Outsourcing your TDM Gateways: SIP Trunking as a Service Provider Cloud Service

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#CNSF2011

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ABSTRACTSIP Trunking is beginning to become a widely deployed offering from Service Providers (SP).One way of looking at SIP Trunking is outsourcing the essential feature of TDM interconnection from an "on premise" TDM gateway to a CLOUD SERVICE from your SP. With the increased prevalence of customers deploying SIP Trunking, it is important to understand what is required to successfully deploy this service and where the future of SIP Trunking is heading.In this session you will learn about how SP offer SIP Trunking Services and what is required for customers to successfully deploy this new CLOUD service.

Cloud?


Cloudis often touted as “the next best thing since sliced bread”

“Cloud Computing will cause a radical shift in IT” – CIO Survey


Cloud is a new computing paradigm. In Cloud, IT resources and services are abstracted from the underlying infrastructure and provided on-demand and at scale in a multi-tenant environment. Cloud has several characteristics:

• Information technology, from infrastructure to applications, is delivered and consumed as a service over the network• Services operate consistently, regardless of the underlying systems• Capacity and performance scale to meet demand and are invoiced by use• Services are shared across multiple organizations, allowing the same underlying systems and applications to meet the demands of a variety of interests, simultaneously and securely • Applications, services, and data can be accessed through a wide range of connected devices (e.g., smart phones, laptops, and other mobile internet devices)

http://www.cisco.com/en/US/netsol/ns976/index.html

http://www.cisco.com/en/US/netsol/ns976/index.html�


Amazon’s Trouble Raises Cloud Computing Doubthttp://www.nytimes.com/2011/04/23/technology/23cloud.html

Does Cloud Computing Mean More Risks to Privacyhttp://bits.blogs.nytimes.com/2009/02/23/does-cloud-computing-mean-more-risks-to-privacy/

Cloud Computing Is for the Birdshttp://www.businessweek.com/debateroom/archives/2011/05/cloud_computing_is_for_the_birds.html

http://www.nytimes.com/2011/04/23/technology/23cloud.html�

http://www.nytimes.com/2011/04/23/technology/23cloud.html�

http://bits.blogs.nytimes.com/2009/02/23/does-cloud-computing-mean-more-risks-to-privacy/�















http://www.businessweek.com/debateroom/archives/2011/05/cloud_computing_is_for_the_birds.html�


In House / Buying

Cloud / Leasing


• Better value to end customers vs an in house solution qLower price of SIP Trunks and SBCs then TDM gateways and TDM Trunks

• Improved redundancy vs an in house solutionqHigher call completion rate and higher uptime then TDM connections

• Easy to debug/support technical issuesqService must result in less headaches in LONG run then in house solutions

• SP can make money on offering serviceqEssential to ensure investment level required to maintain quality

#CNSF2011


BUYING TDM Trunking Gateways

1. Customizable VoIP protocol2. Purchase TDM Gateway hardware

upfront3. Single Purpose equipment4. Mature Technology

OUTSOURING your TDM Gateways, and Buying SIP Trunking Service

1. Standard service based on SIP2. Purchase Enterprise SBC upfront3. Multipurpose equipment4. Cutting edge technology compared

with TDM interconnect

Translating this into SIP Trunking


Is outsourcing TDM Gateway by moving to SIP Trunking a good choice for a CLOUD Service ?

YES

1. Service Providers (SP) have a great deal of experience with TDM to IP Gateways

2. The service can be MORE reliable then traditional TDM Gateways on premise

3. The service is technologically more efficient (ie fewer IP translations) , which means quality improves

4. SP can offer additional service on top base service to increase their value add

5. SP can scale and monetize this service, they understand and have capacity to bill the service and as such will make the investments in this service

6. The function is not core to a high quality Enterprise UC deployment



Enabling Business-to-Business Collaboration

• Changing Landscapes – VoIP Islands to VoIP Interconnects

• Unified communications SIP Trunks to destinations beyond the Enterprise

IPA IP A

Enterprise Domain 1 Enterprise Domain 2

Narrowband voice to Rich-media Interconnect

A A

Enterprise Domain 1

SP VoIPSBC SBCCUBE CUBE

§ Extend rich-media collaboration to vendors, partners and customers

§ A Cisco Unified Border Element (CUBE) provides b2b interconnectivity for secure rich-media services

