Help

Questions? Feedback

Overview
Steps to Use the Tool
Reading the Results Page
Example
Understanding DSP Capacity Engineering

Overview

The total number of PVDMs (Packet Voice/fax Digital Signal Processor (DSP) Modules) required on a Cisco 2800, 3800, 2900 or 3900 series Integrated Services Router (ISR) for a particular voice configuration is calculated by the DSP calculator tool. You provide information about the desired configuration to the tool, and the calculator provides output recommending the PVDMs required for your configuration.

To determine DSP needs on platforms other the Cisco 2800, 3800, 2900 and 3900, please use the previous generation DSP Calculator.

Tool Input

DSP requirements and available features are determined by a number of factors, so the input to the tool includes these attributes of your voice configuration:

  • Router platform type
  • IOS release
  • Interface cards for each slot on the router
  • The number of voice channels needed on each interface and the codec type
  • The number of conferences
  • The codec and size of the conferences
  • The number of transcoding sessions and the codecs involved

Tool Output

Based on the input configuration given to the tool, the calculator determines the number of PVDMs that must be ordered. The output of the calculator has the following attributes:

  • It provides only one recommendation of PVDMs – there are always several other PVDM combinations that can also serve the same configuration.
  • It optimizes for PVDM slot use, i.e. the tool will fill the first slot with the highest density PVDM (the PVDM3-256 or PVDM2-64) before starting to populate the second slot.
  • It assumes a “flex complexity” setting for voice gateway card configurations
  • It rounds up the recommendation when there is ambiguity of how much DSP resources are required.
  • It defaults to using PVDM3s on the motherboard slots if a Cisco 2900 or 2900 ISR platform is selected. If you want output using only PVDM2s, click the “Motherboard DSP type” button on the first page.

Please contact your Cisco account representative, or submit a question via the tool feedback link, if you require DSP engineering advice outside the parameters of the calculator’s output.

The DSP calculator does not perform CPU engineering for the chosen platform, it determines only the DSP resources required for the given configuration. Please perform CPU engineering as a separate exercise.

Steps to Use the Tool

Follow the steps below to use the tool:

  1. Select the Router and Software Release
  2. Specify the voice gateway cards and voice channels required: TDM Services page
  3. Specify the transcoding and conferencing capabilities required: IP Services page
  4. Specify the video channels required: H.320 Transport Protocol page
  5. Get the output of PVDMs needed for the configuration

Step 1: Select Router and Software Release

  1. Select a router model from the drop down list. If you do not see the platform of your choice, try the previous generation DSP Calculator for coverage of older platforms.

 

Model Drop Down Box

 

  1. Select a Cisco IOS release from the drop down list. The list shows only Cisco IOS releases supported on the selected router model. Note that Cisco 2900s and 3900s ISRs are supported only from the 15.0.1M IOS release and later.

IOS Drop Down Box

Step 2: TDM Services Page

This section explains the “TDM Services” page of the calculator which collects input on the voice interface cards, channels and codecs used for voice gateway (i.e. connections to the PSTN or a PBX) configurations.

Note: If you do not have any voice gateway configuration, press the “Next” button at the bottom of the page and go to the next step.

The slot layout shown represents the architecture of the router platform you chose in Step 1 – a Cisco 2951 router in the example below. The area in the red box represents the onboard, or motherboard, HWIC slots – interface cards in these slots use onboard DSPs. The area in the green box represents the Network Module or Service Module slots and there will be as many shown as the router you chose has. Generally, interfaces on these slots use local NM/SM-based DSPs with the exception of the EVM-HD-8FXS/DID card which always uses onboard DSPs.

Hwic's and NM's

  1. For each onboard slot that you want to use, select the onboard interface card(s) and specify the number of voice calls of each codec type on this card. Only the interface cards that are compatible with the selected router model and Cisco IOS release are shown.

 

The “Maximum Supported Voice Channels” column shows the maximum number of calls possible on this interface card. In the codec columns, fill in the maximum number of simultaneous calls of each codec category that you intend to run during the busiest hour of the day. The sum of the input in the codec columns cannot exceed the “Maximum Supported Voice Channels”, but can be less.

Hwic1

  1. For each Network or Service Module slot that you want to use, select an interface card. Only those interface cards that are compatible with the selected router model and Cisco IOS release are shown.

