The RFC 2833 Dual-Tone Multifrequency (DTMF) Media Termination Point
(MTP) Passthrough feature passes DTMF tones transparently between Session
Initiation Protocol (SIP) endpoints that require either transcoding or use of
the Resource Reservation Protocol (RSVP) Agent feature.
Note: Cisco Unified CallManager 5.0 introduced major enhancements for SIP
Cisco recommends that you have knowledge of these topics:
The information in this document is based on these software and
The information in this document was created from the devices in a
specific lab environment. All of the devices used in this document started with
a cleared (default) configuration. If your network is live, make sure that you
understand the potential impact of any command.
Technical Tips Conventions for more information on document
Cisco Unified CallManager 5.0 introduced major enhancements for SIP
trunks and overcame the limitations of earlier releases of Cisco Unified
CallManager, such as single codec support, lack of video support, and the
mandatory media termination point (MTP) for RFC 2833 DTMF support. The other
enhancements to SIP trunks in Cisco Unified CallManager 5.0 are the support for
REFER, header replacement, Subscribe/Notify, message waiting indication (MWI),
MTP removal, video support, multiple SIP trunks per inbound port number, SIP
Redirection 3XX, transport layer security (TLS), digest authentication, call
preservation, and T.38 fax relay.
You can configure Cisco CallManager SIP devices (lines and trunks) to
always use an MTP. If the configuration parameters are not set to use an MTP
(default case), Cisco CallManager attempts to dynamically allocate an MTP if
the DTMF methods for calls that are not compatible. For example, SCCP phones
support only out-of-band DTMF, and Cisco SIP phones (model 7905, 7912, 7940,
7960) support RFC2833. Because the DTMF methods are not identical, Cisco
CallManager dynamically allocates an MTP. If a SCCP phone that supports RFC2833
and out-of-band, such as Cisco IP phone 7971, calls a Cisco SIP IP phone 7940,
Cisco CallManager does not allocate an MTP because both phones support RFC2833.
With the same type of DTMF method supported on each phone, there is no need for
Cisco Unified Communications Manager 5.x is connected to a Cisco IOS
voice gateway with a SIP trunk as shown. PSTN calls to IP phones registered
with Cisco Unified Communications Manager 5.0 drop intermittently. This also
results in cross talk in both inbound, as well as outbound calls.
The DTMF information is transported between SIP endpoints with
out-of-band (OOB) and in-band signaling. In-band DTMF transport methods send
DTMF tones as either raw tones in the RTP media stream or as signaled tones in
the RTP payload with RFC 2833. Among SIP product vendors, RFC 2833 has become
the predominant method to send and receive DTMF tones and is supported by the
majority of Cisco voice products. Endpoints can negotiate the use of RFC 2833
or an out-of-band DTMF method end-to-end but, if a common DTMF method cannot be
negotiated between the endpoints, Cisco Unified Communications Manager 5.x
inserts an MTP dynamically.
Note: In Cisco Unified Communications Manager 5.x, the number of concurrent
SIP calls is limited by the number of MTP resources available.
In Cisco Unified Communications Manager 4.x, all SIP trunks are
required to allocate an MTP for DTMF and Early Offer support. With Cisco
Unified Communications Manager 5.x, the Media Termination Point
Required check-box is disabled by default. If you remove this
restriction, it increases the overall performance and frees up the MTP
resources for other applications to use. Endpoints can negotiate the use of RFC
2833 or an out-of-band DTMF method end-to-end but, if a common DTMF method
cannot be negotiated between the endpoints, Cisco Unified Communications
Manager 5.x inserts an MTP dynamically.
In order to fix this issue, uncheck Media Termination Point
Required under the SIP Trunk configuration in the Cisco Unified
Communications Manager 5.x administration page so that Cisco Unified
Communications Manager 5.x allocates MTP resources dynamically, dependent upon
DTMF signaling is not passed from the Cisco voice gateways (that run
MGCP to CallManager) to the IVR services (that run SIP Trunks to CallManager)
without the enablement of MTP required.
After enabling the require MTP option in CallManager,
this issue is resolved because the MTP is able to translate the out-of-band
DTMF signaling in MGCP to the in-band signaling that is used for the SIP
trunks. Unfortunately, there are limits on the number of software MTPs that can
be enabled on CallManager, which means that the solution does not scale as
required: one MTP is required per call. If the MTP requirement can be moved to
the Cisco voice gateways, more DSP resources are required. It is better to
simply pass the DTMF tones within the media stream straight to the IVR servers,
without any out-of-band DTMF tones. This also provides the added benefit of not
passing DTMF tones in the MGCP signaling back to CallManager, a potential
security issue when customer account numbers / pins are used.
In order to enable RFC 2833 for MGCP gateway with CCM 5.0 SIP trunk,
without the requirement of enabling MTP, configure these two commands in the
mgcp dtmf-relay voip codec all mode
mgcp package-capability fm-package
Note: These commands are included in the Cisco IOS 12.4T IP VOICE feature
set and later.
