Verizon IP Trunking and IP Contact Center Services ... · CISCO UBE performs Delayed-Offer to Early-Offer interworking of the initial SIP INIVTE from CISCO UCM. The Cisco UBE device

© 2010 Cisco Systems, Inc. All rights reserved. Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com

Page 1 of 52 EDCS# 994149 Rev # Initial Version

Note: Testing was conducted in Verizon lab.

Application Note

Verizon IP Trunking and IP Contact Center Services: Connecting Cisco Unified Communications Manager 8.0.3 with Cisco Unified Border Element 8.5 (Enterprise Edition)

May 10, 2011 - Initial Version

Table of Contents

Introduction .............................................................................................................................................................................................................. 2 Verizon IP Trunking Overview ........................................................................................................................................................................... 2 Verizon IPCC Overview ...................................................................................................................................................................................... 2

Network Topology .................................................................................................................................................................................................... 3 System Components ................................................................................................................................................................................................. 3

Hardware Components ........................................................................................................................................................................................ 3 Software Requirements ....................................................................................................................................................................................... 3

Sample Bill of Materials ........................................................................................................................................................................................... 4 Features and Known Limitations .............................................................................................................................................................................. 4

Features Supported (IP Trunking) ....................................................................................................................................................................... 4 Known Limitations (IP Trunking) ....................................................................................................................................................................... 4 Features Supported (IPCC) .................................................................................................................................................................................. 5 Known Limitations (IPCC) ................................................................................................................................................................................. 5 Cisco UBE Features Roadmap ............................................................................................................................................................................ 5 CISCO UCM 8.X SIP Trunk Deployment Considerations .................................................................................................................................. 6

Call Flow Overview .................................................................................................................................................................................................. 6 Outbound Call Flows ........................................................................................................................................................................................... 6 Inbound Call Flows ............................................................................................................................................................................................. 7

Failover ..................................................................................................................................................................................................................... 7 Known Issues ............................................................................................................................................................................................................ 7

Inbound Call Issues ............................................................................................................................................................................................. 7 New Security Operation in Cisco IOS 15.1.2T .................................................................................................................................................... 9 Redirected Dialed Number Identification Service and Diversion Header ........................................................................................................... 9 RDNIS Configuration in Cisco Unified Communications Manager Administration ......................................................................................... 10 CISCO UCM Administrator>Device>Device Settings>SIP Profile .................................................................................................................. 12 CISCO UCM Administrator>Device>Device Settings>SIP Profile .................................................................................................................. 14

Communications Manager Configuration ............................................................................................................................................................... 15 Media Resource Group List ............................................................................................................................................................................... 15 Media Resource Group ...................................................................................................................................................................................... 15 CODEC Selection using Device Pools and Regions ......................................................................................................................................... 17 Clusterwide Parameters (System- Location and Region) .................................................................................................................................. 19 List of Device Pools and the associated Regions ............................................................................................................................................... 20 List of Phones and ATA Devices ...................................................................................................................................................................... 20 SIP Trunk Configuration ................................................................................................................................................................................... 21 Route Group Configuration ............................................................................................................................................................................... 22 Route List for Voice .......................................................................................................................................................................................... 24 Route List Details for Voice .............................................................................................................................................................................. 24 Route List for FAX ............................................................................................................................................................................................ 25 Route List Details for FAX ............................................................................................................................................................................... 26 Route Plan report for Voice and FAX Offnet calls ............................................................................................................................................ 27 CISCO UBE Example Configuration (North America)..................................................................................................................................... 29

© 2010 Cisco Systems, Inc. All rights reserved.

Important notices, privacy statements, and trademarks of Cisco Systems, Inc. can be found on cisco.com Page 2 of 52

EDCS# 994149 Rev # Initial Version


EMEA Configuration.............................................................................................................................................................................................. 38 EMEA CISCO UCM Configuration .................................................................................................................................................................. 38 EMEA CISCO UBE dial-peer Configuration .................................................................................................................................................... 43

IPCC Configuration ................................................................................................................................................................................................ 44 IPCC CISCO UCM Configuration .................................................................................................................................................................... 44 IPCC CISCO UBE dial-peer Configuration ...................................................................................................................................................... 46

Troubleshooting ...................................................................................................................................................................................................... 47 References .............................................................................................................................................................................................................. 50 Acronyms ............................................................................................................................................................................................................... 50 Important Information ............................................................................................................................................................................................ 51

Introduction

This application note describes how to configure a Cisco Unified Communications Manager (Cisco UCM) 8.0 and Cisco Unified Border

Element (Cisco UBE) Enterprise Edition 8.5 for connectivity to Verizon’s IP trunking service. The deployment model covered in this

application note utilizes Verizon’s Private IP (commercial MPLS network) to access Verizon IP Trunking. Supplemental guidelines are also

included for using Verizon IP trunking to interface to their IP-based Contact Center Service or IPCC. Please note that in the context of this

document, “IPCC” refers to a cloud-based Contact Center product from Verizon, and should not be confused with a Cisco product. Additional

supplemental guidelines are provided for an EMEA configuration.

Testing was performed in accordance with the test plans for the Verizon IP trunking (US and EMEA), and IP Contact Center services. All

features were verified.Although this document does not detail the results of the testing performed it provides the essential configurations

required for SIP interoperability with Cisco UCM/Cisco UBE and the Verizon IP Trunking and IPCC services.

