the
 documentation MUST indicate whether the volume field is applicable or
 must be set to 0.
 In view of the tradeoffs between the different reliability mechanisms
 discussed in Section 2.6, documentation of specific events SHOULD
 include a discussion of the appropriate design decisions for the
 applications of those events.
 Legal event codes range from 0 to 255.  The initial registry content
 is shown in Table 7, and consists of the sixteen events defined in
 Section 3 of this document.  The remaining codes have the following
 disposition:
 o  codes 17-22, 50-51, 90-95, 113-120, 169, and 206-255 are available
    for assignment;
 o  codes 23-40, 49, and 52-63 are reserved for events defined in
    [16];
 o  codes 121-137 and 174-205 are reserved for events defined in [17];
 o  codes 16, 41-48, 64-88, 96-112, 138-168, and 170-173 are reserved
    in the first instance for specifications reviving the
    corresponding RFC 2833 events, and in the second instance for
    general assignment after all other codes have been assigned.

Schulzrinne & Taylor Standards Track [Page 39] RFC 4733 Telephony Events and Tones December 2006

      +------------+--------------------------------+-----------+
      | Event Code | Event Name                     | Reference |
      +------------+--------------------------------+-----------+
      |          0 | DTMF digit "0"                 |  RFC 4733 |
      |          1 | DTMF digit "1"                 |  RFC 4733 |
      |          2 | DTMF digit "2"                 |  RFC 4733 |
      |          3 | DTMF digit "3"                 |  RFC 4733 |
      |          4 | DTMF digit "4"                 |  RFC 4733 |
      |          5 | DTMF digit "5"                 |  RFC 4733 |
      |          6 | DTMF digit "6"                 |  RFC 4733 |
      |          7 | DTMF digit "7"                 |  RFC 4733 |
      |          8 | DTMF digit "8"                 |  RFC 4733 |
      |          9 | DTMF digit "9"                 |  RFC 4733 |
      |         10 | DTMF digit "*"                 |  RFC 4733 |
      |         11 | DTMF digit "#"                 |  RFC 4733 |
      |         12 | DTMF digit "A"                 |  RFC 4733 |
      |         13 | DTMF digit "B"                 |  RFC 4733 |
      |         14 | DTMF digit "C"                 |  RFC 4733 |
      |         15 | DTMF digit "D"                 |  RFC 4733 |
      +------------+--------------------------------+-----------+
          Table 7: audio/telephone-event Event Code Registry

7.1. Media Type Registrations

7.1.1. Registration of Media Type audio/telephone-event

 This registration is done in accordance with [6] and [8].
 Type name: audio
 Subtype name: telephone-event
 Required parameters: none.
 Optional parameters:
    The "events" parameter lists the events supported by the
    implementation.  Events are listed as one or more comma-separated
    elements.  Each element can be either a single integer providing
    the value of an event code or an integer followed by a hyphen and
    a larger integer, presenting a range of consecutive event code
    values.  The list does not have to be sorted.  No white space is
    allowed in the argument.  The union of all of the individual event
    codes and event code ranges designates the complete set of event
    numbers supported by the implementation.  If the "events"
    parameter is omitted, support for events 0-15 (the DTMF tones) is
    assumed.

Schulzrinne & Taylor Standards Track [Page 40] RFC 4733 Telephony Events and Tones December 2006

    The "rate" parameter describes the sampling rate, in Hertz.  The
    number is written as an integer.  If omitted, the default value is
    8000 Hz.
 Encoding considerations:
    In the terminology defined by [8] section 4.8, this type is framed
    and binary.
 Security considerations:
    See Section 6, "Security Considerations", in this document.
 Interoperability considerations: none.
 Published specification: this document.
 Applications which use this media:
    The telephone-event audio subtype supports the transport of events
    occurring in telephone systems over the Internet.
 Additional information:
    Magic number(s): N/A.
    File extension(s): N/A.
    Macintosh file type code(s): N/A.
 Person & email address to contact for further information:
    Tom Taylor, taylor@nortel.com.
    IETF AVT Working Group.
 Intended usage: COMMON.
 Restrictions on usage:
    This type is defined only for transfer via RTP [5].
 Author: IETF Audio/Video Transport Working Group.
 Change controller:
    IETF Audio/Video Transport Working Group as delegated from the
    IESG.

