RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
RFC 2833

Network Working Group H. Schulzrinne
Request for Comments: 2833 Columbia University
Category: Standards Track S. Petrack
 MetaTel
 May 2000
 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 Internet Society (2000). All Rights Reserved.
Abstract
 This memo describes how to carry dual-tone multifrequency (DTMF)
 signaling, other tone signals and telephony events in RTP packets.
1 Introduction
 This memo defines two payload formats, one for carrying dual-tone
 multifrequency (DTMF) digits, other line and trunk signals (Section
 3), and a second one for general multi-frequency tones in RTP [1]
 packets (Section 4). Separate RTP payload formats are desirable since
 low-rate voice codecs cannot be guaranteed to reproduce these tone
 signals accurately enough for automatic recognition. Defining
 separate payload formats also permits higher redundancy while
 maintaining a low bit rate.
 The payload formats described here may be useful in at least three
 applications: DTMF handling for gateways and end systems, as well as
 "RTP trunks". In the first application, 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 that low bit-rate codecs
 like G.723.1 render DTMF tones unintelligible. Secondly, an Internet
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RFC 2833 Tones May 2000
 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.
 In the "RTP trunk" application, RTP is used to replace a normal
 circuit-switched trunk between two nodes. This is particularly of
 interest in a telephone network that is still mostly circuit-
 switched. In this case, each end of the RTP trunk encodes audio
 channels into the appropriate encoding, such as G.723.1 or G.729.
 However, this encoding process destroys in-band signaling information
 which is carried using the least-significant bit ("robbed bit
 signaling") and may also interfere with in-band signaling tones, such
 as the MF digit tones. In addition, tone properties such as the phase
 reversals in the ANSam tone, will not survive speech coding. Thus,
 the gateway needs to remove the in-band signaling information from
 the bit stream. It can now either carry it out-of-band in a signaling
 transport mechanism yet to be defined, or it can use the mechanism
 described in this memorandum. (If the two trunk end points are within
 reach of the same media gateway controller, the media gateway
 controller can also handle the signaling.) Carrying it in-band may
 simplify the time synchronization between audio packets and the tone
 or signal information. This is particularly relevant where duration
 and timing matter, as in the carriage of DTMF signals.
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 [2] and
 indicate requirement levels for compliant implementations.
2 Events vs. Tones
 A gateway has two options for handling DTMF digits and events. First,
 it can simply measure the frequency components of the voice band
 signals and transmit this information to the RTP receiver (Section
 4). In this mode, the gateway makes no attempt to discern the meaning
 of the tones, but simply distinguishes tones from speech signals.
 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. (There is at least one tone, the ANSam tone [3] used for
 indicating data transmission over voice lines, that makes use of
 periodic phase reversals.)
 As a second option, a gateway can recognize the tones and translate
 them into a name, such as ringing or busy tone. The receiver then
 produces a tone signal or other indication appropriate to the signal.
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RFC 2833 Tones May 2000
 Generally, since the recognition of signals 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 signaling information that generates the tones and thus
 can generate the RTP packet immediately, without the detour through
 acoustic signals.
 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.
 ISDN terminals may generate dial tone locally and then send a Q.931
 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
 messages, and provides dial tone over the B-channel. The terminal can
 either use the audio signal on the B-channel or can 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
 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.
 Gateways which send signaling events via RTP MAY send both named
 signals (Section 3) and the tone representation (Section 4) as a
 single RTP session, using the redundancy mechanism defined in Section
 3.7 to interleave the two representations. It is generally a good
 idea to send both, since it allows the receiver to choose the
 appropriate rendering.
 If a gateway cannot present a tone representation, it SHOULD send the
 audio tones as regular RTP audio packets (e.g., as payload format
 PCMU), in addition to the named signals.
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3 RTP Payload Format for Named Telephone Events
3.1 Introduction
 The payload format for named telephone events described below is
 suitable for both gateway and end-to-end scenarios. In the gateway
 scenario, an Internet telephony gateway connecting a packet voice
 network to the PSTN recreates the DTMF tones or other telephony
 events 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.
 DTMF digits and named telephone events are carried as part of the
 audio stream, and MUST use the same sequence number and time-stamp
 base as the regular audio channel to simplify the generation of audio
 waveforms at a gateway. The default clock frequency is 8,000 Hz, but
 the clock frequency can be redefined when assigning the dynamic
 payload type.
 The payload format described here achieves a higher redundancy even
 in the case of sustained packet loss than the method proposed for the
 Voice over Frame Relay Implementation Agreement [4].
 If an end system is directly connected to the Internet and does not
 need to generate tone signals again, time alignment and power levels
 are not relevant. These systems rely on PSTN gateways or Internet end
 systems to generate DTMF events 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, such as the IVR example mentioned earlier, it may be
 preferable to use a reliable control protocol rather than RTP
 packets. In those circumstances, this payload format would not be
 used.
3.2 Simultaneous Generation of Audio and Events
 A source MAY send events and coded audio packets for the same time
 instants, using events as the redundant encoding for the audio
 stream, or it MAY block outgoing audio while event tones are active
 and only send named events as both the primary and redundant
 encodings.
