Application Note
Integra Telecom SIP Trunking: Connecting Cisco Unified
Communications Manager 8.5(1) via the Cisco Unified Border
Element using SIP
June 3, 2011
Table of Contents
Introduction .............................................................................................................................................................................................................. 2
Network Topology .................................................................................................................................................................................................... 3
System Components ................................................................................................................................................................................................. 3
Hardware Components ........................................................................................................................................................................................ 3
Software Requirements ....................................................................................................................................................................................... 4
Features .................................................................................................................................................................................................................... 5
Features Supported .............................................................................................................................................................................................. 5
Features Not Supported ....................................................................................................................................................................................... 5
Caveats ..................................................................................................................................................................................................................... 5
Call Flows ................................................................................................................................................................................................................. 6
Configuration ............................................................................................................................................................................................................ 7
Configuring Cisco Unified Border Element ........................................................................................................................................................ 7
Configuring the Cisco Unified Communications Manager ............................................................................................................................... 12
(Optional) Configuring a Cisco 3845 for analog fax ......................................................................................................................................... 30
Acronyms ............................................................................................................................................................................................................... 31
Important Information ............................................................................................................................................................................................ 32
© 2010 Cisco Systems, Inc. All rights reserved.
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Page 1 of 33
Note: Testing was conducted in tekVizion lab.
Introduction
Service Providers today, such as Integra Telecom, are offering alternative methods to connect to the PSTN via their IP network. Most of these
services utilize SIP as the primary signaling method and a centralized IP to TDM gateway to provide on-net and off-net services. Integra
Telecom SIP Trunking is a service provider offering that allows connection to the PSTN and may offer the end customer a viable alternative to
traditional PSTN connectivity via either Analog or T1 lines. A demarcation device between these services and customer owned services is
recommended. The Cisco Unified Border Element provides demarcation, security, interworking and session management services.
This application note describes how to configure a Cisco Unified Communications Manager (Cisco UCM) 8.5(1) with a Cisco Unified
Border Element (Cisco UBE) for connectivity to the Integra Telecom SIP Trunking service. The deployment model covered in this
application note is CPE (Cisco UCM 8.5(1)/Cisco UBE) to PSTN via Integra Telecom SIP Trunking. This document does not
address 911 emergency outbound calls. For 911 feature service details contact Integra Telecom, directly.
Testing was performed in accordance to Cisco’s Service Provider SIP Trunk Validation Test Plan and all features were verified. Key
features verified are:
Basic Calls
Basic Calls with Calling Name and Number as allowed or restricted
DTMF Relay
Call Conference (Intra-site, PSTN)
Call Transfer (Blind, Attended, Early Attended)
Hold and Resume
Voice Mail
T.38 Fax G3/SG3
Simultaneous Calls
Auto Attendant
International Calls
G.711 Fax G3/SG3
Call Forwarding – Find Me (Unconditional, Busy, No Reply)
Codec negotiation
Dial Plans
PRACK with SDP early-media cut-through
The Cisco Unified Border Element configuration detailed in this document is based on a lab environment with a simple dial-plan used to
ensure proper interoperability between Integra Telecom’s SIP network and Cisco Unified Communications. The configuration described
in this document details the important commands to have enabled for interoperability to be successful and care must be taken, by the
network administrator deploying Cisco UBE, to ensure these commands are set per each dial-peer requiring to interoperate to Integra
Telecom’s SIP network.
This application note does not cover the use of calling search spaces (CSS) or partitions on Cisco Unified Communications Manager. To
understand and learn how to apply CSS and partitions refer to the cisco.com link below:
http://www.cisco.com/en/US/partner/docs/voice_ip_comm/cucm/admin/8_0_2/ccmsys/a03ptcss.html
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Page 2 of 33
Note: Testing was conducted at tekVizion Labs®.
Network Topology
Figure 1. Lab Network Topology
System Components
Hardware Components
Cisco 2811 (used as Cisco Unified Border Element)
Cisco 3845 with VIC-4FXS/DID (used for analog fax and hardware MTP)
Cisco Unified Communications Manager (single-node cluster consisting of one Cisco MCS 7800 Series server)
Cisco IP Phones
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Page 3 of 33
Note: Testing was conducted at tekVizion Labs®.
Software Requirements
Cisco Unified Communications Manager 8.5.1.10000-26
Cisco Unified Border Element, IOS version 15.1(3)T (c2800nm-spservicesk9-mz.151-3.T.bin)
Cisco 3845 router, IOS version 15.1(3)T (c3845-adventerprisek9-mz.151-3.T.bin)
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Page 4 of 33
Note: Testing was conducted at tekVizion Labs®.
