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RFC 4733

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

Pages: 49
Proposed Standard
Errata
Obsoletes:  2833
Updated by:  47345244
Part 1 of 2 – Pages 1 to 23
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Network Working Group                                     H. Schulzrinne
Request for Comments: 4733                                   Columbia U.
Obsoletes: 2833                                                T. Taylor
Category: Standards Track                                         Nortel
                                                           December 2006


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

Status of This Memo

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

Copyright Notice

   Copyright (C) The IETF Trust (2006).

Abstract

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

1. Introduction ....................................................4 1.1. Terminology ................................................4 1.2. Overview ...................................................4 1.3. Potential Applications .....................................5 1.4. Events, States, Tone Patterns, and Voice-Encoded Tones .....6 2. RTP Payload Format for Named Telephone Events ...................8 2.1. Introduction ...............................................8 2.2. Use of RTP Header Fields ...................................8 2.2.1. Timestamp ...........................................8 2.2.2. Marker Bit ..........................................8 2.3. Payload Format .............................................8 2.3.1. Event Field .........................................9 2.3.2. E ("End") Bit .......................................9 2.3.3. R Bit ...............................................9 2.3.4. Volume Field ........................................9 2.3.5. Duration Field ......................................9 2.4. Optional Media Type Parameters ............................10 2.4.1. Relationship to SDP ................................10 2.5. Procedures ................................................11 2.5.1. Sending Procedures .................................11 2.5.2. Receiving Procedures ...............................16 2.6. Congestion and Performance ................................19 2.6.1. Performance Requirements ...........................20 2.6.2. Reliability Mechanisms .............................20 2.6.3. Adjusting to Congestion ............................22 3. Specification of Event Codes for DTMF Events ...................23 3.1. DTMF Applications .........................................23 3.2. DTMF Events ...............................................25 3.3. Congestion Considerations .................................25 4. RTP Payload Format for Telephony Tones .........................26 4.1. Introduction ..............................................26 4.2. Examples of Common Telephone Tone Signals .................27 4.3. Use of RTP Header Fields ..................................27 4.3.1. Timestamp ..........................................27 4.3.2. Marker Bit .........................................27 4.3.3. Payload Format .....................................28 4.3.4. Optional Media Type Parameters .....................29 4.4. Procedures ................................................29 4.4.1. Sending Procedures .................................29 4.4.2. Receiving Procedures ...............................30 4.4.3. Handling of Congestion .............................30 5. Examples .......................................................31 6. Security Considerations ........................................38
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   7. IANA Considerations ............................................38
      7.1. Media Type Registrations ..................................40
           7.1.1. Registration of Media Type audio/telephone-event ...40
           7.1.2. Registration of Media Type audio/tone ..............42
   8. Acknowledgements ...............................................43
   9. References .....................................................43
      9.1. Normative References ......................................43
      9.2. Informative References ....................................44
   Appendix A. Summary of Changes from RFC 2833 ......................46
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1. Introduction

1.1. Terminology

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

1.2. Overview

This memo defines two RTP [5] payload formats, one for carrying dual-tone multifrequency (DTMF) digits and other line and trunk signals as events (Section 2), and a second one to describe general multifrequency tones in terms only of their frequency and cadence (Section 4). Separate RTP payload formats for telephony tone signals are desirable since low-rate voice codecs cannot be guaranteed to reproduce these tone signals accurately enough for automatic recognition. In addition, tone properties such as the phase reversals in the ANSam tone will not survive speech coding. Defining separate payload formats also permits higher redundancy while maintaining a low bit rate. Finally, some telephony events such as "on-hook" occur out-of-band and cannot be transmitted as tones. The remainder of this section provides the motivation for defining the payload types described in this document. Section 2 defines the payload format and associated procedures for use of named events. Section 3 describes the events for which event codes are defined in this document. Section 4 describes the payload format and associated procedures for tone representations. Section 5 provides some examples of encoded events, tones, and combined payloads. Section 6 deals with security considerations. Section 7 defines the IANA requirements for registration of event codes for named telephone
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   events, establishes the initial content of that registry, and
   provides the media type registrations for the two payload formats.
   Appendix A describes the changes from RFC 2833 [12] and in particular
   indicates the disposition of the event codes defined in [12].

