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EDCS# 846444 Rev # 2
Application Note
Verizon IP Trunking and IP Contact Center Services: Connecting Cisco Unified Communications Manager 7.1(3) with Cisco Unified Border Element (Enterprise Edition) 1.3
Features............................................................................................................................................................................................................................................................ 5 Features Supported (IP Trunking).............................................................................................................................................................................................................. 5 Features Not Supported (IP Trunking)....................................................................................................................................................................................................... 5 Features Supported (IPCC)........................................................................................................................................................................................................................ 5 Features Not Supported (IPCC).................................................................................................................................................................................................................. 6
Failover ................................................................................................................................................................................................................................................ 6 Example call flow for Voice Calls (G.729)......................................................................................................................................................................................... 7 Example call flow for FAX Calls (G.711ulaw).................................................................................................................................................................................. 7
Route List for FAX: .......................................................................................................................................................................................................................... 20 The previously defined ROUTE GROUP is selected in the FAX Route List (similar to Voice Route List)....................................................................................... 20 Route Pattern for Voice:................................................................................................................................................................................................................... 22 Route Pattern for FAX: .................................................................................................................................................................................................................... 23
Cisco UBE Example Configuration (North America)................................................................................................................................................................................... 34 Configuration of Cisco Unified Border Element (CISCO UBE) 1.3........................................................................................................................................................ 34
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IntroductionIntroductionIntroductionIntroduction
This application note describes how to configure a Cisco Unified Communications Manager (Cisco UCM) 7.1 and Cisco Unified Border Element (Cisco UBE)
Enterprise Edition 1.3 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 (United States, Europe, Middle East and Africa), 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 TrunkingVerizon IP TrunkingVerizon IP TrunkingVerizon IP Trunking Overview Overview Overview 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 offers a native IP Trunking option that provides a SIP trunk directly to your IP PBX, and an IP Integrated Access option that leverages a
gateway device so you can interface with legacy key or PBX systems.
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 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 customer premises equipment (CPE) port capacity. VoIP Inbound includes advanced toll free features -including
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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
System ComponentsSystem ComponentsSystem ComponentsSystem Components
Hardware Components
• Cisco UBE IOS version 12.4(20)T4. 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
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• Ethernet Switch
• WAN router used to terminate the Verizon MPLS network
Software Requirements
• Cisco Unified Call Manager 7.1.3
• Cisco Unified Border Element Cisco UBE running IOS version 12.4(20)T4
FeaturesFeaturesFeaturesFeatures
Features Supported (IP Trunking)
• Voice calls using G.729 codec
• RFC3261 generic feature Support
• Locating SIP Servers via DNS SRV and DNS A Records
• Early Media Cut-Through
• Calling Party Number Presentation and Privacy (P-Asserted Identity)
• FAX calls using G.711ulaw passthrough
• 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 7.1. An MTP resource is required to enable DTMF relay for any calls that utilize a CTI Route-point.
• CISCO UBE performs Delay-Offer-to-Early-Offer interworking of an initial SIP INIVTE from CISCO UCM without SDP
Features Not Supported (IP Trunking)
• 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)
• Mid-call codec negotiation (example: G.729 upspeed to G.711) this capability is not currently supported with CISCO UCM or CISCO UBE.
• Outbound SIP REFER with Replaces. CISCO UCM does not currently support generation of an outbound SIP Refer with replaces messaging.
• CISCO UCM 7.1(3) 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. See configuration section for details
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Features Supported (IPCC)
• RFC3261 generic feature Support
• Locating SIP Servers via DNS SRV and DNS A Records
• Early Media Cut-Through
• Calling Party Number Presentation and Privacy (P-Asserted Identity)
• DTMF as RFC2833 NTE (named telephone events) when a compressed audio codec is used
• Cisco UBE performs Delay-Offer-to-Early-Offer interworking of an initial SIP INIVTE from IPCC to Cisco UCM
Features Not Supported (IPCC)
• Outbound SIP REFER with Replaces. Cisco UCM does not currently support generation of an outbound SIP Refer with replaces messaging.
• Due to the codec negotiation issues for certain IPCC call flows, (Enhanced Transfer) it is necessary to configure the DIDs used for incoming IPCC calls for the G.711ulaw codec only. This will allow all calls presented by IPCC to negotiate a single codec (G.711ulaw) and allow proper media flow when using advanced call transfer services.