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 Note that a period covered by an encoded tone may overlap in time
 with a period of audio encoded by other means. This is likely to
 occur at the onset of a tone and is necessary to avoid possible
 errors in the interpretation of the reproduced tone at the remote
 end. Implementations supporting this payload format must be prepared
 to handle the overlap. It is RECOMMENDED that gateways only render
 the encoded tone 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.
3.3 Event Types
 This payload format is used for five different types of signals:
 o DTMF tones (Section 3.10);
 o fax-related tones (Section 3.11);
 o standard subscriber line tones (Section 3.12);
 o country-specific subscriber line tones (Section 3.13) and;
 o trunk events (Section 3.14).
 A compliant implementation MUST support the events listed in Table 1
 with the exception of "flash". If it uses some other, out-of-band
 mechanism for signaling line conditions, it does not have to
 implement the other events.
 In some cases, an implementation may simply ignore certain events,
 such as fax tones, that do not make sense in a particular
 environment. Section 3.9 specifies how an implementation can use the
 SDP "fmtp" parameter within an SDP description to indicate its
 inability to understand a particular event or range of events.
 Depending on the available user interfaces, an implementation MAY
 render all tones in Table 5 the same or, preferably, use the tones
 conveyed by the concurrent "tone" payload or other RTP audio payload.
 Alternatively, it could provide a textual representation.
 Note that end systems that emulate telephones only need to support
 the events described in Sections 3.10 and 3.12, while systems that
 receive trunk signaling need to implement those in Sections 3.10,
 3.11, 3.12 and 3.14, since MF trunks also carry most of the "line"
 signals. Systems that do not support fax or modem functionality do
 not need to render fax-related events described in Section 3.11.
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 The RTP payload format is designated as "telephone-event", the MIME
 type as "audio/telephone-event". The default timestamp rate is 8000
 Hz, but other rates may be defined. In accordance with current
 practice, this payload format does not have a static payload type
 number, but uses a RTP payload type number established dynamically
 and out-of-band.
3.4 Use of RTP Header Fields
 Timestamp: The RTP timestamp reflects the measurement point for
 the current packet. The event duration described in Section
 3.5 extends forwards from that time. The receiver calculates
 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.
 Marker bit: The RTP marker bit indicates the beginning of a new
 event.
3.5 Payload Format
 The payload format is shown in Fig. 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
 events: The events are encoded as shown in Sections 3.10 through
 3.14.
 volume: 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. The range of valid DTMF is from 0 to -36 dBm0 (must
 accept); lower than -55 dBm0 must be rejected (TR-TSY-000181,
 ITU-T Q.24A). Thus, larger values denote lower volume. This
 value is defined only for DTMF digits. For other events, it
 is set to zero by the sender and is ignored by the receiver.
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 duration: Duration of this digit, in timestamp units. Thus, the
 event 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.
 For a sampling rate of 8000 Hz, this field is sufficient to
 express event durations of up to approximately 8 seconds.
 E: If set to a value of one, the "end" bit indicates that this
 packet contains the end of the event. Thus, the duration
 parameter above measures the complete duration of the event.
 A sender MAY delay setting the end bit until retransmitting
 the last packet for a tone, rather than on its first
 transmission. This avoids having to wait to detect whether
 the tone has indeed ended.
 Receiver implementations MAY use different algorithms to
 create tones, including the two described here. In the first,
 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 it receives a packet with the
 "E" bit set, the next tone, distinguished by a different
 timestamp value or a given time period elapses. This is more
 robust against packet loss, but may extend the tone 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". 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.
 R: This field is reserved for future use. The sender MUST set it
 to zero, the receiver MUST ignore it.
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3.6 Sending Event Packets
 An audio source SHOULD start transmitting event packets as soon as it
 recognizes an event and every 50 ms thereafter or the packet interval
 for the audio codec used for this session, if known. (The sender does
 not need to maintain precise time intervals between event packets in
 order to maintain precise inter-event times, since the timing
 information is contained in the timestamp.)
 Q.24 [5], Table A-1, indicates that all administrations surveyed
 use a minimum signal duration of 40 ms, with signaling velocity
 (tone and pause) of no less than 93 ms.
 If an event continues for more than one period, the source generating
 the events should send a new event packet with the RTP timestamp
 value corresponding to the beginning of the event and the duration of
 the event increased correspondingly. (The RTP sequence number is
 incremented by one for each packet.) If there has been no new event
 in the last interval, the event SHOULD be retransmitted three times
 or until the next event is recognized. This ensures that the duration
 of the event can be recognized correctly even if the last packet for
 an event is lost.
 DTMF digits and events are sent incrementally to avoid having the
 receiver wait for the completion of the event. Since some tones
 are two seconds long, this would incur a substantial delay. The
 transmitter does not know if event length is important and thus
 needs to transmit immediately and incrementally. If the receiver
 application does not care about event length, the incremental
 transmission mechanism avoids delay. Some applications, such as
 gateways into the PSTN, care about both delays and event duration.