Verizon IP Trunking Overview

Verizon IP trunking services simplify management of your network and can help drive operational efficiencies. They do this by consolidating

your voice services onto a SIP-based VoIP network, thereby optimizing your data IP network, and controlling costs associated with maintaining

traditional TDM local lines, trunks, and dedicated PRI circuits. Verizon also offer a native IP Trunking option that provides a SIP trunk directly

to your IPPBX, and an IP Integrated Access option that leverages a gateway device so you can interface with legacy Key or PBX systems.

And, Verizon’s latest Burstable Enterprise Shared Trunking (BEST) feature enhancement allows you to share all your voice trunking resources

across your enterprise and lets you use idle trunk capacity in one location to accommodate a traffic increase in another location. BEST helps

control costs, as fewer concurrent calls need to be purchased at each location and resources can be shared to provide time of day benefits and

peak usage management.

Verizon IPCC Overview

Verizon VoIP Inbound is a component of the IP Contact Center (IPCC) portfolio of internetworking services, which tightly couples signaling

and functionality from the Advanced Toll Free and IP networks to deliver the intelligent routing and call treatment required by contact centers.

The IPCC services are network-based and include IP Interactive Voice Response (IVR) in addition to VoIP Inbound.

VoIP Inbound is standards-compliant and provides single-call service that allows PSTN-originated Toll Free calls to seamlessly terminate and

transfer to a SIP or TDM endpoints, without call re-originations that tie up CPE port capacity. VoIP Inbound includes advanced toll free features





including automatic ISDN User Part and SIP Error overflow for reliable termination to SIP or TDM devices anywhere; and, when combined

with IP IVR, supports customer-driven pre/post call routing and/or call treatment and queuing for customers using Cisco ICM

Network Topology

Figure 1. Typical Reference Network

System Components

Hardware Components

CISCO UBE IOS version 15.1.2T2. Primary and Secondary CISCO UBE routers are used for high availability.

Cisco Unified Border Element is an integrated Cisco IOS Software application that runs on various hardware platforms, for more details:

http://www.cisco.com/go/cube

Packet Voice Data Module (PVDM). You will need to install DSP modules on a supported ISR platform if you require MTP,

Transcoding or Conference Bridge resources. These DSP resources are co-resident on the CISCO UBE routers in our lab configuration.

CISCO UCM cluster with (2) Cisco MCS 7800 Series servers (Cisco Unified Communications Manager)

Cisco Unified IP Phones

Analog Telephony Adapter for FAX, modem, or analog phones

Ethernet Switch

WAN router used to terminate the Verizon MPLS network

Software Requirements

Cisco Unified Call Manager 8.0.3

Cisco Unified Border Element CISCO UBE running IOS version 15.1.2T2






Sample Bill of Materials

Features and Known Limitations

Features Supported (IP Trunking)

For a full list of supported SIP features please refer to the “Verizon Business Retail VoIP Network Interface

Specification (for non-registering devices)” document.

All Tests were performed according to the” Verizon Business Retail VoIP Interoperability Test Plan” and the “EMEA

Retail - Test Plan” documents.

These documents may be obtained by contacting your Verizon Business Account Representative.

Known Limitations (IP Trunking)

DTMF as RFC2833 NTE (named telephone events) when a compressed audio codec is used. Note: RFC2833 is not currently supported

when using CTI Route-Points on CISCO UCM 8.0. An MTP resource is required to enable DTMF relay for any calls that utilize a CTI

Route-point.

CISCO UBE performs Delayed-Offer to Early-Offer interworking of the initial SIP INIVTE from CISCO UCM. The Cisco UBE device

receives the invite with no SDP then forwards the invite to the SIP network with SDP included.

T.38 Fax relay is not supported by Verizon IP Trunking Service at this time Note: If you have a Cisco Fax Server or other T.38 Fax

device, you will need to ensure that design considerations have been made to support this outside of the Verizon IP Trunking service.

(i.e.…T1 PRI)

Product Description Quantity

MCS7835I3-K9-CMD1 Unified CM 8.0 7835-I3 Appliance 2

CAB-AC AC Power Cord (North America), C13, NEMA 5-15P, 2.1m 4

C2921-VSEC-CUBE/K9 C2921 VSEC CUBE Bundle, PVDM3-32, UC SEC Lic, FL-CUBEE-25 2

S29UK9-1512T Cisco 2901-2921 IOS UNIVERSAL 2


WS-C2960G-24TC-L Catalyst 2960 24 10/100/1000, 4 T/SFP LAN Base Image 2


CUCM-USR-LIC Top Level Sku For User License 1

LIC-CUCM-BASIC License - 1 Basic User 50

UCM-7835-80 CUCM 8.0 7835 2

VG202 Cisco VG202 Analog Voice Gateway 1


SVGXAISK9-15001M Cisco Voice Gateway 20x Series ADVANCED IP SERVICES 1

CP-7962G Cisco Unified IP Phone 7962 2

SW-CCM-UL-7962 CUCM 3.x or 4.x RTU lic. for single IP Phone 7962 2

CP-7965G Cisco Unified IP Phone 7965, Gig Ethernet, Color 3

SW-CCM-UL-7965 CUCM 3.x or 4.x RTU lic. for single IP Phone 7965 3





Outbound SIP REFER with Replaces. CISCO UCM does not currently support generation of an outbound SIP Refer with replaces

messaging.

