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Internet Engineering Task Force                                   AVT WG
Internet Draft                                       Schulzrinne/Petrack
ietf-avt-tones-02.txt                                Columbia U./MetaTel
October 22, 1999
Expires: February 2000


   RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

STATUS OF THIS MEMO

   This document is an Internet-Draft and is in full conformance with
   all provisions of Section 10 of RFC2026.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF), its areas, and its working groups.  Note that
   other groups may also distribute working documents as Internet-
   Drafts.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as "work in progress".

   To view the list Internet-Draft Shadow Directories, see
   http://www.ietf.org/shadow.html.

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 a
   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 sytems, as well as
   "RTP trunks". In the first application, the Internet telephony
   gateway detects DTMF on the incoming circuits and sends the RTP



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   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
   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



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   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.
   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 of 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 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 signalling 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



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   audio tones as regular RTP audio packets (e.g., as payload format
   PCMU), in addition to the named signals.

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 SHOULD 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



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   and only send named events as both the primary and redundant
   encodings.

   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 for country-specific subscriber line tones (Section 3.13) and;

        o for trunk events (Section 3.14).

   A compliant implementation MUST support the events listed in Table 1.
   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



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   not need to render fax-related events described in Section 3.11.

   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-



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             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.

        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 set the end bit only when
             retransmitting the last packet for a tone. This avoids
             having to wait to detect whether the tone has indeed ended.

             Receiver implementations MAY use at 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



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   suggested.


        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
   even 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.


    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                             |



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   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |           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      |             4800          |         4         |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |F|   Block PT  |
   |0|     97      |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     digit     |R R| volume    |          duration             |
   |       9       |0 0|     7     |             1600              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     digit     |R R| volume    |          duration             |
   |       1       |0 0|    10     |             2000              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |     digit     |R R| volume    |          duration             |
   |       1       |0 0|    20     |              400              |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+



3.9 Indication of Receiver Capabilities using SDP

   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.

   For example, if the payload format uses the payload type number 100,
   and the implementation can handle the common DTMF tones (events 0
   through 11) and the dial and ringing tones, it would include the
   following description in its SDP message:




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   a=fmtp:100 0-11,66,70



   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

   Tables 1 summarizes the events belonging to the DTMF payload type.


                         Event  encoding (decimal)
                         _________________________
                         0--9                0--9
                         *                     10
                         #                     11
                         A--D              12--15
                         Flash                 16


   Table 1: DTMF 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, this signal is rendered as ANS with a bar
             on top.)

        ANSam: The modified answer tone (ANSam) [3] is a sinewave signal
             at 2100 +/- 1 Hz with phase reversals at an interval of 450
             +/- 25 ms, amplitude-modulated by a sinewave at 15 +/- 0.1


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             Hz. This tone [10,8] is sent by modems [11] and faxes to
             disable echo suppressors.

        /ANSam: This is the same signal as ANSam, except that it
             reverses phase at an interval of 450 +/- 25 ms. It disables
             both echo cancellers and echo suppressors. (In the ITU
             Recommendation, this signal is rendered as ANSam with a bar
             on top.)

        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]

        CRd: Capabilities Request (CRd) [12] is a dual-tone signal with
             tones at tones at 1375 Hz and 2002 Hz for 400 ms for the
             initiating side and 1529 Hz and 2225 Hz for the responding
             side, 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, CRd 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."

        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."

        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.

        MRd: Mode Request (MRd) [12] is a dual-tone signals with tones



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             at 1375 Hz and 2002 Hz for 400 ms for the initiating side
             and 1529 Hz and 2225 Hz for the responding side, 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]

        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 now 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.

   In summary, procedures in Table 2 are used.


         Procedure                      indications
         ________________________________________________________
         V.25 and V.8                   ANS, ANS, ...
         V.25, echo canceller disabled  ANS, /ANS, ANS, /ANS
         V.8                            ANSam, ANSam, ...
         V.8, echo canceller disabled   ANSam, /ANSam, ANSam, ...


   Table 2: Use of ANS, ANSam and /ANSam in V.x recommendations



3.12 Line Events

   Table 4 summarizes the events and tones that can appear on a
   subscriber line.




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                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
                CRd                                      41
                CRe                                      42
                ESi                                      43
                ESr                                      44
                MRd                                      45
                MRe                                      46


   Table 3: Data and fax events

   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.

        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.

        Ringing tone: The call has been placed to the callee and a
             calling signal (ringing) is being transmitted to the
             callee.

        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.


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        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. Replaced by
             Greensleeves

        Record tone: The caller has been connected to an automatic
             answering device and is requested to begin speaking.

        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

        Off-hook warning tone: The caller has left the instrument off-
             hook for an extended period of time.  activated.

   The following tones can be heard be 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. (Use by law enforcement authorities is optional.)

        CPE alerting signal (CAS): A tone used to alert a device to an
             arriving in-band FSK data transmission. A CAS is a combined



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             2130 and 2750 Hz tone, both with tolerances of 0.5% and a
             duration of 80 to 80 ms. CAS is used with ADSI services and
             Call Waiting ID services, see Bellcore GR-30-CORE, Issue 2,
             December 1998, Section 2.5.2.