Features
Features Supported
Voice calls using G.729 and G.711 codecs
RFC 3261 support
Early media cut-through with DTMF relay before 200 OK
Calling number presentation / restriction
Call conferencing
Call transfer (attended and unattended)
Call hold and resume
Call forwarding
DTMF relay (RFC 2833)
T.38 fax
G.711 pass-through fax
Features Not Supported
Caller ID update via SIP UDATE method
Caveats
When a PSTN to CPE call is transferred by the CPE to a second PSTN number, the outgoing call leg fails due to CUBE bug in which
CUBE fails to offer the correct codec list to the service provider.
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Page 5 of 33
Note: Testing was conducted at tekVizion Labs®.
Call Flows
In the sample configuration presented here, CUCM is provisioned with four-digit directory numbers corresponding to the last four DID digits.
No digit manipulation is performed on the CUBE.
For incoming PSTN calls, the CUBE presents the full ten-digit DID number to CUCM. A CUCM Translation Pattern strips all but the last four
digits and routes the call based on those digits. Voice calls are routed to IP phones; fax calls are routed via a 4-digit route pattern to a SIP trunk
that terminates on the router hosting the analog fax endpoint.
CPE callers make outbound PSTN calls by dialing a “90” prefix followed by the destination number. For outbound fax calls from the analog fax
endpoint, the router hosting that endpoint delivers all digits including the “90” prefix. A “90.@” route pattern strips the prefix and routes the call
with the remaining digits via a SIP trunk terminating on the CUBE.
Figure 2. Outbound Voice Call
Figure 3. Inbound Voice Call
Figure 4. Outbound fax call
Figure 5. Inbound fax call
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Page 6 of 33
Note: Testing was conducted at tekVizion Labs®.
Configuration
Configuring Cisco Unified Border Element
Critical commands are marked in Bold with footnotes at the bottom of the page
Version Information:
Cisco IOS Software, 2800 Software (C2800NM-SPSERVICESK9-M), Version 15.1(3)T, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2010 by Cisco Systems, Inc.
Compiled Mon 15-Nov-10 21:41 by prod_rel_team
ROM: System Bootstrap, Version 12.4(13r)T11, RELEASE SOFTWARE (fc1)
MTPnetB uptime is 7 weeks, 4 days, 11 hours, 38 minutes
System returned to ROM by power-on
System image file is "flash:c2800nm-spservicesk9-mz.151-3.T.bin"
Last reload type: Normal Reload
This product contains cryptographic features and is subject to United
States and local country laws governing import, export, transfer and
use. Delivery of Cisco cryptographic products does not imply
third-party authority to import, export, distribute or use encryption.
Importers, exporters, distributors and users are responsible for
compliance with U.S. and local country laws. By using this product you
agree to comply with applicable laws and regulations. If you are unable
to comply with U.S. and local laws, return this product immediately.
A summary of U.S. laws governing Cisco cryptographic products may be found at:
http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
If you require further assistance please contact us by sending email to
export@cisco.com.
Cisco 2811 (revision 53.51) with 512000K/12288K bytes of memory.
Processor board ID FTX1411ALT7
2 FastEthernet interfaces
DRAM configuration is 64 bits wide with parity enabled.
239K bytes of non-volatile configuration memory.
126000K bytes of ATA CompactFlash (Read/Write)
License Info:
License UDI:
-------------------------------------------------
Device# PID SN
-------------------------------------------------
*0 CISCO2811 FTX1411ALT7
Configuration register is 0x2102
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Page 7 of 33
Note: Testing was conducted at tekVizion Labs®.
Running Configuration:
version 15.1
service config
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname MTPnetB
!
boot-start-marker
boot system flash c2800nm-spservicesk9-mz.151-3.T.bin
boot-end-marker
!
!
logging buffered 51200 warnings
no logging console
!
aaa new-model
!
!
!
!
!
!
!
aaa session-id common
clock timezone CST -6 0
clock summer-time CDT recurring
!
dot11 syslog
ip source-route
!
!
ip cef
!
!
!
ip domain name lab.tekvizion.com1
ip name-server 10.64.1.32
no ipv6 cef
multilink bundle-name authenticated
!
!
!
!
!