CISCO UCM 8.0 can only support a single codec between the end device (i.e. IP Phone, ATA) and the SIP Trunk. A workaround for

this used during testing was to create multiple Regions and Device pools in order to control the codec selection prior to being presented

to the SIP Trunk. The end devices were configured with a specific Device Pool based on the codec used for off-net calls. This is

especially true for mid-call codec negotiation, for any calls that require changing of the initial negotiated codec the CISCO UBE device

will insert a transcoder resource in order to avoid a codec mis-match between the SIP provider and the CUCM end-points.

The CISCO UBE device must have transcoder resources configured on the Cisco UBE device and registered with CUCM to support the

mid-call codec change features on Cisco UBE. This feature allows for dissimilar Voice Class Codec configurations on the incoming and

outgoing dial peers.

In order to comply with Verizon’s requirement of supporting g711u as a secondary codec for all calls and for performing Mid-Call

Codec Negotiation, Cisco has provided an acceptable solution of providing this feature via the Cisco UBE by performing transcoding

from Call Manager to SIP Trunk out to the network

Features Supported (IPCC)

For a full list of supported SIP features please refer to the “Verizon Business IP Contact Center (IPCC) Trunk

Interface Network Interface Specification” document.

All Tests were performed according to the” Verizon Business IPCC Interoperability Lab Test Plan”

These documents may be obtained by contacting your Verizon Business Account Representative.

Known Limitations (IPCC)

The IPCC service does not currently support SIP Diversion Headers

The IPCC service does not support FAX

Outbound SIP REFER with Replaces. CISCO UCM does not currently support generation of an outbound SIP Refer with replaces

messaging.

CISCO UCM 8.0 can only support a single codec between the end device (i.e. IP Phone, ATA) and the SIP Trunk. A workaround for

this used during testing was to create multiple Regions and Device pools in order to control the codec selection prior to being presented

to the SIP Trunk. The end devices were configured with a specific Device Pool based on the codec used for off-net calls. This is

especially true for mid-call codec negotiation, for any calls that require changing of the initial negotiated codec the CISCO UBE device

will insert a transcoder resource in order to avoid a codec mis-match between the SIP provider and the CUCM end-points.

The CISCO UBE device must have transcoder resources configured on the Cisco UBE device and registered with CUCM to support the

mid-call codec change features on Cisco UBE. This feature allows for dissimilar Voice Class Codec configurations on the incoming and

outgoing dial peers.

Cisco UBE Features Roadmap

This roadmap lists the features documented in the Cisco Unified Border Element Configuration guide and maps them

to the chapters in which they appear. Also listed here is the Cisco IOS software release that introduced support for a

given feature in a given Cisco IOS software release train.

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb_roadmap_ps5640_TSD_Products_Configuration_Guide_Chapt

er.html

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb_roadmap_ps5640_TSD_Products_Configuration_Guide_Chapter.html

http://www.cisco.com/en/US/docs/ios/voice/cube/configuration/guide/vb_roadmap_ps5640_TSD_Products_Configuration_Guide_Chapter.html





CISCO UCM 8.X SIP Trunk Deployment Considerations

There are several design considerations to be taken into account when deploying SIP trunks. The following URL describes those design

considerations.

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html#wp1045842

Call Flow Overview

Outbound Call Flows

The same SIP trunks are utilized between CISCO UCM to CISCO UBE for both Voice and FAX off-net calls. However, the call type (i.e.,

Voice vs. FAX) must be differentiated to ensure the desired codec is used. This delineation is achieved by performing digit manipulation at the

Route List prior to the call being delivered to the Route Group.

Each type of device (i.e., IP Phones vs. analog devices for FAX) will have separate Route-Patterns that belong to their respective partition. The

route patterns will then route the call to the specified Route List.

The Route List is used to distinguish a Voice call from a FAX call by manipulating the called party numbers. A voice call is forwarded with a

leading 9. FAX calls will strip the leading 9 and prepend the called party number with an 8. After the digit manipulation, the Route List then

forwards the call to the Route Group, which routes the call to the SIP trunks.

The SIP trunks are the same for ALL calls from CISCO UCM to CISCO UBE (see example call flows below).

The CISCO UBE will then forward the 10 digit user ID (DID) to the SIP Provider to allow the appropriate call routing

Outbound calls can either be sent to the SIP Trunks in a “Top-Down” or “Round-Robin” method.

Regardless of the method used, if when the call gets routed to the CISCO UBE and the CISCO UBE is not able to complete the call , the call is

then routed to the next SIP Trunk or CISCO UBE in the Route-group.

This provides redundancy for outbound calls by using multiple CISCO UBE devices connecting the VZ VoIP network.

Example call flow for Voice Calls (G.729)

Example call flow for FAX Calls (G.711ulaw)

Route

Pattern

9@

For Voice Calls

Route

List

Route

Group

CUCM Cluster

CUBE 2

CUBE

CUBE

VZ

VoIP CUBE 1

SIP

Trunk

No digits stripped on

Voice calls in CUCM 9 is stripped in CUBE

for Voice calls






Inbound Call Flows

Inbound calls are received from either the IP Trunking or IPCC services. These services provide a 10 digit DID for

domestic customers and a variable length DID (10, 11,12, or 13 dependent upon country) for EMEA customers for

delivery of the SIP call. The IP PBX (Cisco UCM) is then responsible for routing this call to the appropriate IP Phone

or analog device.

Failover

The VoIP Network sends periodic SIP options messages as a keepalive mechanism to determine the state of the

CISCO UBE devices. If the primary CISCO UBE does not respond to these options messages, the calls are then

routed to the Secondary CISCO UBE router.

Note: The CISCO UBE will respond to the SIP options pings by default. NO additional configuration is necessary.