 

The “Maximum Supported Voice Channels” column shows the maximum number of calls possible on this interface card – 60 for the two T1/E1s in the example below. Continuing the example, if you intend to run no more than 50 calls on these interfaces, of which 80% use G.711 and the remainder G.729A, fill in 40 and 10 respectively into the codec category columns.

TDM Calls

  1. Select the global configuration settings for your situation.
    1. DSP sharing (default no): Select “No” for all common configurations. DSP Sharing may be used in rare situations.
    2. Secure voice (default no): Select “No” for all common configurations. Select “Yes” if your configuration includes voice encryption.
    3. H.320 (default no): Select “No” for all common configurations. Select “Yes” if there are PRI interfaces on the gateway you are configuring that carry both ISDN voice and video calls. Step 4: H.320 Transport Protocol Page covers further configuration of video calls.
    4. Motherboard DSP type (default PVDM3): This option only appears on the Cisco 2900 and 3900 series platforms. The default, and recommended, DSP type for the motherboard slots is PVDM3s. If you want to force the use of PVDM2s in these slots, select the “PVDM2” button. The co-existence of PVDM2s and PVDM3s on the same platform is supported (although the two types cannot be mixed on the motherboard slots).

 

TDM Radio Buttons

  1. Click the Next button

 

Step 3: IP Services Page

This section explains the “IP Service Services” page of the calculator which collects input on the DSP services configured on this router. These services do not involve voice interfaces cards or TDM technologies, and include transcoding, conferencing and MTPs (Media Termination Points).

Note: If you do not have any IP services configuration, press the “Next” button at the bottom of the page and go to the next step.

IP Services do not depend on the location of any interface cards in the router chassis, so no slot layout is shown. Instead, the IP Services page shows a matrix of supported transcoding and conference parameters where you can select the number of each type of session that you require.

  1. Enter the number of transcoding sessions of each codec category required.

 

DSP requirements for transcoding does not depend on the controlling call agent: Cisco Unified Communications Manager, Cisco Unified Communications Manager Express, or Cisco Unified Border Element. DSP transcoding density does depend on the type of transcoding configured. If you are using the “dspfarm profile 1 transcode” IOS CLI, enter your transcoding channels in the “Transcoding” row. If you are using the “dspfarm profile 1 transcode universal” IOS CLI, enter your transcoding channels in the “Universal Transcoding” row.

Each transcoding “session” represents the entire end-to-end call – both call legs are considered part of the same call and count as a single session. For example, for a call being transcoded for G.711 to G.729A, fill in “1” in the Medium Complexity column.

IP Services

  1. Enter the number of conferences of each codec required.

 

DSP requirements for conferencing does not depend on the controlling call agent (Cisco Unified Communications Manager or Cisco Unified Communications Manager Express), the type of conference (Meet-Me or Ad-hoc), or the actual number of participants that join the conference. Conference DSP resource allocation depend on the

    • configured size of the conference (maximum possible participants)
    • the codec combination of the participants
    • whether encryption is used

Enter into the matrix the number of simultaneous conferences of each size (8-party, 16-party, 32-party or 64-party) you intend to configure on the router. If you only intend to do 3-party conferences, then your best fit choice is 8-party conferences.

Enter the conferences into the codec column that represents the worst-case codec of any participant that may join the conference. The codec “weights” from best to worst are: G.711, G.729 (all versions), G.722, iLBC. You attain better channel density on the DSPs if you use “lighter” codecs in conferences and may leverage a separate transcoder to facilitate this design. E.g. if only very few of the potential conference participants use iLBC, you can configure the conferences for G.729 (better channel density than iLBC) and use a separate transcoder for the iLBC participants, rather than penalizing all conferences with the lower channel density required to accommodate iLBC participants natively in all conferences.

Conferencing

  1. Click in the Next button

 

Step 4: H.320 Transport Protocol Page

This section explains the “H.320 Transport Protocol” page of the calculator which collects input on ISDN (H.320) video calls. The “H.320 Transport Protocol” page appears in the calculator when:

  • You configured BRI or PRI interfaces on the “TDM Services” page during Step 1
  • Checked the “Are you using H.320?” radio button to Yes at the bottom of the “TDM Services” page during Step 1

Video calls share ISDN channel with voice calls on BRI and PRI interfaces. Each ISDN B-channel offers 64K worth of video bandwidth, so a 384K video calls uses 6 B-channels. The DSP calculator determines available video bandwidth (shown in the top right of the screen below) by:

  • Taking the BRI and PRI interfaces you defined and the aggregate number of channels available on these
  • Subtracting out the channels you indicated as being used for voice on the “TDM Services” page during Step 1
  1. Select the number of calls of specific bandwidths in the matrix below.