Schulzrinne & Taylor Standards Track [Page 41] RFC 4733 Telephony Events and Tones December 2006

7.1.2. Registration of Media Type audio/tone

 This registration is done in accordance with [6] and [8].
 Type name: audio
 Subtype name: tone
 Required parameters: none
 Optional parameters:
    The "rate" parameter describes the sampling rate, in Hertz.  The
    number is written as an integer.  If omitted, the default value is
    8000 Hz.
 Encoding considerations:
    In the terminology defined by [8] section 4.8, this type is framed
    and binary.
 Security considerations:
    See Section 6, "Security Considerations", in this document.
 Interoperability considerations: none
 Published specification: this document.
 Applications which use this media:
    The tone audio subtype supports the transport of pure composite
    tones, for example, those commonly used in the current telephone
    system to signal call progress.
 Additional information:
    Magic number(s): N/A.
    File extension(s): N/A.
    Macintosh file type code(s): N/A.
 Person & email address to contact for further information:
    Tom Taylor, taylor@nortel.com.
    IETF AVT Working Group.
 Intended usage: COMMON.

Schulzrinne & Taylor Standards Track [Page 42] RFC 4733 Telephony Events and Tones December 2006

 Restrictions on usage:
    This type is defined only for transfer via RTP [5].
 Author: IETF Audio/Video Transport Working Group.
 Change controller:
    IETF Audio/Video Transport Working Group as delegated from the
    IESG.

8. Acknowledgements

 Scott Petrack was the original author of RFC 2833.  Henning
 Schulzrinne later loaned his expertise to complete the document, but
 Scott must be credited with the energy behind the idea of a compact
 encoding of tones over IP.
 In RFC 2833, the suggestions of the Megaco working group were
 acknowledged.  Colin Perkins and Magnus Westerland, Chairs of the AVT
 Working Group, provided helpful advice in the formation of the
 present document.  Over the years, detailed advice and comments for
 RFC 2833, this document, or both were provided by Hisham Abdelhamid,
 Flemming Andreasen, Fred Burg, Steve Casner, Dan Deliberato, Fatih
 Erdin, Bill Foster, Mike Fox, Mehryar Garakani, Gunnar Hellstrom,
 Rajesh Kumar, Terry Lyons, Steve Magnell, Zarko Markov, Tim
 Melanchuk, Kai Miao, Satish Mundra, Kevin Noll, Vern Paxson, Oren
 Peleg, Raghavendra Prabhu, Moshe Samoha, Todd Sherer, Adrian Soncodi,
 Yaakov Stein, Mira Stevanovic, Alex Urquizo, and Herb Wildfeur.

9. References

9.1. Normative References

 [1]   Bradner, S., "Key words for use in RFCs to Indicate Requirement
       Levels", BCP 14, RFC 2119, March 1997.
 [2]   Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley,
       M., Bolot, J., Vega-Garcia, A., and S. Fosse-Parisis, "RTP
       Payload for Redundant Audio Data", RFC 2198, September 1997.
 [3]   Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA
       Considerations Section in RFCs", BCP 26, RFC 2434,
       October 1998.
 [4]   Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
       Session Description Protocol (SDP)", RFC 3264, June 2002.

Schulzrinne & Taylor Standards Track [Page 43] RFC 4733 Telephony Events and Tones December 2006

 [5]   Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
       "RTP: A Transport Protocol for Real-Time Applications", STD 64,
       RFC 3550, July 2003.
 [6]   Casner, S. and P. Hoschka, "MIME Type Registration of RTP
       Payload Formats", RFC 3555, July 2003.
 [7]   Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
       Norrman, "The Secure Real-time Transport Protocol (SRTP)",
       RFC 3711, March 2004.
 [8]   Freed, N. and J. Klensin, "Media Type Specifications and
       Registration Procedures", BCP 13, RFC 4288, December 2005.
 [9]   Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
       Description Protocol", RFC 4566, July 2006.
 [10]  International Telecommunication Union, "Technical features of
       push-button telephone sets", ITU-T Recommendation Q.23,
       November 1988.
 [11]  International Telecommunication Union, "Multifrequency push-
       button signal reception", ITU-T Recommendation Q.24,
       November 1988.