3.7 Reliability
 During an event, the RTP event payload format provides incremental
 updates on the event. The error resiliency depends on the playout
 delay at the receiver. For example, for a playout delay of 120 ms and
 a packet gap of 50 ms, two packets in a row can get lost without
 causing a gap in the tones generated at the receiver.
 The audio redundancy mechanism described in RFC 2198 [6] MAY be used
 to recover from packet loss across events. The effective data rate is
 r times 64 bits (32 bits for the redundancy header and 32 bits for
 the telephone-event payload) every 50 ms or r times 1280 bits/second,
 where r is the number of redundant events carried in each packet. The
 value of r is an implementation trade-off, with a value of 5
 suggested.
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 The timestamp offset in this redundancy scheme has 14 bits, so
 that it allows a single packet to "cover" 2.048 seconds of
 telephone events at a sampling rate of 8000 Hz. Including the
 starting time of previous events allows precise reconstruction of
 the tone sequence at a gateway. The scheme is resilient to
 consecutive packet losses spanning this interval of 2.048 seconds
 or r digits, whichever is less. Note that for previous digits,
 only an average loudness can be represented.
 An encoder MAY treat the event payload as a highly-compressed version
 of the current audio frame. In that mode, each RTP packet during an
 event would contain the current audio codec rendition (say, G.723.1
 or G.729) of this digit as well as the representation described in
 Section 3.5, plus any previous events seen earlier.
 This approach allows dumb gateways that do not understand this
 format to function. See also the discussion in Section 1.
3.8 Example
 A typical RTP packet, where the user is just dialing the last digit
 of the DTMF sequence "911". The first digit was 200 ms long (1600
 timestamp units) and started at time 0, the second digit lasted 250
 ms (2000 timestamp units) and started at time 800 ms (6400 timestamp
 units), the third digit was pressed at time 1.4 s (11,200 timestamp
 units) and the packet shown was sent at 1.45 s (11,600 timestamp
 units). The frame duration is 50 ms. To make the parts recognizable,
 the figure below ignores byte alignment. Timestamp and sequence
 number are assumed to have been zero at the beginning of the first
 digit. In this example, the dynamic payload types 96 and 97 have been
 assigned for the redundancy mechanism and the telephone event
 payload, respectively.
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3.9 Indication of Receiver Capabilities using SDP
 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| 96 | 28 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | timestamp |
 | 11200 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | synchronization source (SSRC) identifier |
 | 0x5234a8 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |F| block PT | timestamp offset | block length |
 |1| 97 | 11200 | 4 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |F| block PT | timestamp offset | block length |
 |1| 97 | 11200 -ひく 6400 =わ 4800 | 4 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |F| Block PT |
 |0| 97 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | digit |E R| volume | duration |
 | 9 |1 0| 7 | 1600 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | digit |E R| volume | duration |
 | 1 |1 0| 10 | 2000 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | digit |E R| volume | duration |
 | 1 |0 0| 20 | 400 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 Figure 2: Example RTP packet after dialing "911"
 Receivers MAY indicate which named events they can handle, for
 example, by using the Session Description Protocol (RFC 2327 [7]).
 The payload formats use the following fmtp format to list the event
 values that they can receive:
 a=fmtp:<format> <list of values>
 The list of values consists of comma-separated elements, which can be
 either a single decimal number or two decimal numbers separated by a
 hyphen (dash), where the second number is larger than the first. No
 whitespace is allowed between numbers or hyphens. The list does not
 have to be sorted.
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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, it would include the following
 description in its SDP message:
 a=fmtp:100 0-15,66,70
 Since all implementations MUST be able to receive events 0 through
 15, listing these events in the a=fmtp line is OPTIONAL.
 The corresponding MIME parameter is "events", so that the following
 sample media type definition corresponds to the SDP example above:
 audio/telephone-event;events="0-11,66,67";rate="8000"
3.10 DTMF Events
 Table 1 summarizes the DTMF-related named events within the
 telephone-event payload format.
 Event encoding (decimal)
 _________________________
 0--9 0--9
 * 10
 # 11
 A--D 12--15
 Flash 16
 Table 1: DTMF named events
3.11 Data Modem and Fax Events
 Table 3.11 summarizes the events and tones that can appear on a
 subscriber line serving a fax machine or modem. The tones are
 described below, with additional detail in Table 7.
 ANS: This 2100 +/- 15 Hz tone is used to disable echo
 suppression for data transmission [8,9]. For fax machines,
 Recommendation T.30 [9] refers to this tone as called
 terminal identification (CED) answer tone.
 /ANS: This is the same signal as ANS, except that it reverses
 phase at an interval of 450 +/- 25 ms. It disables both
 echo cancellers and echo suppressors. (In the ITU
 Recommendation V.25 [8], this signal is rendered as ANS
 with a bar on top.)
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 ANSam: The modified answer tone (ANSam) [3] is a sinewave signal
 at 2100 +/- 1 Hz without phase reversals, amplitude-modulated
 by a sinewave at 15 +/- 0.1 Hz. This tone is sent by modems
 if network echo canceller disabling is not required.
