Table Of Contents
Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events
Related Features and Technologies
Supported Standards, MIBs, and RFCs
Configuring DTMF Relay and NTE Payload Type
Verifying DTMF Relay and NTE Payload Type
Monitoring and Maintaining SIP NTE DTMF relay
DTMF Relay using RTP-NTE Example
RTP Using Payload Type NTE Example
Dual Tone Multifrequency Relay for SIP Calls Using Named Telephone Events
Document Update Alert
This document was originally produced for Cisco IOS Release 12.2(11)T. This feature has been updated in subsequent releases, and more recent documentation is available.
If you are using Cisco IOS Release 12.2(11)T or higher, refer to the following section in the Configuring Additional SIP Features chapter of the Cisco IOS SIP Configuration Guide, Cisco IOS Voice Configuration Library, Release 12.3:
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DTMF Relay for SIPCalls Using NTEs
Feature History
This document describes the Dual Tone Multifrequency (DTMF) Relay for SIP Calls Using Named Telephone Events (NTE) feature in Cisco IOS Release 12.2(11)T. For consistency, the feature is referred to as the SIP NTE DTMF relay feature in this document.
This document includes the following sections:
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Feature Overview
The SIP NTE DTMF relay feature is used for the following applications:
These applications are discussed in more detail in the following sections.
Note
Reliable DTMF Relay
The SIP NTE DTMF relay feature provides reliable digit relay between Cisco VoIP gateways when a low bandwidth codec is used. Using NTE to relay DTMF tones provides a standardized means of transporting DTMF tones in Real-Time Transport Protocol (RTP) packets according to section 3 of RFC 2833, RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals, developed by the Internet Engineering Task Force (IETF) Audio/Video Transport (AVT) working group. RFC 2833 defines formats of NTE RTP packets used to transport DTMF digits, hookflash, and other telephony events between two peer endpoints.
DTMF tones are generated when a button on a touch-tone phone is pressed. When the tone is generated, it is compressed, transported to the other party, and decompressed. If a low-bandwidth codec, such as a G.729 or G.723 is used without a DTMF relay method, the tone may be distorted during compression and decompression.
With the SIP NTE DTMF relay feature, the endpoints perform per-call negotiation of the DTMF relay method. They also negotiate to determine the payload type value for the NTE RTP packets.
In a SIP call, the gateway forms a Session Description Protocol (SDP) message that indicates:
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The SIP NTE DTMF relay feature can relay hookflash events in the RTP stream using NTP packets.
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SIP Phone Support
The SIP NTE DTMF relay feature adds SIP phone support. When SIP IP phones are running software that does not have the capability to generate DTMF tones, the phones use NTE packets to indicate DTMF digits. With the SIP NTE DTMF relay feature, Cisco VoIP gateways can communicate with SIP phones that use NTE packets to indicate DTMF digits. The Cisco VoIP gateways can relay the digits to other endpoints.
Benefits
This feature provides the following benefits:
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Restrictions
The SIP NTE DTMF relay feature is available only for SIP calls on Cisco VoIP gateways. The SIP NTE DTMF relay feature supports only hookflash relay and does not support hookflash generation for advanced features such as call waiting and conferencing.
Related Features and Technologies
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Related Documents
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Supported Platforms
Determining Platform Support Through Cisco Feature Navigator
Cisco IOS software is packaged in feature sets that support specific platforms. To get updated information regarding platform support for this feature, access Cisco Feature Navigator. Cisco Feature Navigator dynamically updates the list of supported platforms as new platform support is added for the feature.
Cisco Feature Navigator is a web-based tool that enables you to quickly determine which Cisco IOS software images support a specific set of features and which features are supported in a specific Cisco IOS image. You can search by feature or release. Under the release section, you can compare releases side by side to display both the features unique to each software release and the features in common.
Cisco Feature Navigator is updated regularly when major Cisco IOS software releases and technology releases occur. For the most current information, go to the Cisco Feature Navigator home page at the following URL:
Availability of Cisco IOS Software Images
Platform support for particular Cisco IOS software releases is dependent on the availability of the software images for those platforms. Software images for some platforms may be deferred, delayed, or changed without prior notice. For updated information about platform support and availability of software images for each Cisco IOS software release, refer to the online release notes or, if supported, Cisco Feature Navigator.
Supported Standards, MIBs, and RFCs
Standards
No new or modified standards are supported.