IP IP

Enterprise Domain 2


A

ACVP

Branch Offices

Campus Contact Center

A

ACVP

SP SIP

A

ACVP

SP SIP

1. TDM Trunking – Yesterday

2. TDM and IP Trunking – Today

3. IP Trunking – TomorrowCampus Contact Center

Campus Contact Center

Branch Offices

Branch Offices


§ Redundant

§ Complex

§ Inefficient

PBX

PBX

PBX

Apps

PBXPBX

PBX

Apps

Apps

Apps

AppsSBC

SBC

SBC

PSTN gwy

PSTN gwy

PSTN gwy

§ Expensive

§ Limited Features

PSTN Tolls Disparate PBXs Integration with Applications

Mobility

§ Expensive

§ Inflexible

§ Server Intensive

Social networking

Enterprise apps

Video


Simplifyby Streamlining Services

Aggregation

Set upfor Future

by Extending to Collaborative Services

Saveby Efficiently Interconnecting

networks

$


§ Enterprise SBC based on ISR G2 or ASR

§ Device Reuse

§ Device consolidation

§ Optional Session Manager

§ Centralization

§ Application integration

CUBE

Unified SME


Today, more Service providers

offer SIP Trunking service in a much wider footprint then ever before

Many of you may already have the Session Border Controller and SIP Gateways

already in your Routers

SP Competition is beginning to driving new fee schedules and lower rates SIP

Trunking driving huge telecom

savings

SME allows customers to

capture savings early from SIP

Trunking, on non-IP sites, during the IPT roll-out


Capture a 53% cost savings opportunity

Go above and beyond IPT, seize up to 53% savings with SIP, SME and IME


Estimate your own savings potential from SIP Trunking: Use the Model


Areas of cost savings: §Reduction in total number of PSTN circuits §Reduction in amount and cost of hardware needed to terminate circuits§Reduction in unused circuits (sites can share capacity)

Flexible: §New routes/numbers/capacity can be provisioned quickly§Calls can be sent to anywhere that IP network can reach

Less Complex:§ Fewer circuits needed at remote sites (no TDM and IP connection)§ Local, LD, WAN, POTS can all be over same link§ Potential reduction in number of carriers required§ Less conversion needed (Remove’s IP -> TDM conversion at customer site)



Site-SP RTPSite-to-Site RTPMPLS

A

CUBE

A

CUBE

PSTNSP VoIP

PSTN

MPLS

SP VoIP

A

CUBE

Centralized

MPLSA

CUBE

CUBE CUBE CUBE CUBE CUBE

PSTNSP VoIPDistributed

Hybrid


All Calls Routed via a Centralized SIP Trunk

§ CUBE at Headquarters Location§ Each site ports Phone numbers to IP address at

HQ (Phone numbers often ports out of region)

100 ms from Branch to HQ

100 ms from HQ to SP

100 ms from HQ to SP100ms from HQ to Branch-------------------------------------Total Delay for Speech 200ms


All Calls Routed via a Local SIP Trunks

§ CUBE at each regional location§ Each site ports Phone numbers to IP address

at that SITE

100 ms from Branch toSP-------------------------------------------Total Delay for Speech 100ms

100 ms from Branch to SP


Benefits Challenges§ Operational and management savings§ Equipment / Energy savings§ Call routing/dial-plan management§ Consolidating oversubscription§ Potential per-minute call savings

§ Increased campus bandwidth, CAC, latency; media optimization§ Number porting (regions, countries)§ HA in campus (single point of failure)§ Survivability (backup branch call processing)§ Emergency services§ Legal/Regulatory, Geographical

A

Distributed PSTN Trunks

Centralized SIP Trunk

A

SIP SP


Centralized SIP Trunk

§ Consolidated PSTN SIP trunks at HQ site§ Remote sites have SRST for phone backup,

but need TDM access for PSTN backup (1 FXO)§ Lower cost of equipment needed for

termination§ Bandwidth requirement increases as “PSTN”

calls from remote site now traverse WAN§ QoS concerns as RTP for remote site PSTN

calls traverse WAN twice§ Requires porting of all DIDs to aggregated SIP

trunks – geographic and SP challenges§ E911 locations tied to HQ as opposed to phone

location

Distributed SIP Trunks

§ Each site has its own SIP Trunk and CUBE for PSTN access § SIP trunk remains active during SRST§ RTP path is optimized§ Remote site CUBE also acts as local MTP and

SRST router § E911 locations tied to local site and hence

more accurate§ SP and customer need to provision dial plan

correctly to ensure optimal call routing§ Cost may not decrease as dramatically as

centralized solution


• In summary, there are three methods of deploying SIP Trunks today: centralized where trunks for all regions are centralized and provided only from a central location; distributed, where each regional office has SIP Trunk from the providers; and hybrid models where different solutions are provided for different types of traffic

Centralized Location of All SIP Trunks

Distributed Trunks to All Locations

Hybrid Trunk Deployment, Deploy Trunks Based on Function

All PSTN Trunks are removed from remote sites and replaced with SIP Trunks that terminate only at the central datacenter or headquarters. This HQ site receives and routes ALL PSTN traffic via SIP Trunks to Service Provider.