9.2. Informative References

 [12]  Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF Digits,
       Telephony Tones and Telephony Signals", RFC 2833, May 2000.
 [13]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
       Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
 [14]  Kreuter, R., "RTP Payload Format for a 64 kbit/s Transparent
       Call", RFC 4040, April 2005.
 [15]  Hellstrom, G. and P. Jones, "RTP Payload for Text
       Conversation", RFC 4103, June 2005.
 [16]  Schulzrinne, H. and T. Taylor, "Definition of Events for Modem,
       Fax, and Text Telephony Signals", RFC 4734, December 2006.
 [17]  Schulzrinne, H. and T. Taylor, "Definition of Events For
       Channel-Oriented Telephony Signalling", Work In Progress ,
       November 2005.

Schulzrinne & Taylor Standards Track [Page 44] RFC 4733 Telephony Events and Tones December 2006

 [18]  International Telecommunication Union, "Technical
       characteristics of tones for the telephone service", ITU-T
       Recommendation E.180/Q.35, March 1998.
 [19]  International Telecommunication Union, "Pulse code modulation
       (PCM) of voice frequencies", ITU-T Recommendation G.711,
       November 1988.
 [20]  International Telecommunication Union, "Speech coders : Dual
       rate speech coder for multimedia communications transmitting at
       5.3 and 6.3 kbit/s", ITU-T Recommendation G.723.1, March 1996.
 [21]  International Telecommunication Union, "Coding of speech at 8
       kbit/s using conjugate-structure algebraic-code-excited linear-
       prediction (CS-ACELP)", ITU-T Recommendation G.729, March 1996.
 [22]  International Telecommunication Union, "ISDN user-network
       interface layer 3 specification for basic call control", ITU-T
       Recommendation Q.931, May 1998.
 [23]  International Telecommunication Union, "Procedures for real-
       time Group 3 facsimile communication over IP networks", ITU-T
       Recommendation T.38, July 2003.
 [24]  International Telecommunication Union, "Procedures for starting
       sessions of data transmission over the public switched
       telephone network", ITU-T Recommendation V.8, November 2000.
 [25]  International Telecommunication Union, "Modem-over-IP networks:
       Procedures for the end-to-end connection of V-series DCEs",
       ITU-T Recommendation V.150.1, January 2003.
 [26]  International Telecommunication Union, "Procedures for
       supporting Voice-Band Data over IP Networks", ITU-T
       Recommendation V.152, January 2005.
 [27]  International Telecommunication Union, "Operational and
       interworking requirements for {DCEs operating in the text
       telephone mode", ITU-T Recommendation V.18, November 2000.
       See also Recommendation V.18 Amendment 1, Nov. 2002.
 [28]  VOIP Troubleshooter LLC, "Indepth: Packet Loss Burstiness",
       2005,
       <http://www.voiptroubleshooter.com/indepth/burstloss.html>.

Schulzrinne & Taylor Standards Track [Page 45] RFC 4733 Telephony Events and Tones December 2006

Appendix A. Summary of Changes from RFC 2833

 The memo has been significantly restructured, incorporating a large
 number of clarifications to the specification.  With the exception of
 those items noted below, the changes to the memo are intended to be
 backwards-compatible clarifications.  However, due to inconsistencies
 and unclear definitions in RFC 2833 [12] it is likely that some
 implementations interpreted that memo in ways that differ from this
 version.
 RFC 2833 required that all implementations be capable of receiving
 the DTMF events (event codes 0-15).  Section 2.5.1.1 of the present
 document requires that a sender transmit only the events that the
 receiver is capable of receiving.  In the absence of a knowledge of
 receiver capabilities, the sender SHOULD assume support of the DTMF
 events but of no other events.  The sender SHOULD indicate what
 events it can send.  Section 2.5.2.1 requires that a receiver
 signalling its capabilities using SDP MUST indicate which events it
 can receive.
 Non-zero values in the volume field of the payload were applicable
 only to DTMF tones in RFC 2833, and for other events the receiver was
 required to ignore them.  The present memo requires that the
 definition of each event indicate whether the volume field is
 applicable to that event.  The last paragraph of Section 2.5.2.2
 indicates what a receiver may do if it receives volumes with zero
 values for events to which the volume field is applicable.  Along
 with the RFC 2833 receiver rule, this ensures backward compatibility
 in both directions of transmission.
 Section 2.5.1.3 and Section 2.5.2.3 introduce a new procedure for
 reporting and playing out events whose duration exceeds the capacity
 of the payload duration field.  This procedure may cause momentary
 confusion at an old (RFC 2833) receiver, because the timestamp is
 updated without setting the E bit of the preceding event report and
 without setting the M bit of the new one.
 Section 2.5.1.5 and Section 2.5.2.4 introduce a new procedure whereby
 a sequence of short-duration events may be packed into a single event
 report.  If an old (RFC 2833) receiver receives such a report, it may
 discard the packet as invalid, since the packet holds more content
 than the receiver was expecting.  In any event, the additional events
 in the packet will be lost.