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).
Failover
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 Verizon VoIP network.
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Example call flow for Voice Calls (G.729)
Example call flow for FAX Calls (G.711ulaw)
Inbound Call Flows
Inbound calls are received from either the IP Trunking or IPCC services. These services provide a variable length user ID for routing the SIP call. The last 4 digits of the User ID are used to route the call within the IP PBX. The IP PBX then connects the call to the corresponding IP Phone or analog device.
Route Pattern 9@ For FAX Calls
Route List
Route Group
CUCM Cluster
CUBE 2
CUBE
CUBE
A VERIZON VoIP
CUBE 1 SIP Trunk
9 is stripped on FAX calls in CUCM and replaced with 8 8 is stripped in CUBE
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Failover
The VoIP Network sends periodic SIP options messages as a keep alive 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:
1. Configure " monitor probe icmp-ping" to any dial-peers connecting to the CISCO UCM cluster.
2. Add " call fallback monitor" to the global configuration
3. 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 Verizon IP trunking service may time-out and the call would be cancelled before trying the secondary CISCO UBE device.
Cisco UCM will invoke the Session Expires Timer if the SIP Session Timer for all SIP calls. This timer is calculated at seconds/2 and the default value is 1800, with this default timer setting SIP calls may disconnect after 15 Min. As a workaround we set this parameter to the maximum value of 86400 in the CISCO UCM Service Parameters. This allows the SIP call to be active for 12 hours before the CISCO UCM SIP session expires timer engages and disconnects the call.
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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).
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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 “Offsite” region will use G.729 and calls between “Default” and “Offsite” use G.711. Applying this configuration to our testbed, the SIP trunk is placed in
a Device Pool with the “Offsite” 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 “Offsite” region. Devices that belong to the same region are configured to use the G.711 codec
Note: With CISCO UCM 7.1 the system defaults for Intra-Region codec preference is to use the highest quality 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.
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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.
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4. Verify that all devices (Phones and Gateways) that are in the same Region to allow use of the G.711alaw codec. This is similar to the above configuration for FAX
end-points.
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.
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CCCCisco isco isco isco UBE UBE UBE UBE Example C Example C Example C Example Configurationonfigurationonfigurationonfiguration (North America) (North America) (North America) (North America)
Configuration of Cisco Unified Border Element (CISCO UBE) 1.3
Critical commands are marked in Bold with footnotes at bottom of the page
version 12.4 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption service sequence-numbers ! hostname cube.1.3825 ! boot-start-marker boot-end-marker ! card type t1 2 1 logging message-counter syslog logging buffered 100000 no logging console enable password cisco ! no aaa new-model no network-clock-participate slot 2 no network-clock-participate wic 0 ! dot11 syslog ip source-route no ip dhcp use vrf connected ip dhcp excluded-address 192.168.0.0 192.168.0.100 ! ! !
ip dhcp pool IPPHONES1
network 192.168.0.0 255.255.255.0 default-router 192.168.0.10 option 150 ip 192.168.0.6 !
1 (Optional ) DHCP Service: automatically assign IP address and TFTP server (option 150) configuration to IP Phones
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! ip cef ! !
ip domain name pipiptrunksit2.gsiv.com2
ip name-server 166.38.98.2 ip name-server 10.0.1.4 ! no ipv6 cef multilink bundle-name authenticated ! ! ! Voice-card 0 dspfarm dsp services dspfarm ! Voice-card 2 dspfarm ! ! Voice service voip address-hiding
allow-connections sip to sip3
sip
early-offer forced4
localhost dns:ciscocm7.pipiptrunksit2.gsiv.com5
midcall-signaling passthru6
! ! voice class codec 1 codec preference 1 g729r8
2 DNS Domain name for SIP Realm and name server list for DNS resolution 3 Allow SIP to SIP call processing 4 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. 5 Configures global settings for substituting a DNS localhost name in place of the physical IP address in the From, Call-ID, and Remote-Party-ID headers of outgoing messages. 6 Enables support for SIP Supplementary Services (Only used for SIP-to-SIP Calls)
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! ! ip ssh version 2 ! ! ! ! interface GigabitEthernet0/0
description CUBE inside interface9
ip address 192.168.0.10 255.255.255.0 ip virtual-reassembly load-interval 30 duplex auto speed auto media-type rj45 ! interface GigabitEthernet0/1 description CUBE outside interface ip address 172.16.2.10 255.255.255.0 ip virtual-reassembly load-interval 600 duplex auto speed auto media-type rj45 ! ip route 0.0.0.0 0.0.0.0 172.16.2.1 no ip http secure-server ! ! ip rtcp report interval 10000 ! ! ! control-plane !