SIP Trunks are provided from the Service Provider to all sites. Each site removes their PSTN access and instead replaces it with SIP Trunks from the provider that terminate at the remote sites. Provider needs to route phone calls to remote site via SIP trunk at remote sites.

SIP Trunks are added in Headquarters and /or remote sites to complement PSTN trunks. Dialplan is altered so that traffic can flow across most effective trunk, and traffic can be effectively routed via both HQ and remote site Trunks.


1. Interoperability with IP PBX2. Fax Calls3. Supplementary Features4. Voice Band Data5. Quality Control


• There is currently no standard for SIP Trunks that can provide the same level of consistency and interoperability of PSTN ISDN Trunks

• There are efforts underway in the industry to have more interoperability; various efforts are being lead by the SIP forum, ATIS, TISPAN

• The problem of interoperability is reduced by having a customer owned border element (CUBE) that can provide signaling interworking/normalization and transcoding

• This problem can be further reduced by having a Service Provider owned Border Element that acts as a demarcation point for signaling

• Customer should test before deployment of their first SIP Trunks solution, and replicate successful deployment procedure to ensure scaling

www.cisco.com/go/interoperability


• Circuit Acceptance Test CasesSP Layer 2 Connection

SP Layer 3 Connection

SP Reachability and Routing

• Connectivity Test CasesRegistration sequence

Session Refresh

Basic outbound/inbound call completion

Quality of Service

Call Admission Control

Management Access

Call Accounting

Voice Quality

FAX Quality

Non-Standard Calls

Stability and Duration

Restart

• SIP Application (Call Flow) Test CasesCaller ID

Codec Negotiation

Call Hold/Resume

Call Forward (Call Forward All to user on PSTN behind SIP Trunk)

Call Transfer

Ad-Hoc Conference

IVR Interaction (Both local and remote IVR)

DTMF

FAX, Mode, TTY

Emergency/911

Call types (Local, Long Distance, International)

• Failover Test CasesLayer 1, 2, 3, 4 failover scenarios

Pg. 243


• SIP Trunks can typically use three different methods to supports FAX calls1. All calls are sent as G7112. Call sends a RE-INVITE to up-speed to G711 when a FAX tone is detected3. T.38 FAX capabilities are exchanged and fax relay is used

• SIP Service provides also occasionally offer a separate fax to -mail service using T.37 Store and Forward fax• Recommend that your SP support T.38

Fax MethodT.38 Fax Capabilities Exchanged

as Part of SIP Messages

All Call Sent as G711Fax Tone Is Detected and RE-INVITE to

up-speed to G711 Is sent

Pros § Highest fax success rates can be achieved§ Cleanest solution from signaling

and media point of view§ Use less bandwidth

than G711§ Fax and Voice calls differentiated

§ Most widely deployed§ Simplest solution

§ Provides benefits of least bandwidth with G729 call initially upspeeding to G711 if call is FAX§ Tone (2100Hz) can be mixed between

Modem and Fax§ Fax Pass-Through

Cons § Degree of interoperability§ Not offered by all Service

Providers

§ Consumes a large amount of bandwidth for all calls§ No ability to distinguish FAX calls from Voice

calls in CDRs

§ Each vendors support of RE-INVITEs is different§ Currently not supported with all Cisco

equipment


• Typical Supplementary ServicesPlacing call on HOLDForward on Busy/No Answer to Number within premiseTransferring call to another extensionCorrect billing for forwarded calls

• Testing of Supplementary Services before deployment is only way to ensure success

Create a test case for each service before deploymentReport findings to Service ProviderDetermine if lack of these functionality should effect deployment

• The supplementary service invoked over the SIP Trunk is not supported or understood by the far end SIP switch

For example, the signaling to place a call on hold and temporarily stop media can be done in one of several ways, all of them are compliant with the standard; mismatching methods may be supported between two SIP switches