 /ANSam: The modified answer tone with phase reversals (ANSam) [3]
 is a sinewave signal at 2100 +/- 1 Hz with phase reversals at
 intervals of 450 +/- 25 ms, amplitude-modulated by a sinewave
 at 15 +/- 0.1 Hz. This tone [10,8] is sent by modems [11] and
 faxes to disable echo suppressors.
 CNG: After dialing the called fax machine's telephone number (and
 before it answers), the calling Group III fax machine
 (optionally) begins sending a CalliNG tone (CNG) consisting
 of an interrupted tone of 1100 Hz. [9]
 CRdi: Capabilities Request (CRd), initiating side, [12] is a
 dual-tone signal with tones at 1375 Hz and 2002 Hz for 400
 ms, followed by a single tone at 1900 Hz for 100 ms. "This
 signal requests the remote station transition from telephony
 mode to an information transfer mode and requests the
 transmission of a capabilities list message by the remote
 station. In particular, CRdi is sent by the initiating
 station during the course of a call, or by the calling
 station at call establishment in response to a CRe or MRe."
 CRdr: CRdr is the response tone to CRdi (see above). It consists
 of a dual-tone signal with tones at 1529 Hz and 2225 Hz for
 400 ms, followed by a single tone at 1900 Hz for 100 ms.
 CRe: Capabilities Request (CRe) [12] is a dual-tone signal with
 tones at tones at 1375 Hz and 2002 Hz for 400 ms, followed by
 a single tone at 400 Hz for 100 ms. "This signal requests the
 remote station transition from telephony mode to an
 information transfer mode and requests the transmission of a
 capabilities list message by the remote station. In
 particular, CRe is sent by an automatic answering station at
 call establishment."
 CT: "The calling tone [8] consists of a series of interrupted
 bursts of binary 1 signal or 1300 Hz, on for a duration of
 not less than 0.5 s and not more than 0.7 s and off for a
 duration of not less than 1.5 s and not more than 2.0 s."
 Modems not starting with the V.8 call initiation tone often
 use this tone.
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 ESi: Escape Signal (ESi) [12] is a dual-tone signal with tones at
 1375 Hz and 2002 Hz for 400 ms, followed by a single tone at
 980 Hz for 100 ms. "This signal requests the remote station
 transition from telephony mode to an information transfer
 mode. signal ESi is sent by the initiating station."
 ESr: Escape Signal (ESr) [12] is a dual-tone signal with tones at
 1529 Hz and 2225 Hz for 400 ms, followed by a single tone at
 1650 Hz for 100 ms. Same as ESi, but sent by the responding
 station.
 MRdi: Mode Request (MRd), initiating side, [12] is a dual-tone
 signal with tones at 1375 Hz and 2002 Hz for 400 ms followed
 by a single tone at 1150 Hz for 100 ms. "This signal requests
 the remote station transition from telephony mode to an
 information transfer mode and requests the transmission of a
 mode select message by the remote station. In particular,
 signal MRd is sent by the initiating station during the
 course of a call, or by the calling station at call
 establishment in response to an MRe." [12]
 MRdr: MRdr is the response tone to MRdi (see above). It consists
 of a dual-tone signal with tones at 1529 Hz and 2225 Hz for
 400 ms, followed by a single tone at 1150 Hz for 100 ms.
 MRe: Mode Request (MRe) [12] is a dual-tone signal with tones at
 1375 Hz and 2002 Hz for 400 ms, followed by a single tone at
 650 Hz for 100 ms. "This signal requests the remote station
 transition from telephony mode to an information transfer
 mode and requests the transmission of a mode select message
 by the remote station. In particular, signal MRe is sent by
 an automatic answering station at call establishment." [12]
 V.21: V.21 describes a 300 b/s full-duplex modem that employs
 frequency shift keying (FSK). It is used by Group 3 fax
 machines to exchange T.30 information. The calling transmits
 on channel 1 and receives on channel 2; the answering modem
 transmits on channel 2 and receives on channel 1. Each bit
 value has a distinct tone, so that V.21 signaling comprises a
 total of four distinct tones.
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 In summary, procedures in Table 2 are used.
 Procedure indications
 ___________________________________________________
 V.25 and V.8 ANS
 V.25, echo canceller disabled ANS, /ANS, ANS, /ANS
 V.8 ANSam
 V.8, echo canceller disabled /ANSam
 Table 2: Use of ANS, ANSam and /ANSam in V.x recommendations
 Event encoding (decimal)
 ___________________________________________________
 Answer tone (ANS) 32
 /ANS 33
 ANSam 34
 /ANSam 35
 Calling tone (CNG) 36
 V.21 channel 1, "0" bit 37
 V.21 channel 1, "1" bit 38
 V.21 channel 2, "0" bit 39
 V.21 channel 2, "1" bit 40
 CRdi 41
 CRdr 42
 CRe 43
 ESi 44
 ESr 45
 MRdi 46
 MRdr 47
 MRe 48
 CT 49
 Table 3: Data and fax named events
3.12 Line Events
 Table 4 summarizes the events and tones that can appear on a
 subscriber line.