call fallback monitor10
call treatment on call threshold global cpu-avg low 68 high 75 call threshold global total-mem low 75 high 85
9 No SIP bind commands configured, SIP is sourced from both inside and outside interfaces 10 The call fallback monitor command is used as a statistics collector of network conditions based upon probes. This is required to monitor the status of the remote destination of the CISCO UCM dial-peer.
11 Global Call Admission Control based on Resource utilization 12 Optional FXS port for FAX devices connected directly to the CISCO UBE 13 DSP Resources for Transcoding registered with CISCO UCM cluster
ip qos dscp af32 signaling no vad ! dial-peer Voice 101 voip description INBOUND G729 Voice SIP calls from VerizonB Voice-class codec 1 session protocol sipv2 session target sip-server
14 DSP resources for Conferencing registered with CISCO UCM cluster 15 Strip the leading “9” from outgoing called number 16 Match on outbound calls from CISCO UCM with leading “9” 17 Forwards DTMF tones by using RTP with the Named Telephone Event (NTE) payload type.
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incoming called-number [1-5]...18
dtmf-relay rtp-nte no vad ! dial-peer Voice 102 voip description To/From CUCM subscriber for Voice preference 2 destination-pattern [1-5]... monitor probe icmp-ping Voice-class codec 1 session protocol sipv2 session target ipv4:192.168.0.4
incoming called-number 9T19
FAX rate disable no vad ! dial-peer Voice 103 voip description To/From CUCM publisher for Voice preference 5 destination-pattern 1... monitor probe icmp-ping Voice-class codec 1 session protocol sipv2
session target ipv4:192.168.0.620
incoming called-number 9T dtmf-relay rtp-nte no vad ! dial-peer Voice 200 voip description inbound FAX dial peer from VZ session protocol sipv2 session target sip-server
incoming called-number 101821
18 Enables CISCO UBE to set configuration parameters to incoming calls based on the received called number. This ensures that both legs of the SIP call have matching CODECs 19 Enables CISCO UBE to set configuration parameters to outgoing calls based on the received called number. This ensures that both legs of the SIP call have matching CODECs 20 Multiple SIP trunks configured for redundant connections to the CISCO UCM cluster 21 Match on inbound call-leg for FAX calls. This ensures that both legs of the SIP call have matching CODECs.
22 CODEC is set on dial-peer to force use of g711ulaw for FAX calls. 23 Disables FAX relay transmission capability. FAX-Passthrough is the supported FAX method. 24 Strip leading “8” from outbound FAX calls before sending to VERIZON 25 Match outbound FAX calls from CISCO UCM cluster with leading “8” 26 Enables monitoring of dial-peer targets using ICMP ping. 27 Enables CISCO UBE to set configuration parameters to outgoing calls based on the received called number. This ensures that both legs of the SIP call have matching CODECs
incoming called-number 8T codec g711ulaw FAX rate disable no vad ! dial-peer voice 800 voip description OUTBOUND to VzB IP Toll Free translation-profile outgoing DIGITSTRIP-9 destination-pattern 91800632T codec g711ulaw session protocol sipv2
session target dns:rchtcsd05011.vzbi.com29
dtmf-relay rtp-nte ip qos dscp af32 signaling no vad ! ! ! dial-peer voice 801 voip description G.711 INBOUND from VzB IP Toll Free codec g711ulaw session protocol sipv2 session target sip-server incoming called-number 1212 dtmf-relay rtp-nte no vad ! ! ! ! dial-peer voice 802 voip description G.711 To/From CUCM subscriber IP Toll Free preference 2
28 Multiple SIP trunks configured for redundant connections to the CISCO UCM cluster 29 A unique SIP server is used to route calls to the IPCC service vs. the IP Trunking service.
g729-annexb override ! line con 0 line aux 0 line vty 0 4 ! scheduler allocate 20000 1000 end
30 Sets the value of the SIP status code that is to correspond with the PSTN cause code. 31 SIP Proxy FQDN name for outbound SIP calls to the IP Trunking service
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