MIBs
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To obtain lists of supported MIBs by platform and Cisco IOS release, and to download MIB modules, go to the Cisco MIB website on Cisco.com at the following URL:
http://www.cisco.com/public/sw-center/netmgmt/cmtk/mibs.shtml
RFCs
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Prerequisites
Before configuring or using the SIP NTE DTMF relay feature, you must have a working VoIP network using SIP on Cisco gateways. See the documents in "Related Documents" for more information.
Configuration Tasks
See the following sections for configuration tasks for the SIP NTE DTMF relay feature. Each task in the list is identified as either optional or required.
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Configuring DTMF Relay and NTE Payload Type
To configure DTMF relay and NTE Payload Type, enter the following commands in global configuration mode:
Verifying DTMF Relay and NTE Payload Type
Enter the show running command to verify that DTMF relay and NTE are configured on the dial peer. For example:
!dial-peer voice 1000 potsdestination-pattern 4961234port 1/0/0!dial-peer voice 2000 voipapplication sessiondestination-pattern 4965678session protocol sipv2session target ipv4:11.0.13.34dtmf-relay rtp-nte! RTP payload type value = 101 (default)!dial-peer voice 3000 voipapplication sessiondestination-pattern 2021010101session protocol sipv2session target ipv4:11.0.13.34dtmf-relay rtp-ntertp payload-type nte 110! RTP payload type value = 110 (user assigned)!Monitoring and Maintaining SIP NTE DTMF relay
Router# debug voip rtp session named-event
Turns on debugging for RTP NTEs.
Router# show voip rtp connections
Shows local and remote Calling ID and IP address and port information.
Configuration Examples
This section provides the following configuration examples:
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DTMF Relay using RTP-NTE Example
Example 1 provides an example of DTMF relay using RTP-NTE:
Example 1 DTMF Relay Using RTP-NTE Example
Router(config)# dial-peer voice 62 voipRouter(config-dial-peer)# session protocol sipv2Router(config-dial-peer)# dtmf-relay rtp-nteRTP Using Payload Type NTE Example
Example 2 provides an example of RTP Using Payload Type NTE with the default value of 101:
Example 2 RTP Using Payload Type NTE
Router(config)# dial-peer voice 62 voipRouter(config-dial-peer)# rtp payload-type nte 101Command Reference
This section documents new or modified commands. All other commands used with this feature are documented in the Cisco IOS Release 12.2 command reference publications.
New Commands
Modified Commands
debug voip rtp
To enable debugging for Real-Time Transport Protocol (RTP) named event packets, use the debug voip rtp command in privileged EXEC mode. To disable debugging output, use the no form of this command.
debug voip rtp {error | session [nse | multicast | conference | dtmf-relay | named-event] | packet remote-ip ipaddress remote-port portnum packetnum | packet callid idnum packetnum}
no debug voip rtp
Syntax Description
Defaults
Debugging for RTP named event packets is not enabled.
Command Modes
EXEC
Command History
Examples
The following example illustrates the output for the debug voip rtp session named-event command. The example is for a gateway that sends digits 1, 2, 3, then receives digits 9,8,7. The payload type, event ID, and additional packet payload are shown in each log.
The first three packets indicate the start of the tone (initial packet and two redundant). The last three packets indicate the end of the tone (initial packet and two redundant). The packets in between are refresh packets that are sent every 50 milliseconds (without redundancy).