1.3. Potential Applications

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

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

This document provides the means for in-band transport over the Internet of two broad classes of signalling information: in-band tones or tone sequences, and signals sent out-of-band in the PSTN. Tone signals can be carried using any of the three methods listed below. Depending on the application, it may be desirable to carry the signalling information in more than one form at once.
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   1.  The gateway or end system can change to a higher-bandwidth codec
       such as G.711 [19] when tone signals are to be conveyed.  See new
       ITU-T Recommendation V.152 [26] for a formal treatment of this
       approach.  Alternatively, for fax, text, or modem signals
       respectively, a specialized transport such as T.38 [23], RFC 4103
       [15], or V.150.1 modem relay [25] may be used.  Finally, 64
       kbit/s channels may be carried transparently using the RFC 4040
       Clearmode payload type [14].  These methods are out of scope of
       the present document, but may be used along with the payload
       types defined here.

   2.  The sending gateway can simply measure the frequency components
       of the voice-band signals and transmit this information to the
       RTP receiver using the tone representation defined in this
       document (Section 4).  In this mode, the gateway makes no attempt
       to discern the meaning of the tones, but simply distinguishes
       tones from speech signals.  An end system may use the same
       approach using configured rather than measured frequencies.

       All tone signals in use in the PSTN and meant for human
       consumption are sequences of simple combinations of sine waves,
       either added or modulated.  (However, some modem signals such as
       the ANSam tone [24] or systems dependent on phase shift keying
       cannot be conveyed so simply.)

   3.  As a third option, a sending gateway can recognize tones such as
       ringing or busy tone or DTMF digit '0', and transmit a code that
       identifies them using the telephone-event payload defined in this
       document (Section 2).  The receiver then produces a tone signal
       or other indication appropriate to the signal.  Generally, since
       the recognition of signals at the sender often depends on their
       on/off pattern or the sequence of several tones, this recognition
       can take several seconds.  On the other hand, the gateway may
       have access to the actual signalling information that generates
       the tones and thus can generate the RTP packet immediately,
       without the detour through acoustic signals.

   The third option (use of named events) is the only feasible method
   for transmitting out-of-band PSTN signals as content within RTP
   sessions.
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2. RTP Payload Format for Named Telephone Events

2.1. Introduction

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

2.2. Use of RTP Header Fields

2.2.1. Timestamp

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

2.2.2. Marker Bit

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

2.3. Payload Format

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

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

2.3.2. E ("End") Bit

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

2.3.3. R Bit

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

2.3.4. Volume Field

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

2.3.5. Duration Field

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

2.4. Optional Media Type Parameters

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

2.4.1. Relationship to SDP

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

2.5. Procedures

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

2.5.1. Sending Procedures

2.5.1.1. Negotiation of Payloads
Events are usually sent in combination with or alternating with other payload types. Payload negotiation may specify separate event and other payload streams, or it may specify a combined stream that mixes other payload types with events using RFC 2198 [2] redundancy headers. The purpose of using a combined stream may be for debugging or to ease the transition between general audio and events. Negotiation of payloads between sender and receiver is achieved by out-of-band means, using SDP, for example. The sender SHOULD indicate what events it supports, using the optional "events" parameter associated with the telephone-event media type. If the sender receives an "events" parameter from the receiver, it MUST restrict the set of events it sends to those listed in the received "events" parameter. For backward compatibility, if no "events" parameter is received, the sender SHOULD assume support for the DTMF events 0-15 but for no other events. Events MAY be sent in combination with older events using RFC 2198 [2] redundancy. Section 2.5.1.4 describes how this can be used to avoid packet and RTP header overheads when retransmitting final event reports. Section 2.6 discusses the use of additional levels of RFC 2198 redundancy to increase the probability that at least one copy of
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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 (RFC 3264 [4]),
   the SDP a=ptime: attribute indicates the packetization period that
   the author of the session description expects when receiving media.
   This value does not have to be the same in both directions.  The
   appropriate period may vary with the application, since increased
   packetization periods imply increased end-to-end response times in
   instances where one end responds to events reported from the other.

   Negotiation of telephone-events sessions using SDP MAY specify such
   differences by separating events corresponding to different
   applications into different streams.  In the example below, events
   0-15 are DTMF events, which have a fairly wide tolerance on timing.
   Events 32-49 and 52-60 are events related to data transmission and
   are subject to end-to-end response time considerations.  As a result,
   they are assigned a smaller packetization period than the DTMF
   events.

      m=audio 12344 RTP/AVP 99
      a=rtpmap:99 telephone-event/8000
      a=fmtp:99 0-15
      a=ptime:50
      m=audio 12346 RTP/AVP 100
      a=rtpmap:100 telephone-event/8000
      a=fmtp:100 32-49,52-60
      a=ptime:30

   For further discussion of packetization periods see Section 2.6.3.

