SIP Trunking is beginning to become a widely deployed offering from SP. One way of looking at SIP Trunking is outsourcing the essential feature of TDM interconnection from an "on premise" TDM gateway to a service from your SP. With more and more 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 presentation you will learn about how SP offer SIP Trunking Services and what is required for customers to successfully deploy this new Cloud service.

1. dsladden@cisco.com
#CNSF2011
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 1

2. ABSTRACT
SIP 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?
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 2

3. Cloud
is often touted as “the next best thing since sliced bread”
“Cloud Computing will cause a radical
shift in IT” – CIO Survey
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 3

4. 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
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 4

5. Amazon’s Trouble Raises Cloud Computing Doubt
http://www.nytimes.com/2011/04/23/technology/23cloud.html
Does Cloud Computing Mean More Risks to Privacy
http://bits.blogs.nytimes.com/2009/02/23/does-cloud-computing-mean-more-risks-to-privacy/
Cloud Computing Is for the Birds
http://www.businessweek.com/debateroom/archives/2011/05/cloud_computing_is_for_the_birds.html
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 5

6. In House Cloud
/ Buying / Leasing
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 6

7. • 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 solution
qHigher call completion rate and higher uptime then TDM connections
• Easy to debug/support technical issues
qService must result in less headaches in LONG run then in house solutions
• SP can make money on offering service
qEssential to ensure investment level required to maintain quality
#CNSF2011
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 7

8. Translating this into SIP Trunking
BUYING TDM Trunking Gateways OUTSOURING your TDM Gateways,
and Buying SIP Trunking Service
1. Customizable VoIP protocol 1. Standard service based on SIP
2. Purchase TDM Gateway hardware 2. Purchase Enterprise SBC upfront
upfront 3. Multipurpose equipment
3. Single Purpose equipment 4. Cutting edge technology compared
4. Mature Technology with TDM interconnect
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 8

9. 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
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 9

10. © 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 10

11. Enabling Business-to-Business Collaboration
Enterprise Domain 1 Enterprise Domain 2
A A
IP IP
• Changing Landscapes – VoIP Islands to
Narrowband voice to
Rich-media Interconnect § Extend rich-media collaboration to
VoIP Interconnects vendors, partners and customers
• Unified communications SIP Trunks to § A Cisco Unified Border Element
destinations beyond the Enterprise (CUBE) provides b2b interconnectivity
for secure rich-media services
Enterprise
Enterprise
Domain 1
Domain 2
A A
IP SP VoIP IP
CUBE SBC SBC CUBE
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 11

12. 1. TDM Trunking – Yesterday
CVP
A
A
Campus Contact Center
2. TDM and IP Trunking – Today
Branch Offices
SP SIP
CVP
A
A
3. IP Trunking – Tomorrow
Campus Contact Center
CVP Branch Offices
A
A
SP SIP
Campus Contact Center
Branch Offices
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 12

13. PSTN Tolls Disparate PBXs Integration with Applications
PBX
Apps Enterprise apps
SBC Apps
PBX
PBX
SBC PBX
PSTN gwy
PBX
Apps
Apps
Mobility
SBC Apps PSTN gwy
PBX
PSTN gwy
Social networking
§ Expensive § Redundant § Expensive
§ Limited Features § Complex § Inflexible
§ Inefficient § Server Intensive
Video
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 13

14. $
Set up
Save Simplify
for Future
by Efficiently Interconnecting by Streamlining Services
networks Aggregation by Extending to
Collaborative Services
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 14

15. § Enterprise SBC based on ISR
G2 or ASR
§ Device Reuse
§ Device consolidation
CUBE
Unified SME
§ Optional Session Manager
§ Centralization
§ Application integration
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 15

16. Today, more Many of you may
Service providers already have the
offer SIP Session Border
Trunking service Controller and
in a much wider SIP Gateways
footprint then already in your
ever before Routers
SP Competition
SME allows
is beginning to
customers to
driving new fee
capture savings
schedules and
early from SIP
lower rates SIP
Trunking, on non-
Trunking driving
IP sites, during
huge telecom
the IPT roll-out
savings
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 16