!
voice rtp send-recv3
!
voice service voip
ip address trusted list4
ipv4 207.173.7.100
ipv4 10.70.18.2
allow-connections sip to sip5
1
Optional. The domain name must match the CUCM enterprise parameter “Cluster Fully Qualified Domain Name” and must
resolve to a DNS A-record. See “DNS Configuration” and “CUCM Configuration” below
2
IP address of DNS server
3
Establish a 2-way voice path when a Real Time Protocol channel is opened
4
Required for toll fraud prevention. IP addresses listed here are those of trusted VoIP peers.
© 2011 Cisco Systems, Inc. All rights reserved.
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Page 8 of 33
Note: Testing was conducted at tekVizion Labs®.
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw6
sip
early-offer forced7
midcall-signaling passthru8
!
voice class codec 19
codec preference 1 g729br8
codec preference 2 g729r8
codec preference 3 g711ulaw
!
!
!
!
!
voice-card 0
!
crypto pki token default removal timeout 0
!
!
!
!
license udi pid CISCO2811 sn FTX1411ALT7
username cisco privilege 15 password 0 cisco
!
!
!
!
!
!
!
interface FastEthernet0/0
ip address 10.70.18.15 255.255.255.0
duplex full
speed 100
!
interface FastEthernet0/1
description This IP address is associated with IntegraTelecom.
ip address 174.46.0.150 255.255.255.128
duplex auto
speed auto
!
ip forward-protocol nd
no ip http server
no ip http secure-server
!
!
ip route 0.0.0.0 0.0.0.0 174.46.0.129
!
logging esm config
!
!
!
!
control-plane
!
!
5
Allow SIP to SIP call processing.
6
Enable T.38 fax relay. Remaining parameters are optional; this example disables redundancy for both low-speed T.30 messaging
and high-speed page data, and enables G.711 fallback if T.38 negotiation fails.
7
Configures CUBE to send a SIP INVITE with SDP on an outbound call leg (Delayed Offer to Early Offer)
8
Enable support for SIP re-INVITE supplementary services
9
Codec preference list, referenced from the dial-peers with the “voice-class codec n” configuration command
© 2011 Cisco Systems, Inc. All rights reserved.
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Page 9 of 33
Note: Testing was conducted at tekVizion Labs®.
!
!
mgcp profile default
!
sccp local FastEthernet0/0
!
dial-peer voice 100 voip
description 1+10 digits to IntegraTelecom
destination-pattern ^1[2-9]..[2-9]......$10
session protocol sipv2
session target sip-server11
voice-class codec 112
voice-class sip early-offer forced13
dtmf-relay rtp-nte14
!
dial-peer voice 102 voip
description International Calls to IntegraTelecom
destination-pattern ^011T15
session protocol sipv2
session target sip-server
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
!
dial-peer voice 300 voip
description 503548XXXX calls to CUCM
huntstop
destination-pattern ^503548....$16
session protocol sipv2
session target dns:clus3pubsub.lab.tekvizion.com17
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
!
dial-peer voice 20 voip
description all calls from CUCM
huntstop
session protocol sipv2
session target dns:clus3pubsub.lab.tekvizion.com
incoming called-number .T18
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte
!
dial-peer voice 10 voip
description calls from Service Provider
session protocol sipv2
session target sip-server
incoming called-number .T
voice-class codec 1
10
Outbound dial peer for 1+10-digit called numbers
11
Sip-server resolves to the Integra Telecom SBC IP address, and is defined in the sip-ua section below.
12
Reference to codec preference list “voice class codec 1”, above
13
Enable delayed-offer to early-offer
14
Forward DTMF tones by using RTP with the Named Telephone Event (NTE) payload type (RFC 2833).
15
Outbound dial peer for international calls (US market)
16
Outbound dial peer (to CUCM). The pattern here should match service provider-assigned DID numbers.
17
Reference to DNS A record. In a single-subscriber deployment, this could either be a DNS reference, or it could be an IP
address: “ipv4:xxx.xxx.xxx.xxx”. In a multiple-node cluster, this would be a reference to a DNS SRV record.
18
This is the default inbound dial peer
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Page 10 of 33
Note: Testing was conducted at tekVizion Labs®.
voice-class sip early-offer forced
dtmf-relay rtp-nte
!
!
gateway
timer receive-rtp 1200
!
sip-ua
authentication username 5035489480 password 7 131504411E08543A733C3C6060364219
retry invite 7
retry bye 7
retry cancel 7
retry register 10
sip-server ipv4:207.173.7.100:506020
!
!
!