2.5.1.2. Transmission of Event Packets
DTMF digits and other named telephone events are carried as part of the audio stream, and they MUST use the same sequence number and timestamp base as the regular audio channel to simplify the generation of audio waveforms at a gateway.
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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
   Section 2.5.1.4).  Intermediate packets for an event MUST NOT have
   either the M bit or the E bit set.

   Sending of a packet with the E bit set is OPTIONAL if the packet
   reports two events that are defined as mutually exclusive states, or
   if the final packet for one state is immediately followed by a packet
   reporting a mutually exclusive state.  (For events defined as states,
   the appearance of a mutually exclusive state implies the end of the
   previous state.)

   A source has wide latitude as to how often it sends event updates.  A
   natural interval is the spacing between non-event audio packets.
   (Recall that a single RTP packet can contain multiple audio frames
   for frame-based codecs and that the packet interval can vary during a
   session.)  Alternatively, a source MAY decide to use a different
   spacing for event updates, with a value of 50 ms RECOMMENDED.

   Timing information is contained in the RTP timestamp, allowing
   precise recovery of inter-event times.  Thus, the sender does not in
   theory need to maintain precise or consistent time intervals between
   event packets.  However, the sender SHOULD minimize the need for
   buffering at the receiving end by sending event reports at constant
   intervals.

      DTMF digits and other tone events are sent incrementally to avoid
      having the receiver wait for the completion of the event.  In some
      cases (for example, data session startup protocols), waiting until
      the end of a tone before reporting it will cause the session to
      fail.  In other cases, it will simply cause undesirable delays in
      playout at the receiving end.

   For robustness, the sender SHOULD retransmit "state" events
   periodically.
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2.5.1.3. Long-Duration Events
If an event persists beyond the maximum duration expressible in the duration field (0xFFFF), the sender MUST send a packet reporting this maximum duration but MUST NOT set the E bit in this packet. The sender MUST then begin reporting a new "segment" with the RTP timestamp set to the time at which the previous segment ended and the duration set to the cumulative duration of the new segment. The M bit of the first packet reporting the new segment MUST NOT be set. The sender MUST repeat this procedure as required until the end of the complete event has been reached. The final packet for the complete event MUST have the E bit set (either on initial transmission or on retransmission as described below).
2.5.1.3.1. Exceptional Procedure for Combined Payloads
If events are combined as a redundant payload with another payload type using RFC 2198 [2] redundancy, the above procedure SHALL be applied, but using a maximum duration that ensures that the timestamp offset of the oldest generation of events in an RFC 2198 packet never exceeds 0x3FFF. If the sender is using a constant packetization period, the maximum segment duration can be calculated from the following formula: maximum duration = 0x3FFF - (R-1)*(packetization period in timestamp units) where R is the highest redundant layer number consisting of event payload. The RFC 2198 redundancy header timestamp offset value is only 14 bits, compared with the 16 bits in the event payload duration field. Since with other payloads the RTP timestamp typically increments for each new sample, the timestamp offset value becomes limiting on reported event duration. The limit becomes more constraining when older generations of events are also included in the combined payload.
2.5.1.4. Retransmission of Final Packet
The final packet for each event and for each segment SHOULD be sent a total of three times at the interval used by the source for updates. This ensures that the duration of the event or segment can be recognized correctly even if an instance of the last packet is lost. A sender MAY use RFC 2198 [2] with up to two levels of redundancy to combine retransmissions with reports of new events, thus saving on header overheads. In this usage, the primary payload is new event
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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 RFC 2198 payload SHOULD NOT be sent except when both
   primary and retransmitted reports are to be included.  All other
   packets of the session SHOULD contain only the simple, non-redundant
   telephone-event payload.  Note that the expected proportion of simple
   versus redundant packets affects the order in which they should be
   specified on an SDP m= line.

      There is little point in sending initial or interim event reports
      redundantly because each succeeding packet describes the event
      fully (except for typically irrelevant variations in volume).

   A sender MAY delay setting the E bit until retransmitting the last
   packet for a tone, rather than setting the bit on its first
   transmission.  This avoids having to wait to detect whether the tone
   has indeed ended.  Once the sender has set the E bit for a packet, it
   MUST continue to set the E bit for any further retransmissions of
   that packet.