 

H320 Video

  1. Click in the Next button

 

Step 5: DSP Module Requirements

The Results page of the calculator displays the PVDM options required to meet the configuration entered during Steps 1-4.

Reading the Results Page

The DSP calculator results page displays the:

  • Router type selected.
  • Cisco IOS version selected.
  • The recommended PVDMs required to meet the minimum requirements of the configuration (red box). This information can be used to complete the part numbers in your order or to check whether the PVDMs present in a router are sufficient to serve the desired configuration. Any other PVDM combination that provides equal or more DSP resources than the minimum recommendation can be substituted.
  • Information on which slot to insert the DSPs in (green box).

Results
 

Note: The calculator optimizes PVDM slot use. The results provided allocate the fewest number of PVDM slots to achieve the DSP resources necessary for the configuration entered. There are many other combinations of PVDMs that can also serve your configuration.

Cisco offers various ISR voice bundles including some level of DSP resources (PVDM2s or PVDM3s). Should more DSP resources be required by your configuration than included in the bundle, the bundle content may be upgraded during ordering with the PVDM upgrade kits including the following part numbers: PVDM2-8U16, PVDM2-8U32, PVDM2-32U48 and PVDM2-32U64 for PVDM2s and PVDM2-16U48, PVDM3-48U64, PVDM3-64U128, PVDM3-128U192 and PVDM3-192U256  for PVDM3s.
 
PVDMs may also be ordered as spares to be added to a router you already own. Part numbers include: PVDM2-8=, PVDM2-16=, PVDM2-32=, PVDM2-48= and PVDM2-64= for PVDM2s and PVDM3-16=, PVDM3-32=, PVDM3-64=, PVDM3-128=, PVDM3-192=, PVDM3-256= for PVDM3s.

Note: The tool calculates DSPs needed for a particular configuration. It is not a system capacity tool and does not check CPU/platform support of the configuration.

Example

 
The following configuration is required for a medium-sized branch office:

  • Centralized Unified Communications Manager at another site
  • Use G.711 within the site, and G.729A to other sites
  • Branch site: 50 users
    • 12 PSTN channels—fractional PRI
      • 66% terminate on local phones (G.711)
      • 33% of calls terminate at another site (G.729A), perhaps after a transfer
    • Four FXO lines for backup
    • Six fax machines
    • Six to ten people in two conferences at any one time
    • Five transcoding channels for calls from other sites into local voicemail

The network deployment of the example is shown below:

Topology

A Cisco 2911 router is used for the branch office configuration with the following interface cards:

  • A VWIC2-1MFT-T1/E1 for the data connection
  • A VWIC2-1MFT-T1/E1 for the T1 PSTN connection
  • An EVM-HD-8FXS/DID with an EM-HDA-6FXO daughtercard for the fax machines and backup FXO lines

Router

The configuration for the PSTN and fax connectivity is entered into the DSP calculator tool on the “TDM Services” page as shown below.

  • HWIC slot 1 is left open for the data connection
  • HWIC slot 2 is filled with a VWIC2-1MFT-T1/E1 with 12 total channels, 8 (66%) G.711 and 4 (33%) G.729A
  • The Service Module slot is filled with the EVM-HD-8FXS/DID with 6 FXS channels for the fax machines and 4 FXO channels for the PSTN backup connectivity. These channels are expected to be local to the branch office only, so they are entered as G.711 channels.

Example TDM

The configuration for the transcoding and conferencing requirements are entered into the DSP calculator tool on the “IP Services” page as shown below.

  • 5 transcoding sessions between G.711 and G.729A are entered into the G.729 (all variations) column as that is the heaviest of the two codecs involved
  • 2 conferences with mixed G.711/G.729A codecs are entered into the G.729 (all variations) column as that is the heaviest of the codecs involved

Example IP

The output page shows that the entire configuration can be served with a PVDM3-64 on the motherboard.

