Dual-tone multifrequency (DTMF) relay is a mechanism for reliably carrying DTMF digits across VoIP connections. If you need to signal the 0 to 9, *, and # keypad digits (DTMF digits) from your IP phone across your VoIP network, you must configure DTMF relay. DTMF digits are also sometimes called TouchTone digits.
You should configure DTMF relay if you want to operate a remotely connected voice mail system, use calling card access for PSTN calls placed through a remote VoIP PSTN gateway, or access any kind of DTMF-driven interactive voice response (IVR) system (for example, telephone banking or airline flight information services).
DTMF describes a method of encoding telephone digits using two audio tones. For a conventional telephone keypad in which the keys are arranged in three columns by four rows, the first audio tone selects the row of the key, and the second audio tone selects the column. Each row-and-column tone uses a different audio frequency (pitch). This method of telephone digit signaling replaced the old-fashioned loop disconnect (dial pulse) digit dialing used by old rotary-style analog phones.
There are 16 DTMF digits (arranged as four columns by four rows). In addition to the standard 12 keypad digits0 to 9, *, and #an additional four digits form an extra fourth column of digits called simply A, B, C, and D. Because the ABCD digits are unavailable on a normal phone keypad, you are unlikely to ever come across these for normal phone calls. They are used occasionally by voice mail systems to operate an intersystem exchange of voice messages between separate voice mail systems using a standard called Analog Message Interchange Standard (AMIS).
Some security-type phones also use the ABCD digits for initial negotiation. You may also see these used in some Cisco CME configuration examples where there is a need to create telephone numbers that can't be directly dialed from a phone keypad. One example of this is if you want to create nondialable phone numbers for intercoms. You normally place an intercom call by pressing a button specifically configured for intercom (this works somewhat like a speed-dial button), so you don't need to be able to enter individual dialed digits.
Transporting DTMF Digits Reliably Using DTMF Relay
In the simple case of analog phones, the phone keypad digits result in the generation of DTMF audio tones. DTMF signaling works fine for analog phones connected directly to PSTN analog subscriber lines that travel only a relatively short distance to reach a central office (CO) telephone exchange. However, when the analog phone is connected to a VoIP system, and telephone calls are made using compressed voice (for example, G.729 at 8 Kbps), there is a substantial risk that the audio tones of the DTMF digits sent through the compressed voice path may become too distorted to be predictably recognized correctly by a remote voice mail system.
Even when you use uncompressed G.711 A-law/µ-law 64 Kbps for VoIP calls, there is still a risk that DTMF digits can get distorted in transit. This is because of the risk of packet loss in the VoIP network. If the network drops an occasional IP packet containing voice, this is usually imperceptible to the human ear. However, if an IP packet is dropped that contains an audio encoding of part of a DTMF tone, this is very likely to keep the DTMF digit detection in the far-end system from detecting the digit (or to make it erroneously detect multiple digits). This is also the reason that other nonvoice audio signals such as fax and data modems need special treatment in VoIP networks.
To work around this issue, you generally have to use some form of DTMF relay. DTMF relay causes the digit press to be detected by the PSTN trunk or analog phone interface on the VoIP gateway. The originating VoIP gateway then signals the digit as an explicit event to the far-end VoIP gateway and removes the audio signal for the DTMF from the voice packet stream. When the far-end VoIP gateway receives the signal for the DTMF event, it regenerates the DTMF audio signal and inserts it into the outgoing audio stream to the PSTN or analog phone.
In the case of the Cisco SCCP IP phones, the digit never exists as an audio signal from the VoIP perspective, because it's directly signaled via the SCCP control protocol. The digit audio that the phone user hears from the phone handset is for the benefit of the phone user only and is not passed to the VoIP connection.
Different Forms of DTMF Relay
In general, there are two main ways to signal DTMF events between VoIP gateways: H.245 digit relay and Real-Time Transport Protocol (RTP)-based DTMF digit relay. This is true for both the H.323 and SIP protocols, although the specific details are different.
The H.245 digit relay option sends a message via the H.323 control channel that is associated with the VoIP call. (This is called H.245 digit relay because it uses the H.245 control channel part of the H.323 protocol to signal the digit event.)