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

2.5.2. Receiving Procedures

2.5.2.1. Indication of Receiver Capabilities Using SDP
Receivers can indicate which named events they can handle, for example, by using the Session Description Protocol (RFC 4566 [9]). SDP descriptions using the event payload MUST contain an fmtp format attribute that lists the event values that the receiver can process.
2.5.2.2. Playout of Tone Events
In the gateway scenario, an Internet telephony gateway connecting a packet voice network to the PSTN re-creates the DTMF or other tones and injects them into the PSTN. Since, for example, DTMF digit recognition takes several tens of milliseconds, the first few milliseconds of a digit will arrive as regular audio packets. Thus, careful time and power (volume) alignment between the audio samples and the events is needed to avoid generating spurious digits at the receiver. The receiver may also choose to delay playout of the tones by some small interval after playout of the preceding audio has ended, to ensure that downstream equipment can discriminate the tones properly. Some implementations send events and encoded audio packets (e.g., PCMU or the codec used for speech signals) for the same time instant for the duration of the event. It is RECOMMENDED that gateways render only the telephone-event payload once it is received, since the audio may contain spurious tones introduced by the audio compression algorithm. However, it is anticipated that these extra tones in general should not interfere with recognition at the far end. Receiver implementations MAY use different algorithms to create tones, including the two described here. (Note that not all implementations have the need to re-create a tone; some may only care about recognizing the events.) With either algorithm, a receiver may impose a playout delay to provide robustness against packet loss or delay. The tradeoff between playout delay and other factors is discussed further in Section 2.6.3.
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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 RFC 2833 [12], where the volume
   field was defined only for DTMF events.

2.5.2.3. Long-Duration Events
If an event report is received with duration equal to the maximum duration expressible in the duration field (0xFFFF) and the E bit for the report is not set, the event report may mark the end of a segment generated according to the procedures of Section 2.5.1.3. If another report for the same event type is received, the receiver MUST compare the RTP timestamp for the new event with the sum of the RTP timestamp of the previous report plus the duration (0xFFFF). The receiver uses the absence of a gap between the events to detect that it is receiving a single long-duration event. The total duration of a long-duration event is (obviously) the sum of the durations of the segments used to report it. This is equal to the duration of the final segment (as indicated in the final packet for that segment), plus 0xFFFF multiplied by the number of segments preceding the final segment.
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2.5.2.3.1. Exceptional Procedure for Combined Payloads
If events are combined as a redundant payload with another payload type using RFC 2198 [2] redundancy, segments are generated at intervals of 0x3FFF or less, rather than 0xFFFF, as required by the procedures of Section 2.5.1.3.1 in this case. If a receiver is using the events component of the payload, event duration may be only an approximate indicator of division into segments, but the lack of an E bit and the adjacency of two reports with the same event code are strong indicators in themselves.
2.5.2.4. Multiple Events in a Packet
The procedures of Section 2.5.1.5 require that if multiple events are reported in the same packet, they are contiguous and non-overlapping. As a result, it is not strictly necessary for the receiver to know the start times of the events following the first one in order to play them out -- it needs only to respect the duration reported for each event. Nevertheless, if knowledge of the start time for a given event after the first one is required, it is equal to the sum of the start time of the preceding event plus the duration of the preceding event.
2.5.2.5. Soft States
If the duration of a soft state event expires, the receiver SHOULD consider the value of the state to be "unknown" unless otherwise indicated in the event documentation.

2.6. Congestion and Performance

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

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

2.6.2. Reliability Mechanisms

To improve reliability, all payload types including telephone-events can use a jitter buffer, i.e., impose a playout delay, at the receiving end. This mechanism addresses the first four requirements listed above, but at the expense of the last one. The named event procedures provide two complementary redundancy mechanisms to deal with lost packets: a. Intra-event updates: Events that last longer than one packetization period (e.g., 50 ms) are updated periodically, so that the receiver can reconstruct the event and its duration if it receives any of the update packets, albeit with delay. During an event, the RTP event payload format provides incremental updates on the event. The error resiliency afforded by this mechanism depends on whether the first or second algorithm in Section 2.5.2.2 is used and on the playout delay at the receiver. For example, if the receiver uses the first algorithm and only places the current duration of tone signal in the playout buffer, for a playout delay of 120 ms and a
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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 Section 2.5.1.4, the last report for an event is
       transmitted a total of three times.  This mechanism adds
       robustness to the reporting of the end of an event.

       It may be necessary to extend the level of redundancy to achieve
       requirement a) (in Section 2.6.1) in a specific network
       environment.  Taking the 25-30% loss rate during congestion
       periods illustrated in [28] as typical, and setting an objective
       that at least 99% of end-of-event reports will eventually get
       through to the receiver under these conditions, simple
       probability calculations indicate that each event completion has
       to be reported four times.  This is one more level of redundancy
       than required by the basic "Repeat last event packet" algorithm.
       Of course, the objective is probably unrealistically stringent;
       it was chosen to make a point.