The VoIP network will also re-route any calls to the secondary CISCO UBE if it receives a temporary call setup

failure SIP message from the primary CISCO UBE. (Example: 503 or 404 messages)

To allow failover for inbound calls when the primary CISCO UBE device is unable to contact the CISCO UCM

cluster.

In the CISCO UBE:

Configure “voice-class sip options-keepalive” on all dial-peers connecting to the CISCO UCM cluster.

Change the PSTN cause code mapping under the SIP-UA configuration "set pstn-cause 1 sip-status 503"

Without this configuration the incoming call setup from the VZ IP trunking service may time-out and the call would

be cancelled before trying the secondary CISCO UBE device.

Known Issues

Inbound Call Issues

When an inbound (from PSTN to Customer IP PBX) call to a DID that terminates on the SIP trunk is not defined/registered on the IP-PBX, the

IP-PBX should respond with a 40X error message.

Route

Pattern

9@

For FAX Calls

Route

List

Route

Group

CUCM Cluster

CUBE 2

CUBE

CUBE

A VZ

VoIP CUBE 1

SIP

Trunk

9 is stripped on FAX calls

in CUCM and replaced

with 8 8 is stripped in CUBE

for FAX calls





There are configurations on the Cisco UBE device that can cause this type of call failure to result in a “call loop”. This is where the call setup is

routed between the VZ VoIP network and the Cisco UBE device continually until it exceeds a timeout threshold.

An Example of this scenario is when the outbound dial-peer on the CISCO UBE is configured with a destination-pattern of .T, which is used as

a gateway of last resort for all calls.

When the Cisco UCM responds with a 40X error message the CISCO UBE will “hunt” for the next available dial-peer to route the call through.

Example:

dial-peer Voice 100 voip

description OUTBOUND G729 Voice SIP calls to VzB

translation-profile outgoing DIGITSTRIP-9

destination-pattern .T **This will match any combination of dialed digits and is not

the recommended configuration for matching outbound calls.

It is recommended to prohibit the matching of assigned DIDs on a dial-peer that is

used to route calls towards the VoIP network.

Voice-class codec 1

session protocol sipv2

session target sip-server

dtmf-relay rtp-nte

ip qos dscp af32 signaling

no vad

If the dial-plan requires the use of the above configuration it will become necessary to configure the CISCO UCM facing dial-peers with the

“huntstop” feature to prevent inbound calls from trying to route back to the Verizon VoIP network.

Example:


description To/From CISCO UCM subscriber for Voice

preference 2

**The preferred dial-peer with a session target of the subscriber CISCO UCM(huntstop

is not applied here).

destination-pattern [1-5]...

voice-class sip options-keepalive

Voice-class codec 1


session target ipv4:192.168.3.11

incoming called-number 9T

FAX rate disable

no vad

!


description To/From CISCO UCM publisher for Voice

preference 5

**The preferred dial-peer with a session target of the subscriber CISCO UCM (huntstop

is applied)

huntstop

destination-pattern 1...






Voice-class codec 1




dtmf-relay rtp-nte

no vad

New Security Operation in Cisco IOS 15.1.2T

To help mitigate toll fraud opportunities, as of 15.1.2T CISCO IOS no longer allows connections from "unknown"

sources to connect by default. Only sources on the IP Trust List are allowed (by default) and all other calls are

rejected.

IP addresses defined in the "session target ipv4:" commands on dial-peers are automatically included in the IP Trust

List. Additional valid source IP addresses can be added manually to the Trust List if needed by using the following

CLI:

voice service voip

ip address trusted list

ipv4 10.0.1.24

While it is recommended to use the increased security operation available in 15.1.2T, pre-15.1.2T IOS operation can

be restored by using the CLI:

no ip address trusted authenticate

Redirected Dialed Number Identification Service and Diversion Header

Starting with CISCO UCM Release 6.1(4) adds the Redirected Dialed Number Identification Service (RDNIS) and

diversion header capability for certain calls that use the Cisco Unified Mobility Mobile Connect feature.

The RDNIS/diversion header for Mobile Connect enhances this Cisco Unified Mobility feature to include the RDNIS

or diversion header information on the forked call to the mobile device. Service providers and customers use the

RDNIS for correct billing of end users who make Cisco Unified Mobility Mobile Connect calls.

For Mobile Connect calls, the Service Providers use the RDNIS/diversion header to authorize and allow calls to

originate from the enterprise, even if the caller ID does not belong to the enterprise Direct Inward Dial (DID) range.

Example Use Case

Consider a user that has the following setup:

Desk phone number specifies 89012345.

Enterprise number specifies 4089012345.





Remote destination number specifies 4088810001.

User gets a call on desk phone number (89012345) that causes the remote destination (4088810001) to ring as well.

If the user gets a call from a nonenterprise number (5101234567) on the enterprise number (4089012345), the user

desk phone (89012345) rings, and the call gets extended to the remote destination (4088810001) as well.

Prior to the implementation of the RDNIS/diversion header capability, the fields populated as follows:

Calling Party Number (From header in case of SIP): 5101234567

Called Party Number (To header in case of SIP): 4088810001

After implementation of the RDNIS/diversion header capability, the Calling Party Number and Called Party Number

fields populate as before, but the following additional field gets populated as specified:

Redirect Party Number (Diversion Header in case of SIP): 4089012345

Thus, the RDNIS/diversion header specifies the enterprise number that is associated with the remote destination.

