<pre>Network Working Group                                     H. Schulzrinne
Request for Comments: 4733                                   Columbia U.
Obsoletes: <a href="./rfc2833">2833</a>                                                T. Taylor
Category: Standards Track                                         Nortel
                                                           December 2006


  <span class="h1">RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals</span>

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 <a href="./rfc2833">RFC 2833</a>.

   This memo captures and expands upon the basic framework defined in
   <a href="./rfc2833">RFC 2833</a>, 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 <a href="./rfc2833">RFC 2833</a> 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 <a href="./rfc2833">RFC 2833</a> 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 <a href="./rfc2833">RFC 2833</a>
   framework: subdivision of long events into segments, reporting of
   multiple events in a single packet, and the concept and reporting of
   state events.







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Table of Contents

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






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   <a href="#section-7">7</a>. IANA Considerations ............................................<a href="#page-38">38</a>
      <a href="#section-7.1">7.1</a>. Media Type Registrations ..................................<a href="#page-40">40</a>
           <a href="#section-7.1.1">7.1.1</a>. Registration of Media Type audio/telephone-event ...<a href="#page-40">40</a>
           <a href="#section-7.1.2">7.1.2</a>. Registration of Media Type audio/tone ..............<a href="#page-42">42</a>
   <a href="#section-8">8</a>. Acknowledgements ...............................................<a href="#page-43">43</a>
   <a href="#section-9">9</a>. References .....................................................<a href="#page-43">43</a>
      <a href="#section-9.1">9.1</a>. Normative References ......................................<a href="#page-43">43</a>
      <a href="#section-9.2">9.2</a>. Informative References ....................................<a href="#page-44">44</a>
   <a href="#appendix-A">Appendix A</a>. Summary of Changes from <a href="./rfc2833">RFC 2833</a> ......................<a href="#page-46">46</a>










































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<span class="h2"><a class="selflink" id="section-1" href="#section-1">1</a>.  Introduction</span>

<span class="h3"><a class="selflink" id="section-1.1" href="#section-1.1">1.1</a>.  Terminology</span>

   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 <a href="./rfc2119">RFC 2119</a> [<a href="#ref-1" title="&quot;Key words for use in RFCs to Indicate Requirement Levels&quot;">1</a>].

   This document uses the following abbreviations:

   ANSam   Answer tone (amplitude modulated) [<a href="#ref-24" title="&quot;Procedures for starting sessions of data transmission over the public switched telephone network&quot;">24</a>]

   DTMF    Dual-Tone Multifrequency [<a href="#ref-10" title="&quot;Technical features of push-button telephone sets&quot;">10</a>]

   IVR     Interactive Voice Response unit

   PBX     Private branch exchange (telephone system)

   PSTN    Public Switched (circuit) Telephone Network

   RTP     Real-time Transport Protocol [<a href="#ref-5" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">5</a>]

   SDP     Session Description Protocol [<a href="#ref-9" title="&quot;SDP: Session Description Protocol&quot;">9</a>]

<span class="h3"><a class="selflink" id="section-1.2" href="#section-1.2">1.2</a>.  Overview</span>

   This memo defines two RTP [<a href="#ref-5" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">5</a>] payload formats, one for carrying
   dual-tone multifrequency (DTMF) digits and other line and trunk
   signals as events (<a href="#section-2">Section 2</a>), and a second one to describe general
   multifrequency tones in terms only of their frequency and cadence
   (<a href="#section-4">Section 4</a>).  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.  <a href="#section-2">Section 2</a> defines the
   payload format and associated procedures for use of named events.
   <a href="#section-3">Section 3</a> describes the events for which event codes are defined in
   this document.  <a href="#section-4">Section 4</a> describes the payload format and associated
   procedures for tone representations.  <a href="#section-5">Section 5</a> provides some
   examples of encoded events, tones, and combined payloads.  <a href="#section-6">Section 6</a>
   deals with security considerations.  <a href="#section-7">Section 7</a> defines the IANA
   requirements for registration of event codes for named telephone



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   events, establishes the initial content of that registry, and
   provides the media type registrations for the two payload formats.
   <a href="#appendix-A">Appendix A</a> describes the changes from <a href="./rfc2833">RFC 2833</a> [<a href="#ref-12" title="&quot;RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals&quot;">12</a>] and in particular
   indicates the disposition of the event codes defined in [<a href="#ref-12" title="&quot;RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals&quot;">12</a>].

<span class="h3"><a class="selflink" id="section-1.3" href="#section-1.3">1.3</a>.  Potential Applications</span>

   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 [<a href="#ref-22" title="&quot;ISDN user-network interface layer 3 specification for basic call control&quot;">22</a>] 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



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   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.

<span class="h3"><a class="selflink" id="section-1.4" href="#section-1.4">1.4</a>.  Events, States, Tone Patterns, and Voice-Encoded Tones</span>

   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.