PSTN

All Signaling Is Translated Resulting in Fewer Interop Issues

SIP Signaling End-to-End Causes Interop Issues

SIPNetwork

CUBE will resolve interop

issues


• Voice Band Data (VBD) is used to send information such as credit card transactions or alarm system information over slow speed modem connections across the voice channel of an PSTN circuit

• Voice Band Data can work reliably up to 56K with TDM Trunks• With SIP Trunks cannot maintain a PCM clock sync ,so 56K connections are not possible; but medium speed

modem connections are possible over G711 (up to about 26.4K)• With compressed codecs (i.e. G729), you cannot reliable send modem tones over VoIP calls (G711 required)• VBD cannot be “guaranteed”, so an important consideration is whether there are PSTN circuits that can be left

to support this at the site where SIP Trunks are being considered; the most used types of VBD are:• Baudot connections for deaf users

• Credit card validation systems

• Security systems

• Pitney Bowes Postage Machines

• These systems should all be tested before a SIP Trunk for PSTN access is used as a replacement at

Sending a Modem Call Over a Codec Is Like Putting It Through a

Cheese Grater: the Signal Will Never Be

the Same

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• Experience has shown that as customers deployed SIP Trunks for PSTN access, the experience for users has sometimes been “inconsistent” (i.e. one calls is great, next is not great)

• A “best practice” is to create a method of flagging calls that are very bad (either via CDRs/CMRs analysis or user feedback)

• Use data from CDRs/CMRs (i.e. Jitter, Packet Loss) to determine if there are trends; these statistics can be gathered from the Customer premise Border Element (ie CUBE) or CUCM

• Try to determine if quality issues correlate with specific events, such as dialing to some area codes or countries or specific times of day; service providers have different methods of routing that can effect quality

• Service providers should ensure that they have a method of measuring quality all the way to the customer premise; this can be used to distinguish their service from others



Service Offerings

Requirements Unacceptable Offering Good Offering Best Offering

Security None IP Address Validation of SIP INVITES TLS Signed SIP INVITES

Fax None G711 for all Calls T.38 support

Voice Band Data None Offer to work each issue individually

Offers SLA for Data speeds for VDB over SIP Trunks

Uptime None SLA with 95% uptime offeredSLA with 99.999% uptime offered and access

from customer for reports with refunds for nonconformance

Calling Plans N/A Per minuteFlat rate with no cost calls between

customers; each trunk can configure their own calling plan; billing records provided via

WEB interface

Redundancy NoneAbility to route calls to a

different phone number or IP address when trunk is down

Call re-routed in real time when the SIP Trunk fails; routing is to both secondary IP

address and PSTN number

Number Porting NonePorting of phone numbers can be accomplished for some area

codes within 30 daysAll area codes can have phone numbers

ported with zero lost calls in 48 hours

Video Calls N/A Plan to offer video calls Current offering Video calls over SIP Trunks


• TDM PBX that need to access PSTN Phones via SIP Trunks:Voice Gateways act as Network side PRI and send to SIP to the Service Provider

SP IP NetworkCUBE

SIPTDM

SBCTDMTDMTDMTDM

SIP Trunking for TDM PBXs saves money without transitioning TDM PBX to IP PBX•Required a Voice Gateway to translate TDM to SIP•Voice Gateways such as (ISR G2) can support both TDM and SIP Trunks with the same equipment

Make sure your SP offers SIP Trunking for TDM PBXs via Cisco Voice Gateways.



How to Successful adopt a Cloud Service

1. Read and review White papers on Communications Transformation• http://www.cisco.com/en/US/products/sw/voicesw/ps5640/prod_white_papers_list.html

2. Find out who offers Services in your region• Ask current provider or VARs, look at who provides Layer 2 connectivity

3. Understand what your Telecom PSTN costs are• Both costs of connections (T1/E1/Analog) and per minute costs• Use the Cost Estimate

4. Understand what your WAN costs are• Upgrading your IP to “gold” service WAN with your layer 2 providers• IP costs for SIP Trunk are not FREE (as is shown in many ROI calculators),

for toll quality voice

5. Deploy trial with some services• Outbound is easy as it does not require porting of phone numbers• Inbound does require porting of phone numbers to IP addresses and this

may not be as easy as SP promise

6. Monitor quality of deployment and use experience to determine where you want to be in five years—change the world

http://www.cisco.com/en/US/products/sw/voicesw/ps5640/prod_white_papers_list.html�


