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SIP DTMF Trigger Detection

The SIP DTMF Trigger Detection and Notification functionality enables the

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to look for specific DTMF trigger patterns across the packet network, and to notify an external SIP entity when such patterns are detected. If a mid-call trigger is configured for a call, it is activated as soon as the call is connected. When the 
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detects DTMF input matching the pattern criteria of the trigger, it sends the DTMF input in a SIP INFO message to the SIP call peer with content type application/DTMF.

The

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supports the following methods of relaying DTMF digits for transcoded calls:

  • In-band—Leaves the DTMF tones in-band as encoded audio.
  • Out-of-band—Carries DTMF in the signaling protocol (SIP or H.323).
  • RFC 2833—Encodes DTMF into RTP using a format and Payload Type (PT) distinct from the audio encoding. The
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    supports up to five DTMF payloads bound to different clock rates if the SIP Trunk Group services flag honorSdpClockRate is enabled (Refer to SIP Trunk Group - Services - CLI or Trunk Group - SIP Trunk Group (EMA) page for configuration details.).

DSP resources are used when interworking between different codecs. When using in-band to out-of-band or in-band to DTMF digit (RFC 2833) interworking (both directions), DSP resources are required.

Pass-through calls are supported when receiving digits through RFC 2833 or out-of-band (SIP INFO).

DTMF Interworking

The

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also supports interworking between different DTMF Relay methods. For packet-to-packet calls both signaling and media (bearer) traffic flow through the
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(whether or not media transcoding is also employed). For signaling traffic, the SBC inherently provides address and port mapping between the addresses used within the carrier's own network and those “visible” to the peering partner.

The

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includes two seeded DTMF trigger profiles:

  • Ingress—Applied to the calling (ingress leg) side.
  • Egress—Applied to the called (egress leg) side.

To enable this feature, configure and enable the

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DTMF trigger for the appropriate leg of the call. Changes to configuration parameters do not affect existing calls.

To configure via CLI, use following syntax:

% set profiles dtmfTrigger <egress | ingress> interdigitTimeout <timeout_value> longdigitDuration <longdigitDuration> pattern <trigger_pattern> state <enabled | disabled>

Below is a CLI example of enabling DTMF interworking for PSP "profile1" with transcoding.

set profiles media packetServiceProfile pprfile1 packetToPacketControl conditionsInAdditionToNoCommonCodec differentDtmfRelay enable
commit

 

Refer to DTMF Trigger - CLI, Packet Service Profile - CLI or Dtmf Trigger Profile - Dtmf Trigger, Packet Service Profile - Flags, Packet To Packet Control - Condition In Addition To No Common Codec (EMA) for configuration details.

DTMF Interworking Without DSP

The

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supports interworking between RFC2833/RFC4733 and out-of-band DTMF methods without using DSPs. Below is a CLI example of enabling DTMF interworking for PSP "profile1" without transcoding.

% set profiles media packetServiceProfile profile1 flags digitDetectSendEnabled enable
% set profiles media packetServiceProfile profile1 flags interworkDtmfWithoutTranscoding enable

To use DTMF interworking without DSP resources, the Packet Service Profile flag digitDetectSendEnabled must be enabled.

 

Refer to Packet Service Profile - CLI or Packet Service Profile - Flags (EMA) for configuration details.

SIP RTP Relay

To preserve the privacy and security model for media flows as well, the SBC implements an RTP Relay. In RTP Relay, the call endpoints send media packets to an address/port on the SBC. As the media packets flow through the SBC, the source and destination addresses are changed such that the SBC is itself always one of the endpoints in each call leg. Thus, the carrier's internal addresses are not exposed to the peering partner, and inbound media flows are directed only to a limited set of well-defined SBC addresses for security filtering purposes.
 

 Figure SIP IP Pipe Example Using SBC 5000 Series below depicts the two SIP IP "pipes" used by the

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:

  • The SIP Signaling Channel exchanges call signaling packets with a SIP Application Server (AS).
  • The IP Media Stream exchanges call data with a SIP Media Server (MS), IAD or SIP phone.

SIP IP Pipe Example Using SBC 5000 Series

The SIP Application Server manages the SIP MS/IAD/phone through the IP cloud, shown by the dotted line in Figure 1 . This interaction is logically removed from the SIP SBC software and hence is not annotated in the following examples.

Audio call data may be encoded according to any of the algorithms below for delivery through the IP Media Stream:

  • G.711
  • G.711 with Silence Suppression
  • Cloned encoding algorithms
  • Isolating the tones and designating them explicitly within the IP Media Stream
  • Isolating the tones, then designating and delivering them explicitly on the SIP Signaling Channel instead of the IP Media Stream

The IP Media Stream uses the Realtime Transport Protocol (RTP) to deliver call data for all audio compression algorithms.

Audio Stream Examples Using DTMF

Audio Stream Examples With DTMF

Audio StreamRepresentation
G.711G.711 audio packet stream containing uncompressed DTMF information encoded in IP Media Stream along with all other audio data.
G.711/RFC2833G.711 audio packet stream containing DTMF information that is placed into the RTP stream using the RFC2833 mechanism. In this case, the underlying DTMF audio also remains embedded in the audio RTP packets.
G.711/RFC2833 (with DTMF Digits Removed)G.711 audio packet stream containing DTMF that is placed into the RTP stream using the RFC2833 mechanism, but further manipulates the RTP stream by removing these tones from the audio RTP packets.
G.711 (with DTMF Digits Removed)G.711 audio packet stream with all DTMF removed. The DTMF is delivered through DTMF INFO messages that are conveyed on the SIP Signaling Channel. The RFC2833 mechanism is not present in this delivery scheme.
G.72XAudio compression packet stream containing compressed DTMF information that is encoded in the IP Media Stream along with all other (compressed) audio data.
G.72X/RFC2833Audio compression packet stream containing DTMF information that is placed into the RTP stream using the RFC2833 mechanism. In this case, the underlying DTMF audio also remains embedded in the audio RTP packets.
G.72X/RFC2833 (with DTMF Digits Removed)Audio compression packet stream containing DTMF that is placed into the RTP stream using the RFC2833 mechanism, as above, but further manipulates the RTP stream by removing these tones from the audio RTP packets.
G.72X/ (with DTMF Digits Removed)Audio compression packet stream that has all DTMF removed. The DTMF is delivered through DTMF INFO messages that are conveyed on the SIP Signaling Channel. The RFC2833 mechanism is not present in this delivery scheme.
SIP Call Signaling Packets without DTMF INFO MessagesExchange signaling information for SIP calls. In this case, these packets do not contain any DTMF information, and are conveyed on the SIP Signaling Channel.
SIP Call Signaling Packets with DTMF INFO MessagesExchange all signaling information for SIP calls, including DTMF information. DTMF digits are delivered via SIP INFO messages. These packets are conveyed on the SIP Signaling Channel.

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