RDNIS Configuration in Cisco Unified Communications Manager Administration

To enable the RDNIS/diversion header capability for Mobile Connect calls, ensure the following configuration takes

place in Cisco Unified Communications Manager Administration:

All gateways and trunks must specify that the Redirecting Number IE Delivery — Outbound check box gets

checked.

In Cisco Unified Communications Manager Administration, you can find this check box by following the following

menu paths:

For H.323 and MGCP gateways, execute Device > Gateway and find the gateway that you need to configure. In the

Call Routing Information - Outbound calls pane, ensure that the Redirecting Number IE Delivery - Outbound

check box gets checked. For T1/E1 gateways, check the Redirecting Number IE Delivery - Outbound check box in

the PRI Protocol Type Information pane.

• For SIP trunks, execute Device > Trunk and find the SIP trunk that you need to configure.

In the Outbound Calls pane, ensure that the Redirecting Diversion Header Delivery - Outbound check box

gets checked









Early-Media Cut-thru: Enable PRACK on CISCO UCM

Early media refers to media (e.g., audio and video) that is exchanged before the called-party accepts a particular session.

Typical examples of early media generated by the called-party are ringing tone and announcements (e.g., queuing status).

Early media generated by the caller typically consists of voice commands or dual tone multi-frequency (DTMF) tones to drive interactive voice

response (IVR) systems.

Enabling PRACK is required in order to allow early media between CISCO UCM and CISCO UBE.

PRACK- Provisional Acknowledgement to a Session not yet established

• Purpose is to acknowledge progress information on a requested process

• The INVITE Includes a Require header stipulating the User Agent Client (UAC) wants a reliable provisional response

SIP Rel1XX Enabled: This parameter determines whether all SIP provisional responses (other than 100 Trying messages) get sent reliably to

the remote SIP endpoint.

If this parameter is disabled, CISCO CallManager does not acknowledge or confirm 18X messages. Valid values specify True (acknowledge

18X messages with PRACK) or False (do not acknowledge 18X messages with PRACK).

The SIP REL1XX parameter is located in the SIP Profile. Once the SIP Profile has been changed to support PRACK for all messages, the

profile will then need to be applied to the appropriate SIP Trunk device.

CISCO UCM Administrator>Device>Device Settings>SIP Profile

Change the SIP Rel1XX Options from default value of disabled to enabled for all 1xx messages





Note: No changes are required on the CISCO UBE. The CISCO UBE supports PRACK and Early Media by default.

Known Issue on CUCM 8.0 with PRACK enabled:

Semi-attended call transfers over SIP Trunk results in one-way audio with Prack enabled.

Current workaround is to disable Prack on the SIP Trunk interface in CUCM 8.0.





CISCO UCM Administrator>Device>Device Settings>SIP Profile

If Early-media is required as mentioned previously in this document, then PRACK will need to be enabled and the end-users will need

to ensure they use fully attended transfer method to transfer calls.





Communications Manager Configuration

Media Resource Group List

List of Media Resource Groups configured for the SIP Trunk MRGL

Media Resource Group

Configured Conference Bridge resource associated with DSP resources configured on CISCO UBE









CODEC Selection using Device Pools and Regions

All Voice calls through the SIP trunk should use G.729 and FAX devices should use G.711. Note in the configuration

below, there are two regions. Calls between the “Default” and “SIP Trunk Offnet” region will use G.729 and calls

between “Default” and “SIP Trunk Offnet” use G.711. Applying this configuration to our testbed, the SIP trunk is

placed in a Device Pool with the “SIP Trunk Offnet” region, and phone devices should be placed in a Device Pool that

with the “Default” region. Devices used for analog FAX should use a Device Pool with the “SIP Trunk Offnet”

region. Devices that belong to the same region are configured to use the G.711 codec





Note: With CISCO UCM 8.0 the system defaults for Intra-Region codec preference is to use the highest quality audio

codec. By default this is G722 or G711.

The system default for Inter-Region codec preference is G729.

The above region configuration is used to ensure that these codecs will be used if the system defaults are changed.





Clusterwide Parameters (System- Location and Region)





List of Device Pools and the associated Regions

List of Phones and ATA Devices

Configured Device Pools will determine the codec used by each endpoint for Offnet SIP calls





SIP Trunk Configuration

The SIP Trunk Offnet Device Pool is configured for codec negotiation and the SIP_Trunk_MRGL is selected for Conference Bridge resources.

Note: MTP required Not Selected





Route Group Configuration

Both SIP Trunks are members of the same Route Group









Route List for Voice

The previously defined ROUTE GROUP is selected in the Voice Route List

Route List Details for Voice

No Digits are discarded for off-net Voice calls.

The leading “9” is preserved when the call is forwarded to the CISCO UBE, this allows the CISCO UBE to differentiate the call as Voice and

use the corresponding G.729 CODEC.





Route List for FAX

The previously defined ROUTE GROUP is selected in the FAX Route List (similar to Voice Route List)





Route List Details for FAX

The “9” is stripped from the called party number and replaced with an “8”.





This dial plan configuration ensures that the user only needs to dial a “9” for Voice and FAX off-net calls.

Route Plan report for Voice and FAX Offnet calls

The configured partition on each endpoint will determine how the Offnet SIP calls get routed and allows for a leading

9 to be dialed regardless of type of device. Phone or FAX devices will be able to use the same dial-plan from the user

perspective.