 ITU Recommendation E.182 [13] defines when certain tones should be
 used. It defines the following standard tones that are heard by the
 caller:
 Dial tone: The exchange is ready to receive address information.
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 PABX internal dial tone: The PABX is ready to receive address
 information.
 Special dial tone: Same as dial tone, but the caller's line is
 subject to a specific condition, such as call diversion or a
 voice mail is available (e.g., "stutter dial tone").
 Second dial tone: The network has accepted the address
 information, but additional information is required.
 Ring: This named signal event causes the recipient to generate an
 alerting signal ("ring"). The actual tone or other indication
 used to render this named event is left up to the receiver.
 (This differs from the ringing tone, below, heard by the
 caller
 Ringing tone: The call has been placed to the callee and a calling
 signal (ringing) is being transmitted to the callee. This
 tone is also called "ringback".
 Special ringing tone: A special service, such as call forwarding
 or call waiting, is active at the called number.
 Busy tone: The called telephone number is busy.
 Congestion tone: Facilities necessary for the call are temporarily
 unavailable.
 Calling card service tone: The calling card service tone consists
 of 60 ms of the sum of 941 Hz and 1477 Hz tones (DTMF '#'),
 followed by 940 ms of 350 Hz and 440 Hz (U.S. dial tone),
 decaying exponentially with a time constant of 200 ms.
 Special information tone: The callee cannot be reached, but the
 reason is neither "busy" nor "congestion". This tone should
 be used before all call failure announcements, for the
 benefit of automatic equipment.
 Comfort tone: The call is being processed. This tone may be used
 during long post-dial delays, e.g., in international
 connections.
 Hold tone: The caller has been placed on hold.
 Record tone: The caller has been connected to an automatic
 answering device and is requested to begin speaking.
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 Caller waiting tone: The called station is busy, but has call
 waiting service.
 Pay tone: The caller, at a payphone, is reminded to deposit
 additional coins.
 Positive indication tone: The supplementary service has been
 activated.
 Negative indication tone: The supplementary service could not be
 activated.
 Off-hook warning tone: The caller has left the instrument off-hook
 for an extended period of time.
 The following tones can be heard by either calling or called party
 during a conversation:
 Call waiting tone: Another party wants to reach the subscriber.
 Warning tone: The call is being recorded. This tone is not
 required in all jurisdictions.
 Intrusion tone: The call is being monitored, e.g., by an operator.
 CPE alerting signal: A tone used to alert a device to an arriving
 in-band FSK data transmission. A CPE alerting signal is a
 combined 2130 and 2750 Hz tone, both with tolerances of 0.5%
 and a duration of 80 to. 80 ms. The CPE alerting signal is
 used with ADSI services and Call Waiting ID services [14].
 The following tones are heard by operators:
 Payphone recognition tone: The person making the call or being
 called is using a payphone (and thus it is ill-advised to
 allow collect calls to such a person).
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 Event encoding (decimal)
 _____________________________________________
 Off Hook 64
 On Hook 65
 Dial tone 66
 PABX internal dial tone 67
 Special dial tone 68
 Second dial tone 69
 Ringing tone 70
 Special ringing tone 71
 Busy tone 72
 Congestion tone 73
 Special information tone 74
 Comfort tone 75
 Hold tone 76
 Record tone 77
 Caller waiting tone 78
 Call waiting tone 79
 Pay tone 80
 Positive indication tone 81
 Negative indication tone 82
 Warning tone 83
 Intrusion tone 84
 Calling card service tone 85
 Payphone recognition tone 86
 CPE alerting signal (CAS) 87
 Off-hook warning tone 88
 Ring 89
 Table 4: E.182 line events
3.13 Extended Line Events
 Table 5 summarizes country-specific events and tones that can appear
 on a subscriber line.
3.14 Trunk Events
 Table 6 summarizes the events and tones that can appear on a trunk.
 Note that trunk can also carry line events (Section 3.12), as MF
 signaling does not include backward signals [15].
 ABCD transitional: 4-bit signaling used by digital trunks. For N-
 state signaling, the first N values are used.
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 Event encoding (decimal)
 ___________________________________________________
 Acceptance tone 96
 Confirmation tone 97
 Dial tone, recall 98
 End of three party service tone 99
 Facilities tone 100
 Line lockout tone 101
 Number unobtainable tone 102
 Offering tone 103
 Permanent signal tone 104
 Preemption tone 105
 Queue tone 106
 Refusal tone 107
 Route tone 108
 Valid tone 109
 Waiting tone 110
 Warning tone (end of period) 111
 Warning Tone (PIP tone) 112
 Table 5: Country-specific Line events
 The T1 ESF (extended super frame format) allows 2, 4, and 16
 state signaling bit options. These signaling bits are named
 A, B, C, and D. Signaling information is sent as robbed bits
 in frames 6, 12, 18, and 24 when using ESF T1 framing. A D4
 superframe only transmits 4-state signaling with A and B
 bits. On the CEPT E1 frame, all signaling is carried in
 timeslot 16, and two channels of 16-state (ABCD) signaling
 are sent per frame.