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   1.  The gateway or end system can change to a higher-bandwidth codec
       such as G.711 [<a href="#ref-19" title="&quot;Pulse code modulation (PCM) of voice frequencies&quot;">19</a>] when tone signals are to be conveyed.  See new
       ITU-T Recommendation V.152 [<a href="#ref-26" title="&quot;Procedures for supporting Voice-Band Data over IP Networks&quot;">26</a>] for a formal treatment of this
       approach.  Alternatively, for fax, text, or modem signals
       respectively, a specialized transport such as T.38 [<a href="#ref-23" title="&quot;Procedures for real- time Group 3 facsimile communication over IP networks&quot;">23</a>], <a href="./rfc4103">RFC 4103</a>
       [<a href="#ref-15" title="&quot;RTP Payload for Text Conversation&quot;">15</a>], or V.150.1 modem relay [<a href="#ref-25" title="&quot;Modem-over-IP networks: Procedures for the end-to-end connection of V-series DCEs&quot;">25</a>] may be used.  Finally, 64
       kbit/s channels may be carried transparently using the <a href="./rfc4040">RFC 4040</a>
       Clearmode payload type [<a href="#ref-14" title="&quot;RTP Payload Format for a 64 kbit/s Transparent Call&quot;">14</a>].  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 (<a href="#section-4">Section 4</a>).  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 [<a href="#ref-24" title="&quot;Procedures for starting sessions of data transmission over the public switched telephone network&quot;">24</a>] 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 (<a href="#section-2">Section 2</a>).  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.











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<span class="h2"><a class="selflink" id="section-2" href="#section-2">2</a>.  RTP Payload Format for Named Telephone Events</span>

<span class="h3"><a class="selflink" id="section-2.1" href="#section-2.1">2.1</a>.  Introduction</span>

   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.

<span class="h3"><a class="selflink" id="section-2.2" href="#section-2.2">2.2</a>.  Use of RTP Header Fields</span>

<span class="h4"><a class="selflink" id="section-2.2.1" href="#section-2.2.1">2.2.1</a>.  Timestamp</span>

   The event duration described in <a href="#section-2.5">Section 2.5</a> 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, <a href="#section-2.5.1.3">Section 2.5.1.3</a>), the
   timestamp indicates the beginning of the segment.

<span class="h4"><a class="selflink" id="section-2.2.2" href="#section-2.2.2">2.2.2</a>.  Marker Bit</span>

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

<span class="h3"><a class="selflink" id="section-2.3" href="#section-2.3">2.3</a>.  Payload Format</span>

   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



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<span class="h4"><a class="selflink" id="section-2.3.1" href="#section-2.3.1">2.3.1</a>.  Event Field</span>

   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, <a href="#section-7">Section 7</a>).  The initial
   content of this registry consists of the events defined in <a href="#section-3">Section 3</a>.

<span class="h4"><a class="selflink" id="section-2.3.2" href="#section-2.3.2">2.3.2</a>.  E ("End") Bit</span>

   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, <a href="#section-2.5.1.3">Section 2.5.1.3</a>), only the final
   packet for the final segment will have the E bit set.

<span class="h4"><a class="selflink" id="section-2.3.3" href="#section-2.3.3">2.3.3</a>.  R Bit</span>

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

<span class="h4"><a class="selflink" id="section-2.3.4" href="#section-2.3.4">2.3.4</a>.  Volume Field</span>

   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.

<span class="h4"><a class="selflink" id="section-2.3.5" href="#section-2.3.5">2.3.5</a>.  Duration Field</span>

   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 (<a href="#section-2.5.1.3">Section 2.5.1.3</a>).

   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.





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   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.

<span class="h3"><a class="selflink" id="section-2.4" href="#section-2.4">2.4</a>.  Optional Media Type Parameters</span>

   As indicated in the media type registration for named events in
   <a href="#section-7.1.1">Section 7.1.1</a>, 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.

<span class="h4"><a class="selflink" id="section-2.4.1" href="#section-2.4.1">2.4.1</a>.  Relationship to SDP</span>

   The recommended mapping of media type optional parameters to SDP is
   given in <a href="./rfc3555#section-3">Section&nbsp;3 of RFC 3555</a> [<a href="#ref-6" title="&quot;MIME Type Registration of RTP Payload Formats&quot;">6</a>].  The "rate" media type parameter
   for the named event payload type follows this convention: it is
   expressed as usual as the &lt;clock rate&gt; component of the a=rtpmap:
   attribute line.

   The "events" media type parameter deviates from the convention
   suggested in <a href="./rfc3555">RFC 3555</a> because it omits the string "events=" before
   the list of supported events.

      a=fmtp:&lt;format&gt; &lt;list of values&gt;

   The list of values has the format and meaning described above.








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   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"

<span class="h3"><a class="selflink" id="section-2.5" href="#section-2.5">2.5</a>.  Procedures</span>

   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.