Enterprise UCM DeploymentA

SBC

TDM-based PSTN

IP Network

Class 4/5SwitchVV

data

voice

VV

TDM Trunk Call PathIP Trunk Call Path

CUBE

SP SBC

1. Initial Deployments have TDM Gateway to Class 4/5 Switch

2. Enterprise SBC is added and connection to SP SIP Trunk is initiated

3. Phone numbers are ported from TDM trunk to IP Trunks



AS5000XM

ASR 1004/6 RP2

Active Voice Call (Session) Capacity

CPS

<5

8-12

50-150+

12-16K+5-30 200-600 600-800 900-1000

3900 ISR G2

2900 ISR G2

17

1500-1700

ASR 1002

3900E ISR G2

2000-2500

20-35

800/1861E

ASR 1001

10-12K

50-100

New Platform

New Platform Even Higher Capacity


An Integrated Network Infrastructure Service

VXML

SRSTRSVP Agent

Cisco Unified Border Element§ Address Hiding§ H.323 and SIP interworking§ DTMF interworking§ SIP security§ Transcoding

Unified CM Conferencing and Transcoding

GK

TDM Gateway§ Voice and Video TDM Interconnect § PSTN Backup

Routing, FW, IPS, QoS

WAN Interfaces

Note: An SBC appliance wouldhave only these features

CUBE

Note: Some features/components may require additional licensing


Security

H.323 and SIPSIP NormalizationDTMF InterworkingTranscodingCodec FilteringFax/Modem Support

InterworkingEncryptionAuthenticationRegistrationSIP ProtectionFW PlacementToll fraud

Session MgmtReal-time session MgmtCall Admissions ControlEnsuring QoSPSTN GW FallbackStatistics and BillingRedundancy/Scalability

DemarcationFault isolation

Topology HidingNetwork Borders

L5/L7 Protocol DemarcStatistics and Billing

Mine

YoursH.323 and SIP

SIP NormalizationDTMF Interworking

TranscodingCodec Filtering

Fax/Modem Support


Continue rich feature development on SIP Interworking and Media Optimizing

CUBE 8.5 Enhancements• Call Preservation with Box to Box Redundancy• Mid Call Codec Renegotiation • Dial Peer Level Bind• RAI in SIP Messages

CUBE 8.6 Enhancements• Registration Proxy support• Full support for UPDATE method• Conditional SIP Profiles

CUBE(Ent) on ASR (RLS 3.2)• H323 to SIP Voice Calls• SIP Video Calls• Scale to 16,000 Calls• Full Stateful failover with Box to Box

Redundancy


Enables rich applications. Affordable for the small branch. Enhanced interoperability.

New Capabilities

• Media forking for call recording on ISR G2

• CUBE functionality extended to 88x/892 platforms

• Improved interoperability including; sRTP-RTP supplementary services; Support for Multi-cast music on hold; Domain based routing; and dynamic REFER handling

• ASR IPv6 improvements: RTCP Pass through and T.38

Customer Benefits

• Enables a simplified, lower cost architecture for call recording

• Makes SIP trunking more cost effective for the small branch/ business

• Improved interworking with SIP trunk service providers and endpoints

Partner Benefits

• Expands the partner business opportunities into recording

• Creates the ability to position CUBE into small deployments



1. New Billing Optionsq “Friends and Family” plans between customersq Flat rate calling throughout Canada

2. New Regions added until all are coveredq Porting number from all areas to single IP addressq SP will start to offer service across multiple countries

3. New redundancy optionsq SP offer the ability to send calls to multiple devices that can

be changed in real time q SP will offer support for Enterprise SBC redundancy

4. New services on top of SIP Trunkingq Managed Enterprise SBC serviceq Outsourced call recordingq Wideband Codec on calls between customersq Video Callsq Call routing of calls to URLs


IP Trunk interconnect

H.323 and now migrating to

SIP

UC Applications Interconnects via

IP

Call Processing: CME, UCM,

etc.