The RTP-based DTMF digit relay method carries the digit event through the voice media channel as a special marked RTP media packet. The problem of possible IP packet drop is overcome by sending multiple redundant copies of the event so that even if one of the copies is lost, there is little chance that all copies will be lost.
H.245 Digit Relay
H.245 digit relay comes in two flavors: signal and alphanumeric. The dial peer commands for these are dtmf-relay h245-signal and dtmf-relay h245-alphanumeric.
In signal mode, two events are sent: one to indicate the start of the digit and one to indicate the end of the digit. This lets the duration of the keypress on the phone be reflected in the duration of the digit regenerated by the far-end VoIP gateway. This is useful for calling card PSTN access where a long-duration press of the # key is sometimes used to indicate the end of a calling card call plus the intention to place a follow-on call (without needing to re-enter a calling card number and its associated PIN).
Alphanumeric mode has only a single event signal. This results in the regeneration of a fixed-duration DTMF signal (usually 200 milliseconds) by the far-end VoIP gateway. In this mode, the length of the regenerated digit is unrelated to how long you press the keypad button on the phone. Some implementations generate the alphanumeric DTMF signal when you press the phone's keypad button, and others generate the signal when you release the keypad button. You can use this duration-of-press-independent property to tell which type of VoIP DTMF digit relay is being used.
H.245 alphanumeric mode is the one that should be used with Cisco CME's IP phones.
RTP Digit Relay
RTP-based digit relay mode also has two flavors:
When you press a keypad digit on an IP phone, you hear a tone in the phone handset that corresponds to the digit you press. Although you hear this audio tone, the far-end party that your call is connected to doesn't. The IP phone sends the keypad audio signal only to the phone's handset (or speaker). It does not insert the audio digit indication into the outgoing voice packet stream. When you press the keypad digit on an SCCP-based IP phone, the phone sends a control message to the Cisco CME router via the SCCP protocol. Because the digit press originates from the phone as a control channel message, the SCCP digit message is simply converted into an H.245 alphanumeric message to send this across H.323 VoIP.
The SCCP digit press event does not indicate the duration of the keypad button press. This means that the H.245 signal method cannot be used, because the SCCP phone does not provide digit-start and digit-stop information. Also, the SCCP phones do not natively support either the RFC 2833 or cisco-rtp RTP-based digit relay mechanisms.
For you to signal DTMF keypad digits across H.323, you need to configure your VoIP dial peers as shown in Example 7-11.
Example 7-11. DTMF Relay Configuration
Router#show running-config dial-peer voice 510200 voip destination-pattern 51055502.. session target ipv4:10.1.1.1 dtmf-relay h245-alphanumeric
If you are using your Cisco CME system in a SIP network, you have to use the RFC 2833 DTMF relay method where possible. Cisco CME 3.2 (and later) software provides automatic conversion from the SCCP control channel DTMF messages received from the SCCP IP phone into standard SIP RFC 2833 RTP digits.
Call Transfer and Call Forwarding in an H 323 Network Using H 450 Services
Part I: Cisco IP Communications Express Overview
Introducing Cisco IPC Express
Building a Cisco IPC Express Network
Cisco IPC Express Architecture Overview
Part II: Feature Operation and Applications
Cisco IP Phone Options
Cisco CME Call Processing Features
Cisco CME PSTN Connectivity Options
Connecting Multiple Cisco CMEs with VoIP
Integrating Cisco CME with Cisco CallManager
Cisco IPC Express Automated Attendant Options
Cisco IPC Express Integrated Voice Mail
Cisco CME External Voice Mail Options
Additional External Applications with Cisco CME
Part III: Administration and Management
Cisco IPC Express General Administration and Initial System Setup
Configuring and Managing Cisco IPC Express Systems
Cisco IPC Express System Configuration Example
Part IV: Maintenance and Troubleshooting
Troubleshooting Basic Cisco IPC Express Features
Troubleshooting Advanced Cisco CME Features
Troubleshooting Cisco CME Network Integration
Troubleshooting Cisco UE System Features
Troubleshooting Cisco UE Automated Attendant
Troubleshooting Cisco UE Integrated Voice Mail Features
Part V: Appendixes
Appendix A. Cisco IPC Express Features, Releases, and Ordering Information
Appendix B. Sample Cisco UE AA Scripts
Appendix C. Cisco Unity Express Database Schema