 Since this information is a state rather than a changing
 signal, implementations SHOULD use the following triple-
 redundancy mechanism, similar to the one specified in ITU-T
 Rec. I.366.2 [16], Annex L. At the time of a transition, the
 same ABCD information is sent 3 times at an interval of 5 ms.
 If another transition occurs during this time, then this
 continues. After a period of no change, the ABCD information
 is sent every 5 seconds.
 Wink: A brief transition, typically 120-290 ms, from on-hook
 (unseized) to off-hook (seized) and back to onhook, used by
 the incoming exchange to signal that the call address
 signaling can proceed.
 Incoming seizure: Incoming indication of call attempt (off-hook).
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 Event encoding (decimal)
 __________________________________________________
 MF 0... 9 128...137
 MF K0 or KP (start-of-pulsing) 138
 MF K1 139
 MF K2 140
 MF S0 to ST (end-of-pulsing) 141
 MF S1... S3 142...143
 ABCD signaling (see below) 144...159
 Wink 160
 Wink off 161
 Incoming seizure 162
 Seizure 163
 Unseize circuit 164
 Continuity test 165
 Default continuity tone 166
 Continuity tone (single tone) 167
 Continuity test send 168
 Continuity verified 170
 Loopback 171
 Old milliwatt tone (1000 Hz) 172
 New milliwatt tone (1004 Hz) 173
 Table 6: Trunk events
 Seizure: Seizure by answering exchange, in response to outgoing
 seizure.
 Unseize circuit: Transition of circuit from off-hook to on-hook at
 the end of a call.
 Wink off: A brief transition, typically 100-350 ms, from off-hook
 (seized) to on-hook (unseized) and back to off-hook (seized).
 Used in operator services trunks.
 Continuity tone send: A tone of 2010 Hz.
 Continuity tone detect: A tone of 2010 Hz.
 Continuity test send: A tone of 1780 Hz is sent by the calling
 exchange. If received by the called exchange, it returns a
 "continuity verified" tone.
 Continuity verified: A tone of 2010 Hz. This is a response tone,
 used in dual-tone procedures.
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4 RTP Payload Format for Telephony Tones
4.1 Introduction
 As an alternative to describing tones and events by name, as
 described in Section 3, 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, across
 all countries, these tones share a number of characteristics [17]:
 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 Tones for telephony events are in the range of 25 (ringing tone
 in Angola) to 1800 Hz. CED is the highest used tone at 2100 Hz.
 The telephone frequency range is limited to 3,400 Hz. (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. (These fractional
 frequencies appear to be derived from older AC power grid
 frequencies.)
 o Tones that are not continuous have durations of less than four
 seconds.
 o ITU Recommendation E.180 [18] notes that different telephone
 companies require a tone accuracy of between 0.5 and 1.5%. The
 Recommendation suggests a frequency tolerance of 1%.
4.2 Examples of Common Telephone Tone Signals
 As an aid to the implementor, Table 7 summarizes some common tones.
 The rows labeled "ITU ..." refer to the general recommendation of
 Recommendation E.180 [18]. Note that there are no specific guidelines
 for these tones. In the table, the symbol "+" indicates addition of
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 the tones, without modulation, while "*" indicates amplitude
 modulation. The meaning of some of the tones is described in Section
 3.12 or Section 3.11 (for V.21).
 Tone name frequency on period off period
 ______________________________________________________
 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 "0" bit, ch. 1 1180 0.00333
 V.21 "1" bit, ch. 1 980 0.00333
 V.21 "0" bit, ch. 2 1850 0.00333
 V.21 "1" bit, ch. 2 1650 0.00333
 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
 U.S. busy tone 480+620 0.5 0.5
 ______________________________________________________
 ITU congestion tone 425
 U.S. congestion tone 480+620 0.25 0.25
 Table 7: Examples of telephony tones
4.3 Use of RTP Header Fields
 Timestamp: The RTP timestamp reflects the measurement point for
 the current packet. The event duration described in Section
 3.5 extends forwards from that time.
4.4 Payload Format
 Based on the characteristics described above, this document defines
 an RTP payload format called "tone" that can represent tones
 consisting of one or more frequencies. (The corresponding MIME type
 is "audio/tone".) The default timestamp rate is 8,000 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 a RTP payload type number
 established dynamically and out-of-band.
 It is shown in Fig. 3.
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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
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | 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 3: 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.
 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.
 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.
 The tone begins at the instant identified by the RTP
 timestamp and lasts for the duration value.
 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,
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 which exceeds the range of telephone systems. A value of zero
 indicates silence. A single tone can contain any number of
 frequencies.
 R: This field is reserved for future use. The sender MUST set it
 to zero, the receiver MUST ignore it.
4.5 Reliability
 This payload format uses the reliability mechanism described in
 Section 3.7.
5 Combining Tones and Named Events
 The payload formats in Sections 3 and 4 can be combined into a single
 payload using the method specified in RFC 2198. Fig. 4 shows an
 example. In that example, the RTP packet combines two "tone" and one
 "telephone-event" payloads. The payload types are chosen arbitrarily
 as 97 and 98, respectively, with a sample rate of 8000 Hz. Here, the
 redundancy format has the dynamic payload type 96.