Applications: Webex, Unity,

CVP

Enterprise Trunk via SIP

SP connect and PSTN

access

Enterprise to Enterprise

direct connections

Management parity with

TDM

Collaborative Trunking

Rich audio: wide-band,

noise cancellation

Auto Video Format

Adaptation

Collaboration: Presence,

calendaring. Rich caller ID

Mobility: SNR, B2B

Telepresence

Collaborative Experiences

Real-timevideo/voice translation

Bandwidth Adapting

Collaboration

Device Adapting

Collaboration

Auto Connection Discovery

Today

IT Cost Optimization

Advanced User Experience

Phase 1 Phase 2 Phase 3 Phase 4 Phase 5


• Enhance Feature statically configured based on phone numbers

• Parameters can be dynamically changed to support different environments

Media is processed to improve quality

CUBESIP or TDM Trunk

CUBE Noise Reduction

SIPRTP

Cisco or Non-CiscoContact Center

DSP

Caller in Noise environment

12

3 Called Party hears voice of caller with background noise removed


• Enterprise SBC, such a CUBE will add more capabilities to improve Voice and Video communications

SP IP NetworkCUBE

SIPSIP/H.323ASBC

DSPs ENHANCE AUDIO and VIDEO

TranscodingInput Gain Noise Cancellation Acoustic ShockMedia Forking / RecordingSynthetic Traffic GenerationVideo MixingAcoustic Echo CancellationText OverlayAudio TranscribingVideo improvement/ enhancement

DSP

Shi

ppin

g no

w o

r soo

n


WAN

Enterprise -A

RTPSource CUBE

Destination – Can be any SIP device or Trunk

RTP

SIP SIPEnterprise -B

CUBE

SIP

RTP

CUBE

CUBE Media Forking

•Media Forking results in 2 INVITES and RTP packets from (A) to (B) and (C)

B

C

A

INVITE sip:[email protected]:5060 SIP/2.0Via: SIP/2.0/UDP 9.44.44.9:7654;branch=z9hG4bK-23006-1-0From: sipp <sip:[email protected]:7654>;tag=23006SIPpTag001To: sut <sip:[email protected]:5060>Call-ID: [email protected]: 1 INVITEContact: sip:[email protected]:7654Max-Forwards: 70Call-Info: <sip:10.10.100.200:5060>;purpose=X-cisco-enableforkingSubject: Performance TestContent-Type: application/sdpContent-Length: 172

v=0o=user1 53655765 2353687637 IN IP4 9.44.44.9s=SIP Callc=IN IP4 9.44.44.9t=0 0m=audio 6768 RTP/AVP 8 19a=rtpmap:8 PCMA/8000a=rtpmap:19 CN/8000a=ptime:20

•INCOMING INVITE (A)

CUBE will provide the functionality for NEW RECORDING ARCHITECTURES on SIP Trunks, recording can be done either on premise or as an outsourced CLOUD Service.


2011-2012CUBE Top of Mind for

• Feature equivalence on ASR and ISR G2

• Media Forking on ISR G2

• Mid Call REINVITE consumption

• Noise Cancellation

• Support for MMOH on SIP Trunks

• SME+CUBE Management and Operation

• Acoustic Shock Prevention

• CUBE on 800 Series

• Advanced SRTP to RTP interworking


Headline

Run a trial

• Learn how to configure SIP Trunkingwww.cisco.com/go/interoperability

• Contact your Cisco account team and work on a trial of SIP Trunking

Headline

HeadlineKnow the $$$ impact

• Read the Whitepaperswww.cisco.com/go/cube

• Complete a detailed inventory of TDM Trunking

• Complete a cost model for transitioning from TDM to IP Trunking

Thank you!

Available at Amazon.ca

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

http://www.cisco.com/go/cube�

http://www.amazon.com/gp/reader/1587059444/ref=sib_dp_pt�


#CNSF2011


Document Coverage LocationCUCM 8.x SRND CUCM Connectivity to SIP Trunks cisco.com/go/srnd

- Unified Communications- Unified Communications System- View Design Guide (CUCM 8.x)- Unified Comms Call Routing- Cisco Unified CM Trunks- Cisco Unified Border Element

CVP 7.x SRND Contact Center: CVP + CUBE cisco.com/go/srnd- Unified Communications- Customer Voice Portal- View Design Guide (CVP 7.x)- Gateway Options- Cisco Unified Border Element

CUBE in Contact Center Configuration Guide

Contact Center: CVP + CUBE http://cisco.com/en/US/docs/voice_ip_comm/unified_communications/cubecc.html

SP SIP Trunk Interop CUCM/CUBE Validation testingwith specific SP Offerings:- AT&T TollFree, FlexReach, VoEVPN- Allstream- Verizon- Paetec…

cisco.com/go/interoperabilityCisco Unified Border Element


#CNSF2011

Thank you.

#CNSF2011