<span class="h4"><a class="selflink" id="section-2.5.1" href="#section-2.5.1">2.5.1</a>.  Sending Procedures</span>

<span class="h5"><a class="selflink" id="section-2.5.1.1" href="#section-2.5.1.1">2.5.1.1</a>.  Negotiation of Payloads</span>

   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 <a href="./rfc2198">RFC 2198</a> [<a href="#ref-2" title="&quot;RTP Payload for Redundant Audio Data&quot;">2</a>] 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 <a href="./rfc2198">RFC 2198</a>
   [<a href="#ref-2" title="&quot;RTP Payload for Redundant Audio Data&quot;">2</a>] redundancy.  <a href="#section-2.5.1.4">Section 2.5.1.4</a> describes how this can be used to
   avoid packet and RTP header overheads when retransmitting final event
   reports.  <a href="#section-2.6">Section 2.6</a> discusses the use of additional levels of <a href="./rfc2198">RFC</a>
   <a href="./rfc2198">2198</a> redundancy to increase the probability that at least one copy of



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   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 (<a href="./rfc3264">RFC 3264</a> [<a href="#ref-4" title="&quot;An Offer/Answer Model with Session Description Protocol (SDP)&quot;">4</a>]),
   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 <a href="#section-2.6.3">Section 2.6.3</a>.

<span class="h5"><a class="selflink" id="section-2.5.1.2" href="#section-2.5.1.2">2.5.1.2</a>.  Transmission of Event Packets</span>

   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.




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   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
   <a href="#section-2.5.1.4">Section 2.5.1.4</a>).  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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<span class="h5"><a class="selflink" id="section-2.5.1.3" href="#section-2.5.1.3">2.5.1.3</a>.  Long-Duration Events</span>

   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).

<span class="h6"><a class="selflink" id="section-2.5.1.3.1" href="#section-2.5.1.3.1">2.5.1.3.1</a>.  Exceptional Procedure for Combined Payloads</span>

   If events are combined as a redundant payload with another payload
   type using <a href="./rfc2198">RFC 2198</a> [<a href="#ref-2" title="&quot;RTP Payload for Redundant Audio Data&quot;">2</a>] 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 <a href="./rfc2198">RFC 2198</a> 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 <a href="./rfc2198">RFC 2198</a> 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.

<span class="h5"><a class="selflink" id="section-2.5.1.4" href="#section-2.5.1.4">2.5.1.4</a>.  Retransmission of Final Packet</span>

   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 <a href="./rfc2198">RFC 2198</a> [<a href="#ref-2" title="&quot;RTP Payload for Redundant Audio Data&quot;">2</a>] 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



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   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 <a href="./rfc2198">RFC 2198</a> 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.

<span class="h5"><a class="selflink" id="section-2.5.1.5" href="#section-2.5.1.5">2.5.1.5</a>.  Packing Multiple Events into One Packet</span>

   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.





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<span class="h5"><a class="selflink" id="section-2.5.1.6" href="#section-2.5.1.6">2.5.1.6</a>.  RTP Sequence Number</span>

   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.

<span class="h4"><a class="selflink" id="section-2.5.2" href="#section-2.5.2">2.5.2</a>.  Receiving Procedures</span>

<span class="h5"><a class="selflink" id="section-2.5.2.1" href="#section-2.5.2.1">2.5.2.1</a>.  Indication of Receiver Capabilities Using SDP</span>

   Receivers can indicate which named events they can handle, for
   example, by using the Session Description Protocol (<a href="./rfc4566">RFC 4566</a> [<a href="#ref-9" title="&quot;SDP: Session Description Protocol&quot;">9</a>]).
   SDP descriptions using the event payload MUST contain an fmtp format
   attribute that lists the event values that the receiver can process.

<span class="h5"><a class="selflink" id="section-2.5.2.2" href="#section-2.5.2.2">2.5.2.2</a>.  Playout of Tone Events</span>

   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 <a href="#section-2.6.3">Section 2.6.3</a>.





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   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





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   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 <a href="./rfc2833">RFC 2833</a> [<a href="#ref-12" title="&quot;RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals&quot;">12</a>], where the volume
   field was defined only for DTMF events.

<span class="h5"><a class="selflink" id="section-2.5.2.3" href="#section-2.5.2.3">2.5.2.3</a>.  Long-Duration Events</span>

   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 <a href="#section-2.5.1.3">Section 2.5.1.3</a>.  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.







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<span class="h6"><a class="selflink" id="section-2.5.2.3.1" href="#section-2.5.2.3.1">2.5.2.3.1</a>.  Exceptional Procedure for Combined Payloads</span>

   If events are combined as a redundant payload with another payload
   type using <a href="./rfc2198">RFC 2198</a> [<a href="#ref-2" title="&quot;RTP Payload for Redundant Audio Data&quot;">2</a>] redundancy, segments are generated at
   intervals of 0x3FFF or less, rather than 0xFFFF, as required by the
   procedures of <a href="#section-2.5.1.3.1">Section 2.5.1.3.1</a> 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.