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
exec-timeout 0 0
password *********
transport input all
!
scheduler allocate 20000 1000
end
19
Authentication credentials provided by service provider
20
Service provider signaling address.
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Page 11 of 33
Note: Testing was conducted at tekVizion Labs®.
Configuring the Cisco Unified Communications Manager
Figure 6. Enterprise Parameters
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Page 12 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 7. SIP Trunk Security Profile
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Page 13 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 8. SIP Trunk to Integra Telecom via CUBE
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Page 14 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 9. SIP Trunk to Integra Telecom via CUBE (cont.)
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Page 15 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 10. SIP Trunk to Integra Telecom via CUBE (cont.)
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Page 16 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 11. SIP Profile
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Page 17 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 12. SIP Profile (cont.)
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Page 18 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 13. SIP Profile (cont.)
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Page 19 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 14. Translation Pattern for incoming calls from Integra Telecom
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Page 20 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 15. Translation Pattern for incoming calls from Integra Telecom (cont.)
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Page 21 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 16. Route Pattern Configuration for SIP trunk to Integra Telecom via CUBE
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Page 22 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 17. Route Pattern Configuration for SIP trunk to Integra Telecom via CUBE (cont)
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Page 23 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 18. SIP Trunk to Cisco 3845 (for analog fax)
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Page 24 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 19. SIP Trunk to Cisco 3845 (for analog fax) (cont)
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Page 25 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 20. SIP Trunk to Cisco 3845 (for analog fax) (cont)
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Page 26 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 21. SIP Trunk to Cisco 3845 (for analog fax) (cont)
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Page 27 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 22. Route Pattern Configuration (fax)
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Page 28 of 33
Note: Testing was conducted at tekVizion Labs®.
Figure 23. Route Pattern Configuration (fax) (cont)
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Page 29 of 33
Note: Testing was conducted at tekVizion Labs®.
(Optional) Configuring a Cisco 3845 for analog fax
Cisco 3845 configuration for analog fax endpoint
Note: only the configuration elements relevant to the fax endpoint and the SIP trunk to CUCM are shown
dial-peer voice 400 pots
description POTS dial peer for analog fax endpoint
destination-pattern 948521
fax rate voice
port 0/0/0
!
dial-peer voice 201 voip
description Outbound dial peer for PSTN fax clls via CUCM
destination-pattern 90T22
session protocol sipv2
session target dns:clus3pubsub.lab.tekvizion.com
dtmf-relay rtp-nte
codec g711ulaw
fax-relay sg3-to-g323
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw24
no vad
!
dial-peer voice 300 voip
description incoming dial peer for fax calls
incoming called-number 9485
codec g711ulaw
fax-relay sg3-to-g3
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
21
CUCM sends 4-digit extension for inbound fax calls to this endpoint.
22
For fax calls to PSTN caller dials 9+PSTN number. CUCM strips the “9” before forwarding to CUBE.
23
Enable sg3 spoofing.
24
Enable T.38 for fax. Fall back to G.711 if T.38 is unsuccessful.
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Page 30 of 33
Note: Testing was conducted at tekVizion Labs®.
Acronyms
Acronym Definitions
SIP Session Initiation Protocol
MGCP Media Gateway Control Protocol
SCCP Skinny Client Control Protocol
Cisco UCM Cisco Unified Communications Manager
Cisco UBE Cisco Unified Border Element
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Page 31 of 33
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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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Page 32 of 33
Note: Testing was conducted at tekVizion Labs®.
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Headquarters Headquarters Headquarters Headquarters
Cisco Systems, Inc. Cisco Systems International Cisco Systems, Inc. Cisco Systems, Inc.
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USA Haarlerbergweg 13-19 USA #22-01 to #29-01
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Tel: 408 526-4000 The Netherlands Tel: 408 526-7660 www.cisco.com
800 553-NETS (6387) www-europe.cisco.com Fax: 408 527-0883 Tel: +65 317 7777
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Fax: 31 0 20 357 1100
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Work, Live, Play, and Learn is a service mark of Cisco Systems, Inc.; and Access Registrar, Aironet, BPX, Catalyst, CCDA, CCDP, CCVP, CCIE, CCIP,
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are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the United States and certain other countries.
All other trademarks mentioned in this document or Website are the property of their respective owners. The use of the word partner does not imply a
partnership relationship between Cisco and any other company. (0705R)
Printed in the USA
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Page 33 of 33
Note: Testing was conducted at tekVizion Labs®.