Example Results

The order for the Cisco 2911 router could use the base chassis or any of the voice bundles (CISCO2921-V/K9, C2921-CME-SRST/K9, C2921-VSEC/K9). As the deployment in the example requires SRST, a CME-SRST bundle order is shown below. This bundle already includes a PVDM3-32, so this is upgraded to a PVDM3-64.

Example Order

Understanding DSP Capacity Engineering

This section summarizes ancillary information that is helpful to consider when doing DSP engineering.

Interface Cards

A range of VIC (analog and BRI) and VWIC (T1/E1) cards are supported on the various ISR platforms. The DSP calculator tool shows only selections that apply to the router model and Cisco IOS release you selected.

All ports on analog and BRI interface cards (all the VIC and EVM card ports) require DSPs at a minimum of G.711 level to be present on the router chassis regardless of how many calls may be run on the ports, or whether the ports are used at all.

T1/E1 interface cards require DSPs at a minimum of G.711 level for each timeslot configured on a ds0-group or pri-group on the controller.

The G.703 VWIC interface cards (VWIC2-1MFT-G703 and VWIC2-2MFT-G703) are primarily data cards supporting unstructured E1 services. Unstructured E1s cannot support voice – voice requires a channelized (structured) E1. A G.703 VWIC interface card can optionally be configured to support a structured E1, and in that mode supports voice in exactly the same manner as the non-G.703 VWICs.

Codec Categories

Codecs include a compression algorithm that reduces the bandwidth of the resulting voice stream. Generally, the more the bandwidth savings, the higher the complexity of the compression algorithm. More DSPs resources are required for the higher complexity (i.e. heavier) algorithms than for the simpler (i.e. lighter) algorithms. Cisco categorizes different codecs into three levels of complexity which in turn map to the amount of DSP resources required to handle a voice stream encoded with a codec belonging to each category. These are:

Low Complexity

Medium Complexity

High Complexity

G.711 a-law

G.726 (16K)

G.729

G.711 u-law

G.726 (24K)

G.729B

Fax passthrough

G.726 (32K)

G.723.1 (5.3K)

Modem passthrough

T.38 Fax Relay

G.723.1 (6.3K)

Clear-channel

Cisco Fax Relay

G.723.1A (5.3K)

 

G.729A

G.723.1A (6.3K)

 

G.729AB

G.728 (16K)

 

G.722 (64K)

Modem Relay

 

 

iLBC (13.3K)

 

 

iLBC (15.2K)

Additionally, the DSPs can be configured in one of three modes:

  • Flex Complexity: This is the default mode and used in almost all deployments. “Flex complexity” mode means the DSP can handle a mix of voice channels using any combination of codecs from the above categories. A certain amount of DSP resources are set aside for each call based on the codec it uses. In this mode the overall channel density of the DSP therefore varies based on the combination of calls you have at any one time. E.g. you could have 4 G.711 calls (light codec) per DSP, but only 3 calls if one of them is G.729A (heavier codec), and only 2 calls if both are G.729A. The advantage is you can use any mix of codecs in your calls.
  • Medium Complexity: In this configuration mode there is a fixed number of voice channels per DSP based on the amount of DSP resources required for a Medium Complexity call. Calls may use codecs only from the Low and Medium Complexity categories. Calls attempting to negotiate High complexity codecs will fail. The advantage of this mode is guaranteed and fixed channel density engineering, but at the cost of disallowing certain codecs and having overall sub-optimal DSP channel density if many calls use codecs from the “Low Complexity” category.
  • High Complexity: In this configuration mode there is a fixed number of voice channels per DSP based on the amount of DSP resources required for a High Complexity call. Calls may use codecs only from any of the categories. The advantage of this mode is guaranteed and fixed channel density engineering allowing all codecs, but at the cost of overall sub-optimal DSP channel density if many calls use codecs from the Low or Medium complexity categories.

Note: Almost all customer deployments use a flex complexity configuration and this is also the recommended configuration mode.

Use the codec complexity CLI to configure the DSPs in a particular mode. “Flex complexity” is the default mode. “Voice-card 0” denotes the motherboard DSPs, “voice-card 1” the DSPs in NM/SM slot 1 etc. An example is shown below.
voice-card 0
       codec complexity medium

The DSP calculator tool assumes a flex complexity voice card configuration – for static medium and high complexity voice card configurations, all channels should be entered into the appropriate column regardless of actual codec used.