<span class="h5"><a class="selflink" id="section-2.5.2.4" href="#section-2.5.2.4">2.5.2.4</a>.  Multiple Events in a Packet</span>

   The procedures of <a href="#section-2.5.1.5">Section 2.5.1.5</a> 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.

<span class="h5"><a class="selflink" id="section-2.5.2.5" href="#section-2.5.2.5">2.5.2.5</a>.  Soft States</span>

   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.

<span class="h3"><a class="selflink" id="section-2.6" href="#section-2.6">2.6</a>.  Congestion and Performance</span>

   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.











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<span class="h4"><a class="selflink" id="section-2.6.1" href="#section-2.6.1">2.6.1</a>.  Performance Requirements</span>

   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.

<span class="h4"><a class="selflink" id="section-2.6.2" href="#section-2.6.2">2.6.2</a>.  Reliability Mechanisms</span>

   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 <a href="#section-2.5.2.2">Section 2.5.2.2</a> 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




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       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 <a href="#section-2.5.1.4">Section 2.5.1.4</a>, 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 <a href="#section-2.6.1">Section 2.6.1</a>) 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 <a href="#section-2.5.1.4">Section 2.5.1.4</a> indicates that it is appropriate to use the
       <a href="./rfc2198">RFC 2198</a> [<a href="#ref-2" title="&quot;RTP Payload for Redundant Audio Data&quot;">2</a>] 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 <a href="./rfc2198">RFC 2198</a> 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 <a href="#section-2.5.2.2">Section 2.5.2.2</a> 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



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   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.

<span class="h4"><a class="selflink" id="section-2.6.3" href="#section-2.6.3">2.6.3</a>.  Adjusting to Congestion</span>

   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)




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   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 <a href="#section-2.5.2.2">Section 2.5.2.2</a> 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.

<span class="h2"><a class="selflink" id="section-3" href="#section-3">3</a>.  Specification of Event Codes for DTMF Events</span>

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

<span class="h3"><a class="selflink" id="section-3.1" href="#section-3.1">3.1</a>.  DTMF Applications</span>

   DTMF signalling [<a href="#ref-10" title="&quot;Technical features of push-button telephone sets&quot;">10</a>] 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.



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   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 [<a href="#ref-20" title="&quot;Speech coders : Dual rate speech coder for multimedia communications transmitting at 5.3 and 6.3 kbit/s&quot;">20</a>] 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 [<a href="#ref-11" title="&quot;Multifrequency push- button signal reception&quot;">11</a>], 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 [<a href="#ref-27" title="&quot;Operational and interworking requirements for {DCEs operating in the text telephone mode&quot;">27</a>] 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 &gt;=40 ms), but should send



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   event durations corresponding to those of a V.18 DTMF sender (tones
   &gt;=70 ms, pauses &gt;=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 <a href="#section-2.5.2.2">Section 2.5.2.2</a>.  This provides some robustness
   against packet loss or congestion.

<span class="h3"><a class="selflink" id="section-3.2" href="#section-3.2">3.2</a>.  DTMF Events</span>

   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

<span class="h3"><a class="selflink" id="section-3.3" href="#section-3.3">3.3</a>.  Congestion Considerations</span>

   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
   <a href="#section-3.1">Section 3.1</a> above, the second playout algorithm of <a href="#section-2.5.2.2">Section 2.5.2.2</a> 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.









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<span class="h2"><a class="selflink" id="section-4" href="#section-4">4</a>.  RTP Payload Format for Telephony Tones</span>

<span class="h3"><a class="selflink" id="section-4.1" href="#section-4.1">4.1</a>.  Introduction</span>

   As an alternative to describing tones and events by name, as
   described in <a href="#section-2">Section 2</a>, 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 [<a href="#ref-18" title="&quot;Technical characteristics of tones for the telephone service&quot;">18</a>] 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 [<a href="#ref-18" title="&quot;Technical characteristics of tones for the telephone service&quot;">18</a>] notes that different telephone
      companies require a tone accuracy of between 0.5 and 1.5%.  The
      Recommendation suggests a frequency tolerance of 1%.











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<span class="h3"><a class="selflink" id="section-4.2" href="#section-4.2">4.2</a>.  Examples of Common Telephone Tone Signals</span>

   As an aid to the implementor, Table 4 summarizes some common tones.
   The rows labeled "ITU ..." refer to ITU-T Recommendation E.180 [<a href="#ref-18" title="&quot;Technical characteristics of tones for the telephone service&quot;">18</a>].
   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

<span class="h3"><a class="selflink" id="section-4.3" href="#section-4.3">4.3</a>.  Use of RTP Header Fields</span>

<span class="h4"><a class="selflink" id="section-4.3.1" href="#section-4.3.1">4.3.1</a>.  Timestamp</span>

   The RTP timestamp reflects the measurement point for the current
   packet.  The event duration described in <a href="#section-4.3.3">Section 4.3.3</a> begins at that
   time.