   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).


               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


   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.



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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


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 [14].

   [NOTE: the list below, below wink, does not agree with the MF
   description in van Bosse, p. 74.]


        ABCD transitional: 4-bit signaling used by digital trunks. For
             N-state signaling, the first N values are used.

             The T1 ESF (extended super frame format) allows 2, 4, and
             16 state signalling bit options. These signalling bits are
             named A, B, C, and D.  Signalling 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 signalling
             with A and B bits. On the CEPT E1 frame, all signalling is
             carried in timeslot 16, and two channels of 16-state (ABCD)
             signalling are sent per frame.

             Since this information is a state rather than a changing
             signal, implementations SHOULD use the following triple-


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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
            Return 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

             redundancy mechanism, similar to the one specified in ITU-T
             Rec. I.366.2 [15], 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).

        Return seizure: Seizure by answering exchange, in response to
             outgoing seizure. [NOTE: Not clear why the difference here,
             but not for Unseize. Should probably be just Seizure.]

        Unseize circuit: Transition of circuit from off-hook to on-hook
             at the end of a call.



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        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.

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 [16]:

        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.

        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



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          seconds.

        o ITU Recommendation E.180 [17] 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 [17]. Note that there are no specific guidelines
   for these tones. In the table, the symbol "+" indicates addition of
   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
          CED                        2100        3.3          --
          ANS                        2100        3.3          --
          ANSam                   2100*15        3.3          --
          V.21 "0" bit, ch. 1        1180      0.033
          V.21 "1" bit, ch. 1         980      0.033
          V.21 "0" bit, ch. 2        1850      0.033
          V.21_"1"_bit,_ch._2________1650______0.033____________
          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.



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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. 2.



      0                   1                   2                   3
      0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |    modulation   |T|  volume   |          duration             |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |R R R R|       frequency       |R R R R|       frequency       |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |R R R R|       frequency       |R R R R|       frequency       |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     ......

     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
     |R R R R|       frequency       |R R R R|      frequency        |
     +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


   Figure 2: Payload format for tones



   The payload contains the following fields:

        modulation: The modulation frequency, in Hz. The field is a 9-
             bit unsigned integer, allowing modulation frequencies up to
             511 Hz. If there is no modulation, this field has a value
             of zero.

        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.




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             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,
             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. 3 shows an
   example. In that example, the RTP packet combines two "tone" and one
   "telephone-event" payload.  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



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   (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.



      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 3: Combining tones and events in a single RTP packet



6 MIME Registration



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6.1 audio/telephone-event

        MIME media type name: audio

        MIME subtype name: telephone-event

        Required parameters: none.

        Optional parameters: The "events" parameter lists the events
             supported by the implementation. Events are listed as one
             or more comma-separated elements. Each element can 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.

             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 occuring 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




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        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.

        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 [18]).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.

8 IANA Considerations

   This document defines two new RTP payload formats, named telephone-



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   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.

   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, Mike Fox, Terry Lyons, Colin Perkins
   and Steve Magnell.

10 Authors

   Henning Schulzrinne
   Dept. of Computer Science
   Columbia University
   1214 Amsterdam Avenue
   New York, NY 10027
   USA
   electronic mail:  schulzrinne@cs.columbia.edu

   Scott Petrack
   MetaTel
   45 Rumford Avenue
   Waltham, MA 02453
   USA
   electronic mail:  scott.petrack@metatel.com

11 Bibliography

   [1] H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, "RTP: a
   transport protocol for real-time applications," Request for Comments
   (Proposed Standard) 1889, Internet Engineering Task Force, Jan. 1996.

   [2] S. Bradner, "Key words for use in RFCs to indicate requirement
   levels," Request for Comments (Best Current Practice) 2119, Internet
   Engineering Task Force, Mar.  1997.

   [3] International Telecommunication Union, "Procedures for starting



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   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.

   [5] International Telecommunication Union, "Multifrequency push-
   button signal reception," Recommendation Q.24, Telecommunication
   Standardization Sector of ITU, Geneva, Switzerland, 1988.

   [6] C. Perkins, I. Kouvelas, O. Hodson, V. Hardman, M. Handley, J. C.
   Bolot, A. Vega-Garcia, and S. Fosse-Parisis, "RTP payload for
   redundant audio data," Request for Comments (Proposed Standard) 2198,
   Internet Engineering Task Force, Sept.  1997.

   [7] M. Handley and V. Jacobson, "SDP: session description protocol,"
   Request for Comments (Proposed Standard) 2327, Internet Engineering
   Task Force, Apr. 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 signalling 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



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   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.

   [14] J. G. van Bosse, Signaling in Telecommunications Networks
   Telecommunications and Signal Processing, New York, New York: Wiley,
   1998.

   [15] 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.

   [16] 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.

   [17] 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.

   [18] H. Schulzrinne, "RTP profile for audio and video conferences
   with minimal control," Request for Comments (Proposed Standard) 1890,
   Internet Engineering Task Force, Jan. 1996.





















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