RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals (original) (raw)
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Obsoleted by: 4733, 4734 PROPOSED STANDARD
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](#ref-1 ""RTP: A Transport Protocol for Real-Time Applications"")] 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](#ref-2 ""Key words for use in RFCs to Indicate Requirement Levels"")] 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](#ref-3 ""Procedures for starting sessions of data transmission over the public switched telephone network,"")] 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.
Schulzrinne & Petrack Standards Track [Page 2]
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.
Schulzrinne & Petrack Standards Track [Page 3]
RFC 2833 Tones May 2000
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](#ref-4 ""Voice over frame relay implementation agreement"")].
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.
Schulzrinne & Petrack Standards Track [Page 4]
RFC 2833 Tones May 2000
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](#section-3.10));
o fax-related tones ([Section 3.11](#section-3.11));
o standard subscriber line tones ([Section 3.12](#section-3.12));
o country-specific subscriber line tones ([Section 3.13](#section-3.13)) and;
o trunk events ([Section 3.14](#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.
Schulzrinne & Petrack Standards Track [Page 5]
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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](#section-3.5)
[3.5](#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](#section-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.Schulzrinne & Petrack Standards Track [Page 6]
RFC 2833 Tones May 2000
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.Schulzrinne & Petrack Standards Track [Page 7]
RFC 2833 Tones May 2000
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](#ref-5 ""Multifrequency push- button signal reception,"")], 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](#ref-6 ""RTP Payload for Redundant Audio Data"")] 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.
Schulzrinne & Petrack Standards Track [Page 8]
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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](#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.
Schulzrinne & Petrack Standards Track [Page 9]
RFC 2833 Tones May 2000
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 - 6400 = 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](#ref-7 ""SDP: Session Description Protocol"")]). The payload formats use the following fmtp format to list the event values that they can receive:
a=fmtp:
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.
Schulzrinne & Petrack Standards Track [Page 10]
RFC 2833 Tones May 2000
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 events3.11 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](#ref-8 ""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,""),[9](#ref-9 ""Procedures for document facsimile transmission in the general switched telephone network,"")]. For fax machines,
Recommendation T.30 [[9](#ref-9 ""Procedures for document facsimile transmission in the general switched telephone network,"")] 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](#ref-8 ""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,"")], this signal is rendered as ANS
with a bar on top.)Schulzrinne & Petrack Standards Track [Page 11]
RFC 2833 Tones May 2000
ANSam: The modified answer tone (ANSam) [[3](#ref-3 ""Procedures for starting sessions of data transmission over the public switched telephone network,"")] 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](#ref-3 ""Procedures for starting sessions of data transmission over the public switched telephone network,"")]
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](#ref-10 ""Echo cancellers,""),[8](#ref-8 ""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,"")] is sent by modems [[11](#ref-11 ""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,"")] 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](#ref-9 ""Procedures for document facsimile transmission in the general switched telephone network,"")]
CRdi: Capabilities Request (CRd), initiating side, [[12](#ref-12 ""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,"")] 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](#ref-12 ""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,"")] 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](#ref-8 ""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,"")] 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.Schulzrinne & Petrack Standards Track [Page 12]
RFC 2833 Tones May 2000
ESi: Escape Signal (ESi) [[12](#ref-12 ""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,"")] 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](#ref-12 ""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,"")] 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](#ref-12 ""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,"")] 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](#ref-12 ""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,"")]
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](#ref-12 ""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,"")] 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](#ref-12 ""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,"")]
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.Schulzrinne & Petrack Standards Track [Page 13]
RFC 2833 Tones May 2000
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 events3.12 Line Events
Table 4 summarizes the events and tones that can appear on a subscriber line.
ITU Recommendation E.182 [[13](#ref-13 ""Application of tones and recorded announcements in telephone services,"")] 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.Schulzrinne & Petrack Standards Track [Page 14]
RFC 2833 Tones May 2000
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.Schulzrinne & Petrack Standards Track [Page 15]
RFC 2833 Tones May 2000
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](#ref-14 ""Functional criteria for digital loop carrier systems,"")].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).Schulzrinne & Petrack Standards Track [Page 16]
RFC 2833 Tones May 2000
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 events3.13 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.Schulzrinne & Petrack Standards Track [Page 17]
RFC 2833 Tones May 2000
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](#ref-16 ""AAL type 2 service specific convergence sublayer for trunking,"")], 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).Schulzrinne & Petrack Standards Track [Page 18]
RFC 2833 Tones May 2000
Event encoding (decimal)
__________________________________________________
MF 0... 9 128...[137](#page-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](#page-143)
ABCD signaling (see below) 144...[159](#page-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.Schulzrinne & Petrack Standards Track [Page 19]
RFC 2833 Tones May 2000
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](#ref-17 ""Various tones used in national networks,"")]:
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](#ref-18 ""Technical characteristics of tones for telephone service,"")] 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](#ref-18 ""Technical characteristics of tones for telephone service,"")]. Note that there are no specific guidelines for these tones. In the table, the symbol "+" indicates addition of
Schulzrinne & Petrack Standards Track [Page 20]
RFC 2833 Tones May 2000
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 tones4.3 Use of RTP Header Fields
Timestamp: The RTP timestamp reflects the measurement point for
the current packet. The event duration described in [Section](#section-3.5)
[3.5](#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.
Schulzrinne & Petrack Standards Track [Page 21]
RFC 2833 Tones May 2000
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 tonesThe 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,Schulzrinne & Petrack Standards Track [Page 22]
RFC 2833 Tones May 2000
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.Schulzrinne & Petrack Standards Track [Page 23]
RFC 2833 Tones May 2000
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.Schulzrinne & Petrack Standards Track [Page 24]
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](#ref-1 ""RTP: A Transport Protocol for Real-Time Applications"")].
Security considerations: See the "Security Considerations"
([Section 7](#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/A6.2 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](#ref-1 ""RTP: A Transport Protocol for Real-Time Applications"")].
Security considerations: See the "Security Considerations"
([Section 7](#section-7)) section in this document.
Interoperability considerations: none
Published specification: This document.Schulzrinne & Petrack Standards Track [Page 25]
RFC 2833 Tones May 2000
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/A7 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](#ref-1 ""RTP: A Transport Protocol for Real-Time Applications"")]), and any appropriate RTP profile (for example RFC 1890 [[19](#ref-19 ""RTP Profile for Audio and Video Conferences with Minimal Control"")]).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](#ref-6 ""RTP Payload for Redundant Audio Data"")].
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 and distributed, in whole or in part, without restriction of any kind, provided that the above copyright notice and this paragraph are 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 "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING 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]