<span class="h4"><a class="selflink" id="section-4.3.2" href="#section-4.3.2">4.3.2</a>.  Marker Bit</span>

   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.



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<span class="h4"><a class="selflink" id="section-4.3.3" href="#section-4.3.3">4.3.3</a>.  Payload Format</span>

   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.



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   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.

<span class="h4"><a class="selflink" id="section-4.3.4" href="#section-4.3.4">4.3.4</a>.  Optional Media Type Parameters</span>

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

<span class="h3"><a class="selflink" id="section-4.4" href="#section-4.4">4.4</a>.  Procedures</span>

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

<span class="h4"><a class="selflink" id="section-4.4.1" href="#section-4.4.1">4.4.1</a>.  Sending Procedures</span>

   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.





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   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 <a href="#section-2">Section 2</a> 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 <a href="./rfc2198">RFC 2198</a> [<a href="#ref-2" title="&quot;RTP Payload for Redundant Audio Data&quot;">2</a>] audio
   redundancy.

<span class="h4"><a class="selflink" id="section-4.4.2" href="#section-4.4.2">4.4.2</a>.  Receiving Procedures</span>

   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.

<span class="h4"><a class="selflink" id="section-4.4.3" href="#section-4.4.3">4.4.3</a>.  Handling of Congestion</span>

   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.









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<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


<span class="h2"><a class="selflink" id="section-5" href="#section-5">5</a>.  Examples</span>

   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    |      |        |      |        |        |      |





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   |   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







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<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


   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



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-34" ></span>
<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


   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 <a href="./rfc2198">RFC</a>
   <a href="./rfc2198">2198</a> 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 [<a href="#ref-21" title="&quot;Coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear- prediction (CS-ACELP)&quot;">21</a>] 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













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       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



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<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


   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 <a href="./rfc2198">RFC 2198</a> header, and thus are
   displaced one octet into successive words.

































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       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














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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-38" ></span>
<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


<span class="h2"><a class="selflink" id="section-6" href="#section-6">6</a>.  Security Considerations</span>

   RTP packets using the payload formats defined in this specification
   are subject to the security considerations discussed in the RTP
   specification (<a href="./rfc3550">RFC 3550</a> [<a href="#ref-5" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">5</a>]), and any appropriate RTP profile (for
   example, <a href="./rfc3551">RFC 3551</a> [<a href="#ref-13" title="&quot;RTP Profile for Audio and Video Conferences with Minimal Control&quot;">13</a>]).  The <a href="./rfc3550">RFC 3550</a> discussion focuses on
   requirements for confidentiality.  Additional security considerations
   relating to implementation are described in <a href="./rfc2198">RFC 2198</a> [<a href="#ref-2" title="&quot;RTP Payload for Redundant Audio Data&quot;">2</a>].

   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) [<a href="#ref-7" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">7</a>].

      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.

<span class="h2"><a class="selflink" id="section-7" href="#section-7">7</a>.  IANA Considerations</span>

   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 <a href="./rfc2434">RFC 2434</a> [<a href="#ref-3" title="">3</a>].  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;





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   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 <a href="./rfc2833">RFC 2833</a> but not specified in any new document.  In this
   case, the event code specified in <a href="./rfc2833">RFC 2833</a> 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 <a href="./rfc2833">RFC 2833</a> 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 <a href="#section-2.6">Section 2.6</a>, 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
   <a href="#section-3">Section 3</a> 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
      [<a href="#ref-16" title="&quot;Definition of Events for Modem, Fax, and Text Telephony Signals&quot;">16</a>];

   o  codes 121-137 and 174-205 are reserved for events defined in [<a href="#ref-17" title="&quot;Definition of Events For Channel-Oriented Telephony Signalling&quot;">17</a>];

   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 <a href="./rfc2833">RFC 2833</a> events, and in the second instance for
      general assignment after all other codes have been assigned.