Router# debug voip rtp session named-event00:09:29: Pt:99 Evt:1 Pkt:03 00 00 <<<Rcv>00:09:29: Pt:99 Evt:1 Pkt:03 00 00 <<<Rcv>00:09:29: Pt:99 Evt:1 Pkt:03 00 00 <<<Rcv>00:09:29: Pt:99 Evt:1 Pkt:03 01 90 <<<Rcv>00:09:29: Pt:99 Evt:1 Pkt:03 03 20 <<<Rcv>00:09:29: Pt:99 Evt:1 Pkt:03 04 B0 <<<Rcv>00:09:29: Pt:99 Evt:1 Pkt:83 04 C8 <<<Rcv>00:09:29: Pt:99 Evt:1 Pkt:83 04 C8 <<<Rcv>00:09:29: Pt:99 Evt:1 Pkt:83 04 C8 <<<Rcv>00:09:29: Pt:99 Evt:2 Pkt:03 00 00 <<<Rcv>00:09:29: Pt:99 Evt:2 Pkt:03 00 00 <<<Rcv>00:09:29: Pt:99 Evt:2 Pkt:03 00 00 <<<Rcv>00:09:29: Pt:99 Evt:2 Pkt:03 01 90 <<<Rcv>00:09:29: Pt:99 Evt:2 Pkt:03 03 20 <<<Rcv>00:09:29: Pt:99 Evt:2 Pkt:03 04 B0 <<<Rcv>00:09:29: Pt:99 Evt:2 Pkt:83 05 18 <<<Rcv>00:09:29: Pt:99 Evt:2 Pkt:83 05 18 <<<Rcv>00:09:29: Pt:99 Evt:2 Pkt:83 05 18 <<<Rcv>00:09:29: Pt:99 Evt:3 Pkt:03 00 00 <<<Rcv>00:09:29: Pt:99 Evt:3 Pkt:03 00 00 <<<Rcv>00:09:29: Pt:99 Evt:3 Pkt:03 00 00 <<<Rcv>00:09:30: Pt:99 Evt:3 Pkt:03 01 90 <<<Rcv>00:09:30: Pt:99 Evt:3 Pkt:03 03 20 <<<Rcv>00:09:30: Pt:99 Evt:3 Pkt:03 04 B0 <<<Rcv>00:09:30: Pt:99 Evt:3 Pkt:03 06 40 <<<Rcv>00:09:30: Pt:99 Evt:3 Pkt:83 06 80 <<<Rcv>00:09:30: Pt:99 Evt:3 Pkt:83 06 80 <<<Rcv>00:09:30: Pt:99 Evt:3 Pkt:83 06 80 <<<Rcv>00:09:31: <Snd>>> Pt:99 Evt:9 Pkt:02 00 0000:09:31: <Snd>>> Pt:99 Evt:9 Pkt:02 00 0000:09:31: <Snd>>> Pt:99 Evt:9 Pkt:02 00 0000:09:31: <Snd>>> Pt:99 Evt:9 Pkt:02 01 9000:09:31: <Snd>>> Pt:99 Evt:9 Pkt:02 03 2000:09:31: <Snd>>> Pt:99 Evt:9 Pkt:02 04 B000:09:31: <Snd>>> Pt:99 Evt:9 Pkt:02 06 4000:09:31: <Snd>>> Pt:99 Evt:9 Pkt:82 06 5800:09:31: <Snd>>> Pt:99 Evt:9 Pkt:82 06 5800:09:31: <Snd>>> Pt:99 Evt:9 Pkt:82 06 5800:09:31: <Snd>>> Pt:99 Evt:8 Pkt:02 00 0000:09:31: <Snd>>> Pt:99 Evt:8 Pkt:02 00 0000:09:31: <Snd>>> Pt:99 Evt:8 Pkt:02 00 0000:09:31: <Snd>>> Pt:99 Evt:8 Pkt:02 01 9000:09:31: <Snd>>> Pt:99 Evt:8 Pkt:02 03 2000:09:31: <Snd>>> Pt:99 Evt:8 Pkt:02 04 B000:09:31: <Snd>>> Pt:99 Evt:8 Pkt:02 06 4000:09:31: <Snd>>> Pt:99 Evt:8 Pkt:82 06 9000:09:31: <Snd>>> Pt:99 Evt:8 Pkt:82 06 9000:09:31: <Snd>>> Pt:99 Evt:8 Pkt:82 06 9000:09:31: <Snd>>> Pt:99 Evt:7 Pkt:02 00 0000:09:31: <Snd>>> Pt:99 Evt:7 Pkt:02 00 0000:09:31: <Snd>>> Pt:99 Evt:7 Pkt:02 00 0000:09:31: <Snd>>> Pt:99 Evt:7 Pkt:02 01 9000:09:31: <Snd>>> Pt:99 Evt:7 Pkt:02 03 2000:09:31: <Snd>>> Pt:99 Evt:7 Pkt:02 04 B000:09:32: <Snd>>> Pt:99 Evt:7 Pkt:02 06 4000:09:32: <Snd>>> Pt:99 Evt:7 Pkt:82 06 5800:09:32: <Snd>>> Pt:99 Evt:7 Pkt:82 06 5800:09:32: <Snd>>> Pt:99 Evt:7 Pkt:82 06 58Related Commands
show voip rtp connections
Shows local and remote Call ID number, IP address, and port number.