CISCO UBE Example Configuration (North America)

Configuration of Cisco Unified Border Element (CISCO UBE) IOS version 15.1.2T2

Critical commands are marked in Bold with footnotes at bottom of each page

version 15.1

service timestamps debug datetime msec localtime

service timestamps log datetime msec

no service password-encryption

service sequence-numbers

!

hostname CUBE

!

boot-start-marker

boot-end-marker

!

!

logging buffered 5000000

no logging rate-limit

no logging console

!

no aaa new-model

!

no ipv6 cef

ip source-route

ip cef

!

!

ip dhcp pool IPPHONES1

network 192.168.3.0 255.255.255.0

option 150 ip 192.168.3.10

default-router 192.168.3.103

!

ip domain name pipiptrunksit2.gsiv.com2

ip name-server 166.38.98.2

ip name-server 10.0.1.4

multilink bundle-name authenticated

!

!

!

!

!

crypto pki token default removal timeout 0

!

!

voice-card 0

dspfarm

1 (Optional ) DHCP Service: automatically assign IP address and TFTP server (option 150) configuration to IP Phones

2 DNS Domain name for SIP Realm and name server list for DNS resolution





dsp services dspfarm

!

!

!

voice service pots

!

voice service voip

ip address trusted list3

ipv4 10.0.1.13 255.255.255.255

ipv4 10.0.1.17 255.255.255.255

ipv4 10.1.0.25 255.255.255.255

ipv4 10.1.0.24 255.255.255.255

address-hiding

allow-connections sip to sip4

fax protocol none

sip

early-offer forced5

midcall-signaling passthru6

!

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

!

voice class codec 2

codec preference 1 g711ulaw

!

!

voice cause-code

!

!

!

voice translation-rule 5

rule 1 /^91\(.*\)/ /+1\1/

rule 2 /^2/ /5302222/

rule 3 /^1/ /5302221/

rule 4 /^4/ /5302224/

rule 5 /^9\(.*\)/ /\1/

!


rule 2 /^9\(.*\)/ /\1/

!


rule 2 /^8\(.*\)/ /\1/

3 Only sources on the IP Trust List are allowed (by default) and all other calls are rejected.

4 Allow SIP to SIP call Processing

5 Use this command to forcefully configure a Cisco Unified Border Element to send a SIP invite with SDP on the Out-Leg

(OL), Delayed-Offer to Early-Offer for SIP calls. This is applied to all voip dial-peers. 6 Enables support for SIP Supplementary Services





!

!

voice translation-profile DIGITSTRIP87

translate called 11

translate redirect-target 5

translate redirect-called 5

!

voice translation-profile DIGITSTRIP98

translate calling 5

translate called 10

translate redirect-target 5

translate redirect-called 5

!

!

license udi pid CISCO2911/K9

hw-module pvdm 0/0

!

!

!

!

redundancy

!

!

!

translation-rule 711

!

!

!

!

!

interface GigabitEthernet0/0

description connection to Vz IP Network

ip address 172.17.8.10 255.255.255.0

duplex auto

speed auto

!


description connection to CUCM LAN

ip address 192.168.3.103 255.255.255.0

duplex auto

speed auto

!


ip address dhcp

shutdown

duplex auto

speed auto

7 Strip the leading “8” from outgoing called number, also performs digit manipulation for transferred calls.

8 Strip the leading “9” from outgoing called number, also performs digit manipulation for transferred calls and calling

party number.





!

ip forward-protocol nd

!

no ip http server

no ip http secure-server

!

ip route 0.0.0.0 0.0.0.0 172.17.8.1

!

!

!

!

control-plane

!

call treatment on9

call threshold global cpu-avg low 68 high 75

call threshold global total-mem low 75 high 85

call threshold global total-calls low 20 high 40

!

voice-port 0/0/010

input gain -6

output attenuation 4

no non-linear

no vad

playout-delay maximum 120

playout-delay nominal 15

playout-delay minimum low

timeouts interdigit 2

timing digit 300

station-id number 2168

caller-id enable

!

voice-port 0/0/1

input gain -6

output attenuation 4

no non-linear

no vad

playout-delay maximum 120

playout-delay nominal 15

playout-delay minimum low

timeouts interdigit 2

timing digit 300

station-id number 5302221167

caller-id enable

!

voice-port 0/0/2

!

voice-port 0/0/3

!

9 Global Call Admission Control based on Resource utilization

10 Optional FXS port for FAX devices connected directly to the CISCO UBE





!

mgcp fax t38 ecm

!

sccp local GigabitEthernet0/1

sccp ccm 192.168.3.10 identifier 2 priority 2 version 7.0

sccp ccm 192.168.3.11 identifier 1 priority 1 version 7.0

sccp

!

sccp ccm group 10

associate ccm 1 priority 1

associate ccm 2 priority 2

associate profile 12 register CONF001

associate profile 10 register XCODE001

!

dspfarm profile 10 transcode11

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 12

associate application SCCP

!

dspfarm profile 12 conference12

description conference bridge

codec g711ulaw

codec g729ar8

maximum sessions 12

associate application SCCP

!

dial-peer voice 1 pots

service session

destination-pattern 2168

incoming called-number 2168

port 0/0/0

!

!

!

dial-peer voice 100 voip

description OUTBOUND to VzB

translation-profile outgoing DIGITSTRIP913

destination-pattern 9T14



11 DSP Resources for Transcoding registered with CISCO UCM cluster 12

DSP resources for Conferencing registered with CISCO UCM cluster 13

Strip the leading “9” from outgoing called number 14

Match on outbound calls from CISCO UCM with leading “9”





voice-class codec 1 offer-all15

voice-class sip asserted-id pai

dtmf-relay rtp-nte16


no vad

!


description INBOUND from VzB



incoming called-number [1-5]...17

voice-class codec 1 offer-all

dtmf-relay rtp-nte

no vad

!

