SIP Trunk between CUCM 7.X to Service Provider 2011

Introduction

This document covers the Network Design and Configuration of SIP Trunk deployement between the Cisco Unified Communications Manager (CUCM) and Service Provider Soft Switch.
This model of SIP Trunk Interconnection is widely used by large enterprises with branch office connecting to SIP Service Provider for routing Long distance Calls. It covers the Network topology with Components used and the Configuration screenshots.

Components Used

• Cisco Unified Communications Manager 7.1(3)
• Cisco Unified Border Element (Cisco UBE)
• 7962 IP Phones

Configuration in this document are tested on specific lab environment using CUCM 7.1(3). The Procedures remains the same for CUCM 7.x and 8.x. Hence this document will be applicable for both CUCM 7.x and CUCM 8.x

Network Topology

Cisco Unified Communications Manager CUCM is located at the Headquarters with IP Phones Interconnected and Cisco UBE with SRST feature is located at the Branch office Interconnecting the IP Phones in the Branch location.

Case Study

SIP Trunk has to be established between the CUCM 172.16.101.210 at Headquarters, Cisco Unified SRST and the SIP Service Provider Soft Switch 10.6.33.32.
Long Distance Calls with Route Pattern 98400 has to be routed from the CUCM IP Phones at the Headquarters and Branch towards the SIP Service Provider thru the Cisco UBE in the respective locations.

CUCM: 172.16.101.210
Cisco UBE at Headquarters: 10.30.97.5
Cisco UBE at Branch: 10.40.95.2
SIP SP Softswitch: 10.6.33.32

Configuring Cisco Unified Communications Manage

1. Configuring the System Parameters : Server

Go to Cisco Unified CM > System > Server and Configure the following

· Host Name / IP address : 172.16.101.210

· Description : CUCM - HDQ

2. Configuring the System Parameters : Region

Go to Cisco Unified CM > System > Region and Configure the following

· Name : Region-HQ-IPPhone

· Audio codec : G.729

· Video call Bandwdth

Similarly Create the Region, "Region –Branch-IPPhone" and "Region WAN"

3. Configuring the System Parameters : Device

Go to Cisco Unified CM > System > Device and Configure the following

· Device Pool Name : DevicePool _Region_HDQ

· Region : Region-HQ-IPPhone

· Video call Bandwdth

Similarly Create the Device Pool, "DevicePool_Branch" and "DevicePool_WAN"

4. Configuring the System Parameters : Location

Go to Cisco Unified CM > System > Location and Configure the following

· Name : HUB_HDQ

· Audio Bandwidth

· Video Bandwidth

Similarly Create the Location "HUB_REGION"

5. Configuring the System Parameters : SRST

Go to Cisco Unified CM > System > SRST and Configure the following

· Name : SRST-to-BRANCH1

· Port : 2000

· IP address : 172.16.103.210 (Branch CUBE IP address with SRST enabled)

· SIP Port : 5060

6. Configuring the Call Routing Parameters : Class of control

Go to Cisco Unified CM > Call Routing > Class of control and Configure the following

· Name : Partition_HDQ_IPPhones

· Description : Partition_HDQ_IPPhones

Similarly Create the Partitions, "Partition_Region_IPPhones"

7. Configuring the Call Routing Parameters : Calling Search Space

Go to Cisco Unified CM > Call Routing > Calling search space and Configure the following

· Name : CSS_HDQ_IPPhone

· Description : CSS_HDQ_IPPhone

· Select the Partitions to Route Partitions for the Calling search space

Similarly Create the Calling search space, "CSS_Branch_IPPhone".

8. Configuring the Device Parameters : Gateway

Go to Cisco Unified CM > Device> Gateway and Configure the following

· Product : Cisco 3845

· Gateway : 3845BR

· Protocol : MGCP

· Domain Name : 3845BR

· Description : Branch1 GW ( located at the Branch office )

9. SIP Trunk Creation from CUCM to Service Provider

9.1. Configuring the Device Parameters : Trunk

Go to Cisco Unified CM > Device> Trunk and Configure the following

· Product : SIP Trunk

· Device Protocol : SIP

· Device Name : GW-SIP-HQ

· Device Pool : DevicePool_Region_HDQ

· Class classification : Use system default

· Location : HUB_HDQ

· Under the section Inbound calls : Calling Search Space – CSS_HDQ_IPPhone

SIP Trunk Creation from CUCM to Gateway at Headquarters IP : 10.30.97.5 and then Call Routed to Service Provider IP : 10.6.33.32

SIP Trunk Creation from CUCM to Gateway at Branch IP : 10.40.95.2 and then Call Routed to Service Provider IP : 10.6.33.32