<span class="grey">Schulzrinne &amp; Taylor        Standards Track                    [Page 39]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-40" ></span>
<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


        +------------+--------------------------------+-----------+
        | Event Code | Event Name                     | Reference |
        +------------+--------------------------------+-----------+
        |          0 | DTMF digit "0"                 |  <a href="./rfc4733">RFC 4733</a> |
        |          1 | DTMF digit "1"                 |  <a href="./rfc4733">RFC 4733</a> |
        |          2 | DTMF digit "2"                 |  <a href="./rfc4733">RFC 4733</a> |
        |          3 | DTMF digit "3"                 |  <a href="./rfc4733">RFC 4733</a> |
        |          4 | DTMF digit "4"                 |  <a href="./rfc4733">RFC 4733</a> |
        |          5 | DTMF digit "5"                 |  <a href="./rfc4733">RFC 4733</a> |
        |          6 | DTMF digit "6"                 |  <a href="./rfc4733">RFC 4733</a> |
        |          7 | DTMF digit "7"                 |  <a href="./rfc4733">RFC 4733</a> |
        |          8 | DTMF digit "8"                 |  <a href="./rfc4733">RFC 4733</a> |
        |          9 | DTMF digit "9"                 |  <a href="./rfc4733">RFC 4733</a> |
        |         10 | DTMF digit "*"                 |  <a href="./rfc4733">RFC 4733</a> |
        |         11 | DTMF digit "#"                 |  <a href="./rfc4733">RFC 4733</a> |
        |         12 | DTMF digit "A"                 |  <a href="./rfc4733">RFC 4733</a> |
        |         13 | DTMF digit "B"                 |  <a href="./rfc4733">RFC 4733</a> |
        |         14 | DTMF digit "C"                 |  <a href="./rfc4733">RFC 4733</a> |
        |         15 | DTMF digit "D"                 |  <a href="./rfc4733">RFC 4733</a> |
        +------------+--------------------------------+-----------+

            Table 7: audio/telephone-event Event Code Registry

<span class="h3"><a class="selflink" id="section-7.1" href="#section-7.1">7.1</a>.  Media Type Registrations</span>

<span class="h4"><a class="selflink" id="section-7.1.1" href="#section-7.1.1">7.1.1</a>.  Registration of Media Type audio/telephone-event</span>

   This registration is done in accordance with [<a href="#ref-6" title="&quot;MIME Type Registration of RTP Payload Formats&quot;">6</a>] and [<a href="#ref-8" title="&quot;Media Type Specifications and Registration Procedures&quot;">8</a>].

   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.



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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-41" ></span>
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      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 [<a href="#ref-8" title="&quot;Media Type Specifications and Registration Procedures&quot;">8</a>] <a href="#section-4.8">section 4.8</a>, this type is framed
      and binary.

   Security considerations:

      See <a href="#section-6">Section 6</a>, "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 &amp; 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 [<a href="#ref-5" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">5</a>].

   Author: IETF Audio/Video Transport Working Group.

   Change controller:

      IETF Audio/Video Transport Working Group as delegated from the
      IESG.






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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-42" ></span>
<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


<span class="h4"><a class="selflink" id="section-7.1.2" href="#section-7.1.2">7.1.2</a>.  Registration of Media Type audio/tone</span>

   This registration is done in accordance with [<a href="#ref-6" title="&quot;MIME Type Registration of RTP Payload Formats&quot;">6</a>] and [<a href="#ref-8" title="&quot;Media Type Specifications and Registration Procedures&quot;">8</a>].

   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 [<a href="#ref-8" title="&quot;Media Type Specifications and Registration Procedures&quot;">8</a>] <a href="#section-4.8">section 4.8</a>, this type is framed
      and binary.

   Security considerations:

      See <a href="#section-6">Section 6</a>, "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 &amp; email address to contact for further information:

      Tom Taylor, taylor@nortel.com.
      IETF AVT Working Group.

   Intended usage: COMMON.




<span class="grey">Schulzrinne &amp; Taylor        Standards Track                    [Page 42]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-43" ></span>
<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


   Restrictions on usage:

      This type is defined only for transfer via RTP [<a href="#ref-5" title="&quot;RTP: A Transport Protocol for Real-Time Applications&quot;">5</a>].

   Author: IETF Audio/Video Transport Working Group.

   Change controller:

      IETF Audio/Video Transport Working Group as delegated from the
      IESG.

<span class="h2"><a class="selflink" id="section-8" href="#section-8">8</a>.  Acknowledgements</span>

   Scott Petrack was the original author of <a href="./rfc2833">RFC 2833</a>.  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 <a href="./rfc2833">RFC 2833</a>, 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
   <a href="./rfc2833">RFC 2833</a>, 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.

<span class="h2"><a class="selflink" id="section-9" href="#section-9">9</a>.  References</span>

<span class="h3"><a class="selflink" id="section-9.1" href="#section-9.1">9.1</a>.  Normative References</span>

   [<a id="ref-1">1</a>]   Bradner, S., "Key words for use in RFCs to Indicate Requirement
         Levels", <a href="https://www.rfc-editor.org/bcp/bcp14">BCP 14</a>, <a href="./rfc2119">RFC 2119</a>, March 1997.

   [<a id="ref-2">2</a>]   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", <a href="./rfc2198">RFC 2198</a>, September 1997.

   [<a id="ref-3">3</a>]   Narten, T. and H. Alvestrand, "Guidelines for Writing an IANA
         Considerations Section in RFCs", <a href="https://www.rfc-editor.org/bcp/bcp26">BCP 26</a>, <a href="./rfc2434">RFC 2434</a>,
         October 1998.