!

!

!

!


description OUTBOUND FAX to VzB


destination-pattern 8T

no modem passthrough




voice-class sip asserted-id pai

voice-class sip privacy disable

fax rate 14400


no vad

!


description INBOUND FAX dial peer from VzB





15

Sends a list of all available CODECs to the SIP Network. The “offer-all” keyword sends all available codecs without

filtering based on list configured in the associated voice-class

16

Forwards DTMF tones by using RTP with the Named Telephone Event (NTE) payload type.

17

Enables CISCO UBE to set configuration parameters to incoming calls based on the received called number. This is

required to set the DTMF, FAX, and CODEC parameters for the “In-Leg” of the VoIP call. 18

Strip the leading “8” from outgoing called number







no vad

!

!

!

!

!

!

!

!

!

!

!


description connection to CM3

preference 5

destination-pattern [1-5]...20



incoming called-number 9T21

voice-class codec 1

voice-class sip options-keepalive22

dtmf-relay rtp-nte

fax rate 14400

no vad

!


description connection to CM4

preference 2





voice-class codec 1


dtmf-relay rtp-nte

19


required to set the DTMF, FAX, and CODEC parameters for the “In-Leg” of the VoIP call.

20

Enables CISCO UBE to set configuration parameters and call routing for incoming calls destined to CISCO UCM

endpoints. This will match the incoming called party information.

21


required to set the DTMF, FAX, and CODEC parameters for the “In-Leg” of the VoIP call.

22 Enables monitoring of dial-peer targets using Out of Dialog Options PING messages. Used here to monitor the status of

the CISCO UCM SIP interface.





fax rate 14400

no vad

!

dial-peer voice 2 pots

service session


port 0/0/1

!


description INBOUND FAX dial peer from VzB to local FXS






no vad

!


description FAX dial peer from/to CUCM

preference 10





voice-class codec 224


no vad

!


description FAX dial peer from/to CUCM

preference 1




incoming called-number 8T25

voice-class codec 2


no vad

!

!

sip-ua

set pstn-cause 1 sip-status 503

set pstn-cause 3 sip-status 50326

23

Enables CISCO UBE to set configuration parameters and call routing for incoming calls destined to CISCO UCM

endpoints. This will match the incoming called party information.

24

CODEC is set on dial-peer to force use of g711ulaw for FAX calls. 25

Match on outbound calls from CISCO UCM with leading “8”





retry invite 2

retry bye 2

retry cancel 2

timers trying 550

sip-server dns:pcclv1n0022.pipiptrunksit2.gsiv.com27

g729-annexb override

!

!

!

gatekeeper

shutdown

!

!

line con 0

stopbits 1

line aux 0

stopbits 1

line vty 0 4

exec-timeout 0 0

privilege level 15

logging synchronous

login local

transport input ssh

line vty 5 15

exec-timeout 0 0

privilege level 15

password password

logging synchronous

login local

transport input ssh

!

exception data-corruption buffer truncate

scheduler allocate 20000 1000

ntp master 3

ntp peer 199.249.19.1

ntp peer 199.249.18.1

end

26 Overrides the default value of the SIP status code to correspond with the PSTN cause code. 27

SIP Proxy FQDN name for outbound SIP calls to the IP Trunking service





EMEA Configuration

EMEA CISCO UCM Configuration

The following steps are required to enable localised Network tones and User Interface:

1. Download necessary localisation files from http://www.cisco.com/cisco/web/download/index.html (requires valid CCO account)

2. Install localisation software on every Communications Manager in the cluster.

http://www.cisco.com/cisco/web/download/index.html





This does require a restart to enable the localisation file after installation.





3. Using the CISCO UCM Administration website either change the locale information at the device pool level or at the Phone device level.

Example shows change to Network Locale on the Phone configuration page:

Note: User Locale changes the User interface only and is controlled independently of the network tones.





4. All EMEA Phones should be setup in similar regions as the North America phone configuration to ensure G729 is the preferred CODEC.

5. Next create a variable-length Route-Pattern with “#” as terminating digit.

Example: 9.011!#

Note: The previously configured Voice Route List is utilized for this route-pattern in order to allow the complete

calling number to be sent to CISCO UBE.









EMEA CISCO UBE dial-peer Configuration

The CISCO UBE configuration for EMEA is very similar to the US (Domestic) IP Trunking configuration.


description OUTBOUND G729 Voice SIP calls to VzB






dtmf-relay rtp-nte


no vad

!


description INBOUND Voice SIP calls from VzB EMEA




incoming called-number [1-5]...

dtmf-relay rtp-nte

no vad

!


description To/From CISCO UCM subscriber for Voice

preference 2



voice-class codec 1




FAX rate disable

no vad

!


description To/From CISCO UCM publisher for Voice

preference 5

destination-pattern 1...


voice-class codec 1




dtmf-relay rtp-nte

no vad





IPCC Configuration

IPCC CISCO UCM Configuration

The CISCO UCM Configuration changes required for IPCC services to work properly are:

Verify all IPCC end-points (Phones and Gateways) are in the same Region to allow negotiation of the G.711ulaw

codec.

Disable diversion-header support on the SIP Trunk device configuration.***Need to check this graphic





For out-bound IPCC calls a 9.1800632XXXX Route-pattern must be configured in the Communications Manager.