dtmf-relay
To specify how an H.323 or Session Initiation Protocol (SIP) gateway relays dual tone multifrequency (DTMF) tones between telephony interfaces and an IP network, use the dtmf-relay command in dial-peer configuration mode. To remove all signalling options and send the DTMF tones as part of the audio stream, use the no form of this command.
dtmf-relay [cisco-rtp] [h245-alphanumeric] [h245-signal] [rtp-nte]
no dtmf-relay [cisco-rtp] [h245-alphanumeric] [h245-signal] [rtp-nte]
Syntax Description
Defaults
No default behavior or values.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
DTMF is the tone generated when you press a digit on a touch-tone phone. This tone is compressed at one end of a call; when the tone is decompressed at the other end, it can become distorted, depending on the codec used. The DTMF relay feature transports DTMF tones generated after call establishment out of band using either a standard H.323 out-of-band method and a proprietary RTP-based mechanism, or for SIP calls, an NTE RTP packet.
The gateway only sends DTMF tones in the format you specify if the remote device supports it. If the remote device supports multiple formats, the gateway chooses the format based on the following priority:
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The principal advantage of the dtmf-relay command is that it sends DTMF tones with greater fidelity than is possible in-band for most low-bandwidth codecs, such as G.729 and G.723. Without the use of DTMF relay, calls established with low-bandwidth codecs may have trouble accessing automated DTMF-based systems, such as voice-mail, menu-based ADC/Kentrox systems, and automated banking systems.
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Examples
The following example demonstrates use of the dtmf-relay command with the SIP NTE DTMF relay feature:
Router(config-dial-peer)# dtmf-relay rtp-nteRelated Commands
rtp payload-type
To identify the payload type of a Real-Time Transport Protocol (RTP) packet, use the rtp payload-type command in dial-peer configuration mode. To remove the RTP payload type, use the no form of this command.
rtp payload-type {cisco-cas-payload number| cisco-clear-channel number | cisco-codec-fax-ack number | cisco-codec-fax-ind number | cisco-fax-relay number | cisco-pcm-switch-over-alaw number | cisco-pcm-switch-over-ulaw number| cisco-rtp-dtmf-relay number | nte number | nse number} [comfort-noise {13 | 19}]
no rtp payload-type nte
Syntax Description
Defaults
The default number value is 101.
Command Modes
Dial-peer configuration
Command History
Usage Guidelines
Use the rtp payload-type nte command to identify the payload type of an RTP NTE. Use this command after the dtmf-relay command is used to choose the NTE method of dual tone multifrequency (DTMF) relay for a Session Initiation Protocol (SIP) call.
Examples
The following example demonstrates the use of the rtp payload-type nte command with the SIP NTE DTMF relay feature:
Router(config-dial-peer)# rtp payload-type nte 99Related Commands
dtmf-relay
Specifies how an H.323 or SIP gateway relays DTMF tones between telephony interfaces and an IP network.
Glossary
DTMF—dual tone multifrequency. Tones that are generated when a button on a touch-tone phone is pressed. When the tone is generated, it is compressed, transported to the other party, and decompressed.
IVR—Interactive voice response. Scripts which are used to collect information from a user to process commands; for example, to retrieve voice mail. DTMF digits are entered in response to IVR scripts. In low-bandwidth compression, DTMF digits can become distorted and unrecognizable by IVR scripts.
NTE—Named Telephony Event. An event such as DTMF digits that must be encoded and transported in an RTP packet. RFC 2833 specifies the format of the RTP NTE payload.
RTP—Real-Time Transport Protocol. A protocol for transporting multimedia over IP; see RFC 1889, RTP: A Transport Protocol for Real-Time Applications.
SDP—Session Description Protocol. A protocol for defining information needed to establish multimedia transport over IP. SDP transmits information such as session announcement, session invitation, transport addresses, and media types. In a SIP call, SDP messages indicates if NTE will be used, which events will be sent using NTE, and the NTE payload type value. See RFC 2327, SDP: Session Description Protocol.
SIP—Session Initiation Protocol. A protocol for transporting multimedia that is independent of the underlying packet control layer, such as User Datagram Protocol (UDP), and is based on a client/server architecture. See RFC 2543, SIP: Session Initiation Protocol.