   [<a id="ref-4">4</a>]   Rosenberg, J. and H. Schulzrinne, "An Offer/Answer Model with
         Session Description Protocol (SDP)", <a href="./rfc3264">RFC 3264</a>, June 2002.




<span class="grey">Schulzrinne &amp; Taylor        Standards Track                    [Page 43]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-44" ></span>
<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


   [<a id="ref-5">5</a>]   Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
         "RTP: A Transport Protocol for Real-Time Applications", STD 64,
         <a href="./rfc3550">RFC 3550</a>, July 2003.

   [<a id="ref-6">6</a>]   Casner, S. and P. Hoschka, "MIME Type Registration of RTP
         Payload Formats", <a href="./rfc3555">RFC 3555</a>, July 2003.

   [<a id="ref-7">7</a>]   Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
         Norrman, "The Secure Real-time Transport Protocol (SRTP)",
         <a href="./rfc3711">RFC 3711</a>, March 2004.

   [<a id="ref-8">8</a>]   Freed, N. and J. Klensin, "Media Type Specifications and
         Registration Procedures", <a href="https://www.rfc-editor.org/bcp/bcp13">BCP 13</a>, <a href="./rfc4288">RFC 4288</a>, December 2005.

   [<a id="ref-9">9</a>]   Handley, M., Jacobson, V., and C. Perkins, "SDP: Session
         Description Protocol", <a href="./rfc4566">RFC 4566</a>, July 2006.

   [<a id="ref-10">10</a>]  International Telecommunication Union, "Technical features of
         push-button telephone sets", ITU-T Recommendation Q.23,
         November 1988.

   [<a id="ref-11">11</a>]  International Telecommunication Union, "Multifrequency push-
         button signal reception", ITU-T Recommendation Q.24,
         November 1988.

<span class="h3"><a class="selflink" id="section-9.2" href="#section-9.2">9.2</a>.  Informative References</span>

   [<a id="ref-12">12</a>]  Schulzrinne, H. and S. Petrack, "RTP Payload for DTMF Digits,
         Telephony Tones and Telephony Signals", <a href="./rfc2833">RFC 2833</a>, May 2000.

   [<a id="ref-13">13</a>]  Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
         Conferences with Minimal Control", STD 65, <a href="./rfc3551">RFC 3551</a>, July 2003.

   [<a id="ref-14">14</a>]  Kreuter, R., "RTP Payload Format for a 64 kbit/s Transparent
         Call", <a href="./rfc4040">RFC 4040</a>, April 2005.

   [<a id="ref-15">15</a>]  Hellstrom, G. and P. Jones, "RTP Payload for Text
         Conversation", <a href="./rfc4103">RFC 4103</a>, June 2005.

   [<a id="ref-16">16</a>]  Schulzrinne, H. and T. Taylor, "Definition of Events for Modem,
         Fax, and Text Telephony Signals", <a href="./rfc4734">RFC 4734</a>, December 2006.

   [<a id="ref-17">17</a>]  Schulzrinne, H. and T. Taylor, "Definition of Events For
         Channel-Oriented Telephony Signalling", Work In Progress ,
         November 2005.






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<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-45" ></span>
<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


   [<a id="ref-18">18</a>]  International Telecommunication Union, "Technical
         characteristics of tones for the telephone service", ITU-T
         Recommendation E.180/Q.35, March 1998.

   [<a id="ref-19">19</a>]  International Telecommunication Union, "Pulse code modulation
         (PCM) of voice frequencies", ITU-T Recommendation G.711,
         November 1988.

   [<a id="ref-20">20</a>]  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.

   [<a id="ref-21">21</a>]  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.

   [<a id="ref-22">22</a>]  International Telecommunication Union, "ISDN user-network
         interface layer 3 specification for basic call control", ITU-T
         Recommendation Q.931, May 1998.

   [<a id="ref-23">23</a>]  International Telecommunication Union, "Procedures for real-
         time Group 3 facsimile communication over IP networks", ITU-T
         Recommendation T.38, July 2003.

   [<a id="ref-24">24</a>]  International Telecommunication Union, "Procedures for starting
         sessions of data transmission over the public switched
         telephone network", ITU-T Recommendation V.8, November 2000.

   [<a id="ref-25">25</a>]  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.

   [<a id="ref-26">26</a>]  International Telecommunication Union, "Procedures for
         supporting Voice-Band Data over IP Networks", ITU-T
         Recommendation V.152, January 2005.

   [<a id="ref-27">27</a>]  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,
         &lt;<a href="http://www.voiptroubleshooter.com/indepth/burstloss.html">http://www.voiptroubleshooter.com/indepth/burstloss.html</a>&gt;.