Video

The Cisco IOS routers are capable of supporting up to a 1Mbps H.320 video call, which is equivalent to 16 ISDN B-channels. Each channel provides 64K of bandwidth, and up to 16 B-channels can be bonded into a single video call. H.320 video is supported on BRI and PRI ISDN TDM interfaces and all channels bonded into a single video call must be mapped to the same DSP. B-channels from multiple BRI or PRI interfaces can be bonded into the same video call.
Currently security (SRTP) for video calls is not supported. Choosing “yes” for secure voice on the TDM Services page will have no effect on the DSP calculation for video calls.

H320 Radio Button

Cisco IOS routers support H.261, H.263, H.263+ and H.264 video codecs.
            dial-peer voice 500 voip
                        video codec {h261|h263|h263+|h264}

Some attributes of configuring and deploying H.320 video include:

  • Video is supported for H.323 and SIP deployments
  • If the minimum bandwidth for a video call is not available, the call falls back to audio
  • H.320 is supported using either PVDM2s or PVDM3s
  • The maximum bandwidth for an H.320 is 1 Mbps (16 bonded B-channels)
  • There is no support for High-Definition H.320 calls

For more information on H.320 video use the configuration guide.

Transcoding

The Cisco IOS router DSPs serve a controlling call agent for function such as transcoding. The DSPs never invoke transcoding services by themselves. Three different call agents can use the Cisco IOS router DSPs for transcoding services:

  • Cisco Unified Communications Manager (UCM)
  • Cisco Unified Communications Manager Express (UCME)
  • Cisco Unified Border Element (UBE)

The call agent used to invoke transcoding services does not directly affect DSP density, but it does affect the transcoding options and codec choices that may be available. Up to and including Cisco UCM and Cisco UCME 8.0, these call agents support only traditional, or G.711-any, transcoding. This means one side of the transcoding stream must be G.711, while the other side can be G.729 (all variations). With subsequent releases of these call agents, universal transcoding may also be supported.

Cisco UBE as a call agent supports both traditional and universal transcoding. Universal, or any-to-any, transcoding means any combination of codecs can be transcoded, e.g. G.729 to iLBC. Traditional G.711-any transcoding has higher DSP density than universal any-any transcoding because it is known that one side of the call is the lightweight G.711 codec.

Traditional, or G.711-any, transcoding supports the following codec combinations:

  • G.711 a-law 64 Kbps
  • G.711 µ-law 64 Kbps
  • G.729, G.729A, G.729B, G.729AB 8 Kbps

Universal, or any-any, transcoding supports the following codec combinations:

  • G.711 a-law 64 Kbps
  • G.711 µ-law 64 Kbps
  • G.723 5.3 and 6.3 Kbps
  • G.729, G.729A 8 Kbps
  • G.729B, G.729AB 8 Kbps
  • iLBC 13.3 and 15.2 Kbps
  • G.722 64 Kbps

Note: In the DSP calculator transcoding fields, enter the number of transcoding sessions into the column corresponding to the “heavier” of the two codecs involved in the transcoding session you want to support, using the table of codec categories given earlier.

The number of sessions (transcoded calls, both sides of the call counts as a single session) entered into the DSP calculator matches the “session” CLI shown below:
dspfarm profile nn transcode
maximum sessions 20

Conferencing

DSPs for conferences are statically assigned based on the configuration. It does not matter – for purposes of DSP engineering – how many actual participants join the conference (this metric does matter to the CPU engineering of the router). In general, the following size conferences are supported:

  • 8-party
  • 16-party
  • 32-party
  • 64-party

Conference support in the DSPs also depends on the codec mix of the participants. The heavier codecs limit the number and size of conferences that can be supported by the DSP. E.g. 64-party conferences are supported only if all participants are G.711. 8-Party conferences are supported for all codecs. When secure conferencing is used, only 8-party conferences are supported.

The number of conferences entered into the DSP calculator matches the “session” CLI shown below:
dspfarm profile nn conference
maximum sessions 5

MTP

The Cisco IOS routers support two types of Media Termination Point (MTP) functions for call agents:

  • Software MTP
  • Hardware MTP

A software MTP does not use DSPs at all, so it is not applicable to the DSP calculator. Software MTP capacity sizing is a CPU engineering exercise only. All typical MTP functions used by Cisco Unified Communications Manager (e.g. RFC2833, SIP Early-offer generation, media anchoring for supplementary services) can be achieved with a software MTP only.