 The packet represents a snapshot of U.S. ringing tone, 1.5 seconds
 (12,000 timestamp units) into the second "on" part of the 2.0/4.0
 second cadence, i.e., a total of 7.5 seconds (60,000 timestamp units)
 into the ring cycle. The 440 + 480 Hz tone of this second cadence
 started at RTP timestamp 48,000. Four seconds of silence preceded it,
 but since RFC 2198 only has a fourteen-bit offset, only 2.05 seconds
 (16383 timestamp units) can be represented. Even though the tone
 sequence is not complete, the sender was able to determine that this
 is indeed ringback, and thus includes the corresponding named event.
6 MIME Registration
6.1 audio/telephone-event
 MIME media type name: audio
 MIME subtype name: telephone-event
 Required parameters: none.
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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 |P|X| CC |M| PT | sequence number |
 | 2 |0|0| 0 |0| 96 | 31 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | timestamp |
 | 48000 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | synchronization source (SSRC) identifier |
 | 0x5234a8 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |F| block PT | timestamp offset | block length |
 |1| 98 | 16383 | 4 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |F| block PT | timestamp offset | block length |
 |1| 97 | 16383 | 8 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |F| Block PT |
 |0| 97 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | event=ring |0|0| volume=0 | duration=28383 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | modulation=0 |0| volume=63 | duration=16383 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |0 0 0 0| frequency=0 |0 0 0 0| frequency=0 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 | modulation=0 |0| volume=5 | duration=12000 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 |0 0 0 0| frequency=440 |0 0 0 0| frequency=480 |
 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
 Figure 4: Combining tones and events in a single RTP packet
 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 either be a
 single integer or two integers separated by a hyphen. No
 white space is allowed in the argument. The integers
 designate the event numbers supported by the implementation.
 All implementations MUST support events 0 through 15, so that
 the parameter can be omitted if the implementation only
 supports these events.
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RFC 2833 Tones May 2000
 The "rate" parameter describes the sampling rate, in Hertz.
 The number is written as a floating point number or as an
 integer. If omitted, the default value is 8000 Hz.
 Encoding considerations: This type is only defined for transfer
 via RTP [1].
 Security considerations: See the "Security Considerations"
 (Section 7) section 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:
 1. Magic number(s): N/A
 2. File extension(s): N/A
 3. Macintosh file type code: N/A
6.2 audio/tone
 MIME media type name: audio
 MIME subtype name: tone
 Required parameters: none
 Optional parameters: The "rate" parameter describes the sampling
 rate, in Hertz. The number is written as a floating point
 number or as an integer. If omitted, the default value is
 8000 Hz.
 Encoding considerations: This type is only defined for transfer
 via RTP [1].
 Security considerations: See the "Security Considerations"
 (Section 7) section in this document.
 Interoperability considerations: none
 Published specification: This document.
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 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:
 1. Magic number(s): N/A
 2. File extension(s): N/A
 3. Macintosh file type code: N/A
7 Security Considerations
 RTP packets using the payload format defined in this specification
 are subject to the security considerations discussed in the RTP
 specification (RFC 1889 [1]), and any appropriate RTP profile (for
 example RFC 1890 [19]).This implies that confidentiality of the media
 streams is achieved by encryption. Because the data compression used
 with this payload format is applied end-to-end, encryption may be
 performed after compression so there is no conflict between the two
 operations.
 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.
 In older networks employing in-band signaling and lacking appropriate
 tone filters, the tones in Section 3.14 may be used to commit toll
 fraud.
 Additional security considerations are described in RFC 2198 [6].
8 IANA Considerations
 This document defines two new RTP payload formats, named telephone-
 event and tone, and associated Internet media (MIME) types,
 audio/telephone-event and audio/tone.
 Within the audio/telephone-event type, additional events MUST be
 registered with IANA. Registrations are subject to approval by the
 current chair of the IETF audio/video transport working group, or by
 an expert designated by the transport area director if the AVT group
 has closed.
Schulzrinne & Petrack Standards Track [Page 26]
RFC 2833 Tones May 2000
 The meaning of new events MUST be documented either as an RFC or an
 equivalent standards document produced by another standardization
 body, such as ITU-T.
9 Acknowledgements
 The suggestions of the Megaco working group are gratefully
 acknowledged. Detailed advice and comments were provided by Fred
 Burg, Steve Casner, Fatih Erdin, Bill Foster, Mike Fox, Gunnar
 Hellstrom, Terry Lyons, Steve Magnell, Vern Paxson and Colin Perkins.
10 Authors' Addresses
 Henning Schulzrinne
 Dept. of Computer Science
 Columbia University
 1214 Amsterdam Avenue
 New York, NY 10027
 USA
 EMail: schulzrinne@cs.columbia.edu
 Scott Petrack
 MetaTel
 45 Rumford Avenue
 Waltham, MA 02453
 USA
 EMail: scott.petrack@metatel.com
11 Bibliography
 [1] Schulzrinne, H., Casner, S., Frederick, R. and V. Jacobson,
 "RTP: A Transport Protocol for Real-Time Applications", RFC
 1889, January 1996.