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<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


<span class="h2"><a class="selflink" id="appendix-A" href="#appendix-A">Appendix A</a>.  Summary of Changes from <a href="./rfc2833">RFC 2833</a></span>

   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 <a href="./rfc2833">RFC 2833</a> [<a href="#ref-12" title="&quot;RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals&quot;">12</a>] it is likely that some
   implementations interpreted that memo in ways that differ from this
   version.

   <a href="./rfc2833">RFC 2833</a> required that all implementations be capable of receiving
   the DTMF events (event codes 0-15).  <a href="#section-2.5.1.1">Section 2.5.1.1</a> 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.  <a href="#section-2.5.2.1">Section 2.5.2.1</a> 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 <a href="./rfc2833">RFC 2833</a>, 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 <a href="#section-2.5.2.2">Section 2.5.2.2</a>
   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 <a href="./rfc2833">RFC 2833</a> receiver rule, this ensures backward compatibility
   in both directions of transmission.

   <a href="#section-2.5.1.3">Section 2.5.1.3</a> and <a href="#section-2.5.2.3">Section 2.5.2.3</a> 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 (<a href="./rfc2833">RFC 2833</a>) 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.

   <a href="#section-2.5.1.5">Section 2.5.1.5</a> and <a href="#section-2.5.2.4">Section 2.5.2.4</a> introduce a new procedure whereby
   a sequence of short-duration events may be packed into a single event
   report.  If an old (<a href="./rfc2833">RFC 2833</a>) 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.








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<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


   <a href="#section-2.3.5">Section 2.3.5</a> introduces the possibility of "state" events and
   defines procedures for setting the duration field for reports of such
   events.  <a href="#section-2.5.1.2">Section 2.5.1.2</a> 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 [<a href="#ref-7" title="&quot;The Secure Real-time Transport Protocol (SRTP)&quot;">7</a>] 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 [<a href="#ref-16" title="&quot;Definition of Events for Modem, Fax, and Text Telephony Signals&quot;">16</a>] and
   [<a href="#ref-17" title="&quot;Definition of Events For Channel-Oriented Telephony Signalling&quot;">17</a>] 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 <a href="./rfc2833">RFC 2833</a> assignments.  The remaining events defined in <a href="./rfc2833">RFC 2833</a>
   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 <a href="./rfc2833">RFC 2833</a> and continue to be unused.

   +-------------+---------------------------------------+-------------+
   | Event Codes | <a href="./rfc2833">RFC 2833</a> Description                  | Disposition |
   +-------------+---------------------------------------+-------------+
   |        0-15 | DTMF digits                           | <a href="./rfc4733">RFC 4733</a>    |
   |          16 | Line flash (deprecated)               | Reserved    |
   |       23-31 | Unused                                | [<a href="#ref-16" title="&quot;Definition of Events for Modem, Fax, and Text Telephony Signals&quot;">16</a>]        |
   |       32-40 | Data and fax                          | [<a href="#ref-16" title="&quot;Definition of Events for Modem, Fax, and Text Telephony Signals&quot;">16</a>]        |
   |       41-48 | Data and fax (V.8bis, deprecated)     | Reserved    |
   |       52-63 | Unused                                | [<a href="#ref-16" title="&quot;Definition of Events for Modem, Fax, and Text Telephony Signals&quot;">16</a>]        |
   |       64-89 | E.182 line events (deprecated)        | Reserved    |
   |      96-112 | Country-specific line events          | Reserved    |
   |             | (deprecated)                          |             |
   |     121-127 | Unused                                | [<a href="#ref-17" title="&quot;Definition of Events For Channel-Oriented Telephony Signalling&quot;">17</a>]        |
   |     128-137 | Trunks: MF 0-9                        | [<a href="#ref-17" title="&quot;Definition of Events For Channel-Oriented Telephony Signalling&quot;">17</a>]        |
   |     138-143 | Trunks: other MF (deprecated)         | Reserved    |
   |     144-159 | Trunks: ABCD signalling               | [<a href="#ref-17" title="&quot;Definition of Events For Channel-Oriented Telephony Signalling&quot;">17</a>]        |
   |     160-168 | Trunks: various (deprecated)          | Reserved    |
   |     170-173 | Trunks: various (deprecated)          | Reserved    |
   |     174-205 | Unused                                | [<a href="#ref-17" title="&quot;Definition of Events For Channel-Oriented Telephony Signalling&quot;">17</a>]        |
   +-------------+---------------------------------------+-------------+

           Table 8: Disposition of <a href="./rfc2833">RFC 2833</a>-defined Event Codes



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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































<span class="grey">Schulzrinne &amp; Taylor        Standards Track                    [Page 48]</span></pre>
<hr class='noprint'/><!--NewPage--><pre class='newpage'><span id="page-49" ></span>
<span class="grey"><a href="./rfc4733">RFC 4733</a>               Telephony Events and Tones          December 2006</span>


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Schulzrinne &amp; Taylor        Standards Track                    [Page 49]
</pre>