A Hardware MTP does use DSPs and is only used when the MTP functions described above additionally require G.711 repacketization, e.g. from 10ms to 20ms. For purposes of sizing these applications in the DSP calculator, use the G.711-G.711 transcoding fields.

Secure Calls

Secure voice solutions are achieved with a variety of encryption technologies including TLS, IPSec and Secure RTP (SRTP). The only aspect of voice encryption that affects DSP density is SRTP, as the RTP packet encryption is done in the DSP as the IP packet is formed, and encrypted voice streams therefore have lower DSP channel density than non-encrypted streams.

SRTP is supported for TDM voice interfaces (analog, BRI and T1/E1) for G.711, G.729 (all variations) iLBC and G.722 codecs. When the “Secure voice” radio button on the “TDM Services” page of the DSP calculator is set to “Yes”, it means SRTP applies to all TDM voice interfaces configured on the router.

Encrypted (SRTP) transcoding configurations under control of Cisco Unified Communications Manager Express or Cisco Unified Border Element are supported. It is not currently supported by Cisco Unified Communications Manager.

Encrypted (SRTP) conference configurations under control of Cisco Unified Communications Manager are supported. It is not currently supported by Cisco Unified Communications Manager Express.

DSP Sharing

DSP sharing is seldom used and is not the default operation of the Cisco IOS router DSPs. DSP sharing should be turned on only when:

  • absolutely necessary
  • when the implications of this mode on the router architecture are fully understood

DSP sharing is not a means to using fewer DSPs for the same configuration although it can, in select cases, result in slightly lower overall DSP needs. Rather, DSP sharing allows “borrowing” of DSPs from slots that are not local to the voice interface card because not enough local DSPs are available.

DSP sharing allows DSPs from the motherboard to be used by voice interfaces in NM/SM slots, and vice versa, i.e. DSPs located on NM/SM slots to be used by voice interface on the motherboard HWIC slots. DSP sharing also allows voice interfaces in different NM/SM slots to use each other’s DSPs. Default operation is always to use local DSPs to where the voice interface is located, with the exception of the EVM-HD-8FXS/DID card which always uses motherboard DSPs no matter where it is located.

DSP sharing can never be used for analog/BRI interfaces, only for T1/E1 interfaces. DSP sharing is applicable only to TDM voice interfaces services, it does not affect IP services such as transcoding, conferencing or MTPs.

DSP sharing alters the clocking architecture of the router and therefore changes which T1/E1 controllers can receive an external clock from a service provider connection. Please consult with your Cisco sales representative to understand these implications before turning on DSP sharing in a configuration. Turning the “DSP sharing” radio button in the DSP calculator to “Yes” while not implementing the DSP sharing CLI on the actual router will result in not enough DSPs being available, or their being in the wrong slots.

PVDM2 and PVDM3 Co-Existence

The Cisco 2800 and 3800 series router platforms support only PVDM2s in both motherboard and NM slots.

The Cisco 2900 and 3900 series router platforms support both PVDM2s and PVDM3s. The motherboard DSP slots can take either type of PVDM, but are natively wired for PVDM3s. To use PVDM2s in the motherboard slots, an adapter card (the PVDM2-ADPTR card) is necessary. PVDM2s and PVDM3s cannot be mixed in the motherboard slots, all slots must contain the same type of DSP card.

The NM-HDV2 is supported in the SM slots of the Cisco 2900 and 3900 platforms, and this card supports only PVDM2s. SM/NM slots populated with PVDM2s are supported on the same router that has PVDM3s in the motherboard slots. DSP sharing between domains with different types of DSP cards is not supported.

DSP Oversubscription

In general oversubscription of the DSPs is not allowed by the Cisco IOS routers. Configuration CLI with fail when an attempt is made to add more features requiring DSPs (such as transcoding and conferencing profiles) if not enough spare DSP resources are available. DSP resources are allocated statically (during configuration time) to most DSP services. The allocation algorithm cannot be controlled via CLI.

Oversubscription of T1/E1 voice interface channels is allowed. Enough DSPs for G.711 density of all T1/E1 timeslots configured is required, but if heavier codecs than G.711 are used, there is the possibility of oversubscription. Ensure that you enter the correct number of channels of the appropriate codecs in the “TDM Services” page of the DSP calculator to make sure your T1/E1 configuration is not oversubscribed.