IPCC CISCO UBE dial-peer Configuration

The CISCO UBE dial-peers must be configured to negotiate only the G.711 codec for all IPCC inbound calls.

This requires specific incoming called numbers for IP Toll-Free calls.

Example: User calls 8005551212 and IPCC routes the call to 1212 with the following dial-peer configured on the

CISCO UBE router.

Note: In this example the IPCC network is only sending the last 4 digits of the called number.


description OUTBOUND to VzB IP Toll Free



codec g711ulaw


session target dns:rchtcsd05011.vzbi.com

dtmf-relay rtp-nte


no vad

!

!

!


description G.711 INBOUND from VzB IP Toll Free

codec g711ulaw




dtmf-relay rtp-nte

no vad

!

!

!

!


description G.711 To/From CISCO UCM subscriber IP Toll Free

preference 2



codec g711ulaw



dtmf-relay rtp-nte

no vad

!

!

!

!






description G.711 To/From CISCO UCM publisher IP Toll Free

preference 5



voice-class codec 2

voice-class sip early-offer forced



dtmf-relay rtp-nte

no vad

Troubleshooting

Always capture logs by enabling logging buffer: “logging buffered 200000”

Remember to disable the console logging: “no logging console”

Add sequence numbering for debugs: “service sequence-number”

Debug Commands

debug ccsip all

debug voip ccapi inout

debug voip dialpeer inout

debug transcoding

debug dspfarm all

The following table lists key "show" commands giving output that enables you to monitor Cisco UBE health, traffic and activity.

Key "show" Commands on Cisco UBE





Category Command Information Provided

Configuration show version Displays the version of the image on the router

show flash: Displays information about flash: file system

show ip interface brief Displays brief summary of IP status and configuration

show startup-config Displays the startup configuration on the router

show running-config Displays the present/running con configuration on the

router

show debug Displays the debugs currently enabled

show voice iec desc <> Displays definition of an Internal Error Code

show logging Displays the contents of logging buffers

Traffic show call active voice Displays complete details of an active call like media

settings, call statistics, SRTP on/off, etc.

show call active voice brief Displays a brief version of active voice calls, e.g.

transmitted and received packets and duration of call

show call active voice compact Displays a compact version of active voice calls

show voip rtp connections Displays active RTP connections

show call history voice Displays calls stored in the history table for voice

Router Health show processes cpu sorted

<1min/5min/5sec>"

Displays sorted output based on percentage of CPU

utilization

show processes cpu sorted history Displays CPU history information in a graph format

show memory processor Displays memory statistics

show process memory <> Displays memory per process

show memory debug leaks Runs the memory leak detector

show alignment Displays alignment data and spurious memory references

CAC show call threshold config Displays configured resource information





show call treatment config Displays call admission control information

show call treatment stats Displays call treatment statistics

SIP show sip-ua connections udp brief Displays summary of SIP UDP connection information

show sip-ua connections udp detail Displays details of SIP UDP connection information

show sip-ua connections tcp brief Displays summary of SIP TCP connection information

show sip-ua connections tcp detail Displays details of SIP TCP connection information

show sip-ua register status Displays SIP registration status

Transcoding and DSPs show diag Displays diagnostic and hardware information for port

adapters and modules

show sdspfarm units Displays transcoder registration status

show sccp connection Displays the active SCCP connections

show sccp Displays SCCP protocol information

show dspfarm dsp active Displays the active DSPs

show call active voice | inc

CoderTypeRate="

Displays call connectivity, codec and the media type

information

show call active voice comp Displays codec information for transcoding calls

DTMF Relay show call active voice | inc

tx_DtmfRelay

Displays the DTMF-relay used for the call





References

Cisco UBE on Cisco.com


CISCO UCM 8x SIP Trunk Documentation:

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/8x/trunks.html - wp1044916

Cisco UBE PBX / SP Interoperability

http://www.cisco.com/go/interoperability

Verizon Business IP Trunking Services

http://www.verizonbusiness.com/us/products/voip/trunking/

Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)

http://www.ietf.org/rfc/rfc3960.txt

Redirected Dialed Number Identification Service and Diversion Header

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/rel_notes/6_1_4/cucm-rel_note-614.html#wp854592

Cisco Unified Border Element (CUBE) Management and Manageability Specification

http://www.cisco.com/en/US/partner/prod/collateral/voicesw/ps6790/gatecont/ps5640/white_paper_c11-613550.html

Acronyms

Acronym Definition

SIP Session Initiation Protocol

SCCP Skinny Client Control Protocol

TDM Time Division Multiplexing

CISCO UCM Cisco Unified Communications Manager

CISCO UBE Cisco Unified Border Element

PRACK Provisional Response Acknowledgement

TUI Telephony User Interface



http://www.cisco.com/g

http://www.verizonbusiness.com/us/products/voip/trunking/

http://www.ietf.org/rfc/rfc3960.txt

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/rel_notes/6_1_4/cucm-rel_note-614.html#wp854592

http://www.cisco.com/en/US/partner/prod/collateral/voicesw/ps6790/gatecont/ps5640/white_paper_c11-613550.html





Important Information

THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE

WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO

BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE

FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS. IN NO EVENT SHALL CISCO OR ITS

SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING,

WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY

TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH

DAMAGES.

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Documents

Verizon IP Trunking and IP Contact Center Services ... · CISCO UBE performs Delayed-Offer to Early-Offer interworking of the initial SIP INIVTE from CISCO UCM. The Cisco UBE device