 [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
 Levels", BCP 14, RFC 2119, March 1997.
 [3] International Telecommunication Union, "Procedures for starting
 sessions of data transmission over the public switched telephone
 network," Recommendation V.8, Telecommunication Standardization
 Sector of ITU, Geneva, Switzerland, Feb. 1998.
 [4] R. Kocen and T. Hatala, "Voice over frame relay implementation
 agreement", Implementation Agreement FRF.11, Frame Relay Forum,
 Foster City, California, Jan. 1997.
Schulzrinne & Petrack Standards Track [Page 27]
RFC 2833 Tones May 2000
 [5] International Telecommunication Union, "Multifrequency push-
 button signal reception," Recommendation Q.24, Telecommunication
 Standardization Sector of ITU, Geneva, Switzerland, 1988.
 [6] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M.,
 Bolot, J., Vega-Garcia, A. and S. Fosse-Parisis, "RTP Payload
 for Redundant Audio Data", RFC 2198, September 1997.
 [7] Handley M. and V. Jacobson, "SDP: Session Description Protocol",
 RFC 2327, April 1998.
 [8] International Telecommunication Union, "Automatic answering
 equipment and general procedures for automatic calling equipment
 on the general switched telephone network including procedures
 for disabling of echo control devices for both manually and
 automatically established calls," Recommendation V.25,
 Telecommunication Standardization Sector of ITU, Geneva,
 Switzerland, Oct. 1996.
 [9] International Telecommunication Union, "Procedures for document
 facsimile transmission in the general switched telephone
 network," Recommendation T.30, Telecommunication Standardization
 Sector of ITU, Geneva, Switzerland, July 1996.
 [10] International Telecommunication Union, "Echo cancellers,"
 Recommendation G.165, Telecommunication Standardization Sector
 of ITU, Geneva, Switzerland, Mar. 1993.
 [11] International Telecommunication Union, "A modem operating at
 data signaling rates of up to 33 600 bit/s for use on the
 general switched telephone network and on leased point-to-point
 2-wire telephone-type circuits," Recommendation V.34,
 Telecommunication Standardization Sector of ITU, Geneva,
 Switzerland, Feb. 1998.
 [12] International Telecommunication Union, "Procedures for the
 identification and selection of common modes of operation
 between data circuit-terminating equipments (DCEs) and between
 data terminal equipments (DTEs) over the public switched
 telephone network and on leased point-to-point telephone-type
 circuits," Recommendation V.8bis, Telecommunication
 Standardization Sector of ITU, Geneva, Switzerland, Sept. 1998.
 [13] International Telecommunication Union, "Application of tones and
 recorded announcements in telephone services," Recommendation
 E.182, Telecommunication Standardization Sector of ITU, Geneva,
 Switzerland, Mar. 1998.
Schulzrinne & Petrack Standards Track [Page 28]
RFC 2833 Tones May 2000
 [14] Bellcore, "Functional criteria for digital loop carrier
 systems," Technical Requirement TR-NWT-000057, Telcordia
 (formerly Bellcore), Morristown, New Jersey, Jan. 1993.
 [15] J. G. van Bosse, Signaling in Telecommunications Networks
 Telecommunications and Signal Processing, New York, New York:
 Wiley, 1998.
 [16] International Telecommunication Union, "AAL type 2 service
 specific convergence sublayer for trunking," Recommendation
 I.366.2, Telecommunication Standardization Sector of ITU,
 Geneva, Switzerland, Feb. 1999.
 [17] International Telecommunication Union, "Various tones used in
 national networks," Recommendation Supplement 2 to
 Recommendation E.180, Telecommunication Standardization Sector
 of ITU, Geneva, Switzerland, Jan. 1994.
 [18] International Telecommunication Union, "Technical
 characteristics of tones for telephone service," Recommendation
 Supplement 2 to Recommendation E.180, Telecommunication
 Standardization Sector of ITU, Geneva, Switzerland, Jan. 1994.
 [19] Schulzrinne, H., "RTP Profile for Audio and Video Conferences
 with Minimal Control", RFC 1890, January 1996.
Schulzrinne & Petrack Standards Track [Page 29]
RFC 2833 Tones May 2000
12 Full Copyright Statement
 Copyright (C) The Internet Society (2000). All Rights Reserved.
 This document and translations of it may be copied and furnished to
 others, and derivative works that comment on or otherwise explain it
 or assist in its implementation may be prepared, copied, published
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 included on all such copies and derivative works. However, this
 document itself may not be modified in any way, such as by removing
 the copyright notice or references to the Internet Society or other
 Internet organizations, except as needed for the purpose of
 developing Internet standards in which case the procedures for
 copyrights defined in the Internet Standards process must be
 followed, or as required to translate it into languages other than
 English.
 The limited permissions granted above are perpetual and will not be
 revoked by the Internet Society or its successors or assigns.
 This document and the information contained herein is provided on an
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 TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING
 BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION
 HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF
 MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Acknowledgement
 Funding for the RFC Editor function is currently provided by the
 Internet Society.
Schulzrinne & Petrack Standards Track [Page 30]