Schulzrinne & Taylor Standards Track [Page 46] RFC 4733 Telephony Events and Tones December 2006

 Section 2.3.5 introduces the possibility of "state" events and
 defines procedures for setting the duration field for reports of such
 events.  Section 2.5.1.2 defines special exemptions from the setting
 of the E bit for state events.  Three more sections mention
 procedures related to these events.
 The Security Considerations section is updated to mention the
 requirement for protection of integrity.  More importantly, it makes
 implementation of SRTP [7] mandatory for compliant implementations,
 without specifying a mandatory-to-implement method of key
 distribution.
 Finally, this document establishes an IANA registry for event codes
 and establishes criteria for their documentation.  This document
 provides an initial population for the new registry, consisting
 solely of the sixteen DTMF events.  Two companion documents [16] and
 [17] describe events related to modems, fax, and text telephony and
 to channel-associated telephony signalling, respectively.  Some
 changes were made to the latter because of errors and redundancies in
 the RFC 2833 assignments.  The remaining events defined in RFC 2833
 are deprecated because they do not appear to have been implemented,
 but their codes have been conditionally reserved in case any of them
 is needed in the future.  Table 8 indicates the disposition of the
 event codes in detail.  Event codes not mentioned in this table were
 not allocated by RFC 2833 and continue to be unused.
 +-------------+---------------------------------------+-------------+
 | Event Codes | RFC 2833 Description                  | Disposition |
 +-------------+---------------------------------------+-------------+
 |        0-15 | DTMF digits                           | RFC 4733    |
 |          16 | Line flash (deprecated)               | Reserved    |
 |       23-31 | Unused                                | [16]        |
 |       32-40 | Data and fax                          | [16]        |
 |       41-48 | Data and fax (V.8bis, deprecated)     | Reserved    |
 |       52-63 | Unused                                | [16]        |
 |       64-89 | E.182 line events (deprecated)        | Reserved    |
 |      96-112 | Country-specific line events          | Reserved    |
 |             | (deprecated)                          |             |
 |     121-127 | Unused                                | [17]        |
 |     128-137 | Trunks: MF 0-9                        | [17]        |
 |     138-143 | Trunks: other MF (deprecated)         | Reserved    |
 |     144-159 | Trunks: ABCD signalling               | [17]        |
 |     160-168 | Trunks: various (deprecated)          | Reserved    |
 |     170-173 | Trunks: various (deprecated)          | Reserved    |
 |     174-205 | Unused                                | [17]        |
 +-------------+---------------------------------------+-------------+
         Table 8: Disposition of RFC 2833-defined Event Codes

Schulzrinne & Taylor Standards Track [Page 47] RFC 4733 Telephony Events and Tones December 2006

Authors' Addresses

 Henning Schulzrinne
 Columbia U.
 Dept. of Computer Science
 Columbia University
 1214 Amsterdam Avenue
 New York, NY  10027
 US
 EMail: schulzrinne@cs.columbia.edu
 Tom Taylor
 Nortel
 1852 Lorraine Ave
 Ottawa, Ontario  K1H 6Z8
 Canada
 EMail: taylor@nortel.com

Schulzrinne & Taylor Standards Track [Page 48] RFC 4733 Telephony Events and Tones December 2006

Full Copyright Statement

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 contained in BCP 78, and except as set forth therein, the authors
 retain all their rights.
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Acknowledgement

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Schulzrinne & Taylor Standards Track [Page 49]

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