· Product : SIP Trunk

· Device Protocol : SIP

· Device Name : GW-SIP-BR

· Device Pool : DevicePool_Branch

· Class classification : Use system default

· Location : HUB_REGION

· Under the section Inbound calls : Calling Search Space – CSS_Branch_IPPhone

10. Configuring the Route Pattern for Routing the Calls Towards the SIP Service Provider

10.1. Configuring the Call Routing Parameters : Route/Hunt

Go to Cisco Unified CM > Call Routing > Route/Hunt and Configure the following

Creating Route Pattern for long distance calling from Headquarters

· Route Pattern : 98400xxxxx

· Route Partition : Partition_HDQ_IPPhones

· Description : Longdistance call to SIP SP

· Gateway/Routelist : GW-SIP-HQ( GW: 10.30.97.5, Refer Section 9 : Configuring the Device Parameter :Trunk )

Creating Route Pattern for long distance calling from Branch

· Route Pattern : 98400xxxxx

· Route Partition : Partition_Region_IPPhones

· Description : Longdistance call to SP

· Gateway/Routelist : GW-SIP-BR( GW: 10.40.95.2, Refer Section 9 : Configuring the Device Parameter :Trunk )

11. Configuring the Device Parameters : Phone

Go to Cisco Unified CM > Device> Phone and Configure the following

Create IP Phones in the Headquarters

· MAC Address

· Directory Number : 4155000

· Device Pool : DevicePool_Region_HDQ

· Calling search space : CSS_HDQ_IPPhone

Create IP Phones in the Branch

· MAC Address

· Directory Number : 4156000

· Device Pool : DevicePool_Branch

· Calling search space : CSS_Branch_IPPhone

Configuring the Cisco Unified Border Elements

Configure the Dial-peer for Routing Long distance Outgoing Call from CUCM To SIP Service Provider

CUCM IP : 172.16.101.210
SIP Service Provider IP : 10.6.33.32

Sample Dial-Peer Configuration for Routing Calls to SIP S

Sample CUBE-HQ Configuration
CUBEHQ # Voice service voip allow-connections sip to sip ! dial-peer voice 3000 voip description Incoming Dial-Peer codec g729r8 session protocol sipv2 incoming called-number 4155T dtmf-relay rtp-nte digit-drop no vad dial-peer voice 3001 voip description Outgoing Dial-Peer-CUCM destination-pattern 4155T codec g729r8 max-redirects 5 session protocol sipv2 session target ipv4:172.16.101.210 dtmf-relay rtp-nte digit-drop no vad dial-peer voice 3003 voip description Incoming Dial-Peer codec g729r8 session protocol sipv2 incoming called-number .T dtmf-relay rtp-nte digit-drop no vad ********* Routing 98400xxxxx towards SIP Service Provider IP : 10.6.33.32 *********** dial-peer voice 4000 voip description Outgoing Dial-Peer to SIP SP destination-pattern 98400xxxxx codec g729r8 voice-class sip early-offer forced max-redirects 5 session protocol sipv2 session target ipv4:10.6.33.32 dtmf-relay rtp-nte digit-drop no vad
Note:- This section covers only the Dial-Peer Configuration for routing calls towards the SIP Service Provider

Sample Dial-Peer Configuration of CUBE located at Branch office

Sample CUBE-Branch Configuration
BRGW # voice service voip address-hiding allow-connections sip to sip ! mgcp mgcp call-agent 172.16.101.210 2427 service-type mgcp version 0.1 mgcp dtmf-relay voip codec all mode out-of-band mgcp sdp simple mgcp fax t38 inhibit mgcp bind control source-interface GigabitEthernet0/1.1 mgcp bind media source-interface GigabitEthernet0/1.1 ********* Routing 98400xxxxx towards SIP Service Provider IP : 10.6.33.32 *********** dial-peer voice 2000 voip description Outgoing Dial-Peer to SIP SP destination-pattern 98400xxxxx codec g729r8 voice-class sip early-offer forced max-redirects 5 session protocol sipv2 session target ipv4:10.6.33.32 dtmf-relay rtp-nte digit-drop no vad dial-peer voice 5000 voip description Incoming Dial-Peer codec g729r8 session protocol sipv2 incoming called-number 4156T dtmf-relay rtp-nte digit-drop no vad dial-peer voice 5001 voip description Outgoing Dial-Peer-CUCM destination-pattern 4156T codec g729r8 max-redirects 5 session protocol sipv2 session target ipv4:172.16.101.210 dtmf-relay rtp-nte digit-drop no vad dial-peer voice 5000 voip description Incoming Dial-Peer codec g729r8 session protocol sipv2 incoming called-number .T dtmf-relay rtp-nte digit-drop no vad
Note:- This section covers only the Dial-Peer Configuration for routing calls towards the SIP Service Provider