       Where Section 2.5.1.4 indicates that it is appropriate to use the
       RFC 2198 [2] audio redundancy mechanism to carry retransmissions
       of final event reports, this mechanism MAY also be used to extend
       the number of final report retransmissions.  This is done by
       using more than two levels of redundancy when necessary.  The use
       of RFC 2198 helps to mitigate the extra bandwidth demands that
       would be imposed simply by retransmitting final event packets
       more than three times.

   These two redundancy mechanisms clearly address requirement a) in the
   previous section.  They also help meet requirement c), to the extent
   that the redundant packets arrive before playout of the events they
   report is due to expire.  They are not helpful in meeting the other
   requirements, although they do not directly cause impairments
   themselves in the way that a large jitter buffer increases end-to-end
   delay.

   The playout algorithm is an additional mechanism for meeting the
   performance requirements.  In particular, using the second algorithm
   in Section 2.5.2.2 will meet requirement d) of the previous section
   by preventing gaps in playout, but at the potential cost of increases
   in duration (requirement c)).

   Finally, there is an interaction between the packetization period
   used by a sender, the playout delay used by the receiver, and the
   vulnerability of an event flow to packet losses.  Assuming packet
   losses are independent, a shorter packetization interval means that
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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.

2.6.3. Adjusting to Congestion

So far, the discussion has been about meeting performance requirements. However, there is also the question of whether applications of events can adapt to congestion to the point that they reduce their demands on the networks during congestion. In theory this can be done for events by increasing the packetization interval, so that fewer packets are sent per second. This has to be accompanied by an increased playout delay at the receiving end. Coordination between the two ends for this purpose is an interesting issue in itself. If it is done, however, such an action implies a one-time gap or extended playout of an event when the packetization interval is first extended, as well as increased end-to-end delay during the whole period of increased playout delay. The benefit from such a measure varies primarily depending on the average duration of the events being handled. In the worst case, as a first example shows, the reduction in aggregate bandwidth usage due to an increased packetization interval may be quite modest. Suppose the average event duration is 3.33 ms (V.21 bits, for instance). Suppose further that four transmissions in total are required for a given event report to meet the loss objective. Table 1 shows the impact of varying packetization intervals on the aggregate bit rate of the media stream. +--------------------+-----------+---------------+------------------+ | Packetization | Packets/s | IP Packet | Total IP Bit | | Interval (ms) | | Size (bits) | Rate (bits/s) | +--------------------+-----------+---------------+------------------+ | 50 | 20 | 2440 | 48800 | | 33.3 | 30 | 1800 | 54000 | | 25 | 40 | 1480 | 59200 | | 20 | 50 | 1288 | 64400 | +--------------------+-----------+---------------+------------------+ Table 1: Data Rate at the IP Level versus Packetization Interval (three retransmissions, 3.33 ms per event)
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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 Section 2.5.2.2 must be used in this application.)
   The reduction in number of packets per second with longer
   packetization periods is countered by the increase in packet size due
   to the increase in number of events per packet.

   For events of longer duration, the reduction in bandwidth is more
   proportional to the increase in packetization interval.  The loss of
   final event reports may also be less critical, so that lower
   redundancy levels are acceptable.  Table 2 shows similar data to
   Table 1, but assuming 70-ms events separated by 50 ms of silence (as
   in an idealized DTMF-based text messaging session) with only the
   basic two retransmissions for event completions.

   +--------------------+-----------+---------------+------------------+
   | Packetization      | Packets/s |     IP Packet |     Total IP Bit |
   | Interval (ms)      |           |   Size (bits) |    Rate (bits/s) |
   +--------------------+-----------+---------------+------------------+
   | 50                 |        20 |       448/520 |            10040 |
   | 33.3               |        30 |       448/520 |            14280 |
   | 25                 |        40 |       448/520 |            18520 |
   | 20                 |        50 |           448 |            22400 |
   +--------------------+-----------+---------------+------------------+

     Table 2: Data Rate at the IP Level versus Packetization Interval
       (two retransmissions, 70 ms per event, 50 ms between events)

   In the third column of the table, the packet size is 448 bits when
   only one event is being reported and 520 bits when the previous event
   is also included.  No more than one level of redundancy is needed up
   to a packetization interval of 50 ms, although at that point most
   packets are reporting two events.  Longer intervals require a second
   level of redundancy in at least some packets.



(page 23 continued on part 2)

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