When an off-net call is attempted through the Session Initiation
Protocol (SIP) trunk, the caller gets a fast busy tone. All inbound calls to
the IP phones are fine.
In order to resolve the issue, follow these steps.
Verify whether or not there are enough media resources
Verify whether or not the Cisco IP Voice Media Streaming Service
has started. If it has not, restart the service.
Verify whether or not the proper codecs are used. Then, reset the
Software Media Termination Point (MTP) in the Cisco CallManager.
Check the protocol configured under System > Security
Profile > SIP Trunk Security Profile for outbound and inbound
calls. By default, the Cisco CallManager tries Transport Control Protocol (TCP)
on outbound calls. This can time out if it is not able to establish a TCP
connection with the SIP gateway. If the protocol is set to TCP, change it to
User Datagram Protocol (UDP).
A SIP trunk configuration is used to set up communication with a SIP
User Agent such as another Cisco Unified Communications Manager cluster or a
SIP gateway. Today, SIP is arguably the most commonly chosen protocol when
connecting to service providers and Unified Communications applications. Cisco
Unified Communications Manager 8.5 and later releases provide these SIP trunk
and call routing enhancements:
Can run on all Unified Communications Manager nodes
Up to 16 destination IP addresses per trunk
SIP OPTIONS ping keepalives
SIP Early Offer support for voice and video calls (insert MTP if
QSIG over SIP
SIP trunk normalization and transparency
Supports the use of route lists on all Unified Communications Manager
DTMF does not work on Cisco 7936 conference phones.
Cisco 7936 conference units are configured to route out a CUBE to a SIP
provider. In this case, the issue is that DTMF is not sent out. Voicemail and
WebEx sessions cannot be used.
The reason the SIP trunk has DTMF issues only for CP7936 is due to
hardware limitations of the phone. The phones do not support RTP-NTE. In order
to resolve this issue, you must have a Media Termination Point (MTP) on CUCM to
convert out of band DTMF to RTP-NTE for the SIP provider. In this case, you
must either use a 7937 model or enable MTP or HW Transcoder so that the out of
band DTMF sent by the 7936 is converted to NTE. The 7937 does not demonstrate
the same problem since it supports NTE.
A fast busy tone is received when a call is made from one CallManager
cluster to another.
In this case, you must collect the CUCM traces of the timestamp in
order to analyze them.
The traces show that the codec that is negotiated is G.711 because the
MTP box is checked and the preferred codec is set as G.711 (under the SIP trunk
configuration in the administration page of CUCM) even though the Region used
by the phone and SIP trunk are set to use G.729 instead. In addition, it shows
that when the 183 Session Progress is received by CUCM the codec is accepted
and the call moves to connect status.
Media Termination Points (MTPs) are generally not required for Delayed
Offer calls from Unified CallManager SIP trunks. For this reason, Cisco
recommends Delayed Offer as the call setup method for outbound calls from
Unified CallManager SIP trunks. For outbound Early Offer calls from Unified
CallManager, MTP resources are required (SIP Trunk MTP required box is checked)
and remain in the media path for the duration of the call.
For calls inbound and outbound from Unified CallManager, endpoints can
negotiate the use of RFC 2833 or an out-of-band DTMF method (for example, KPML)
end-to-end. If a common DTMF method cannot be negotiated between the endpoints,
Cisco Unified Communications Manger 5.x and later releases will insert an MTP
dynamically. Cisco Unified Communications Manger 5.x and later releases support
Delayed Offer (Invite without SDP) by default. Although Delayed Offer is a
mandatory part of the SIP RFC 3261 specification, some SIP applications do not
support it. In those cases, you must configure the SIP trunk to support Early
Offer by pre-allocating an MTP under the SIP trunk configuration.
MTPs are available in three forms:
Software-based MTPs in Cisco IOS gateways—Available with any Cisco
IOS T-train software release and scaling up to 500 sessions (calls) on the
Cisco 3845 Integrated Services Router.
Hardware-based MTPs in Cisco IOS gateways—Available with any Cisco
IOS T-train software release; hardware MTPs use on-board DSP resources and
scale calls according to the number of DSPs supported on the Cisco router
Software-based MTPs using the Cisco IP Voice Media Streaming
Application on a Cisco Media Convergence Server
This example configuration is for a Cisco IOS software-based
sccp local Vlan5
sccp ccm 10.10.5.1 identifier 5 version 5.0.1
! Communications Manager IP address (10.10.5.1)
sccp ccm group 5
bind interface Vlan5
associate ccm 5 priority 1
associate profile 5 register MTP000E83783C50
! MTP name (MTP000E83783C50) ... must match the Unified CM MTP name.
dspfarm profile 5 mtp
description software MTP
maximum sessions software 500
associate application SCCP