17. Go above and beyond IPT, seize up to 53% savings with SIP, SME and IME
Capture a 53% cost savings
opportunity
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 17

18. Estimate
your own
savings
potential
from SIP
Trunking:
Use the
Model
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 18

19. 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)
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 19

20. © 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 20

21. Centralized SP VoIP Distributed SP VoIP PSTN
PSTN
A
A
CUBE MPLS
CUBE MPLS
CUBE CUBE CUBE CUBE CUBE
Hybrid
SP VoIP PSTN
A A
CUBE CUBE
Site-SP RTP
MPLS Site-to-Site RTP
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 21

22. 100 ms from HQ to SP
100ms from HQ to Branch
-------------------------------------
Total Delay for Speech 200ms
100 ms from HQ to
SP
100 ms from Branch
to HQ
§ CUBE at Headquarters Location
§ Each site ports Phone numbers to IP address at
HQ (Phone numbers often ports out of region)
All Calls Routed via a Centralized SIP Trunk
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 22

23. 100 ms from Branch toSP
------
-------------------------------------
Total Delay for Speech 100ms
100 ms from Branch
to SP
§ CUBE at each regional location
§ Each site ports Phone numbers to IP address
at that SITE
All Calls Routed via a Local SIP Trunks
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 23

24. Benefits Challenges
§ Operational and management savings § Increased campus bandwidth, CAC, latency; media
§ Equipment / Energy savings optimization
§ Call routing/dial-plan management § Number porting (regions, countries)
§ Consolidating oversubscription § HA in campus (single point of failure)
§ Potential per-minute call savings § Survivability (backup branch call processing)
§ Emergency services
§ Legal/Regulatory, Geographical
Distributed PSTN Centralized SIP
Trunks Trunk
A A
SIP SP
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 24

25. Centralized SIP Trunk Distributed SIP Trunks
§ Consolidated PSTN SIP trunks at HQ site § Each site has its own SIP Trunk and CUBE for
§ Remote sites have SRST for phone backup, PSTN access
but need TDM access for PSTN backup (1 FXO) § SIP trunk remains active during SRST
§ Lower cost of equipment needed for § RTP path is optimized
termination § Remote site CUBE also acts as local MTP and
§ Bandwidth requirement increases as “PSTN” SRST router
calls from remote site now traverse WAN § E911 locations tied to local site and hence
§ QoS concerns as RTP for remote site PSTN more accurate
calls traverse WAN twice § SP and customer need to provision dial plan
§ Requires porting of all DIDs to aggregated SIP correctly to ensure optimal call routing
trunks – geographic and SP challenges § Cost may not decrease as dramatically as
§ E911 locations tied to HQ as opposed to phone centralized solution
location
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 25

26. • 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
Hybrid Trunk Deployment, Deploy
Centralized Location of Distributed Trunks to
Trunks Based
All SIP Trunks All Locations
on Function
All PSTN Trunks are removed from SIP Trunks are provided from the Service SIP Trunks are added in Headquarters
remote sites and replaced with SIP Provider to all sites. Each site removes and /or remote sites to complement
Trunks that terminate only at the their PSTN access and instead replaces PSTN trunks. Dialplan is altered so
central datacenter or headquarters. it with SIP Trunks from the provider that that traffic can flow across most
This HQ site receives and routes ALL terminate at the remote sites. Provider effective trunk, and traffic can be
PSTN traffic via SIP Trunks to needs to route phone calls to remote site effectively routed via both HQ and
Service Provider. via SIP trunk at remote sites. remote site Trunks.
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 26

27. 1. Interoperability with IP PBX
2. Fax Calls
3. Supplementary Features
4. Voice Band Data
5. Quality Control
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 27

28. • 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
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 28

29. • Circuit Acceptance Test Cases • SIP Application (Call Flow) Test Cases
SP Layer 2 Connection Caller ID
SP Layer 3 Connection Codec Negotiation
SP Reachability and Routing Call Hold/Resume
Call Forward (Call Forward All to user on PSTN behind SIP Trunk)
• Connectivity Test Cases
Call Transfer
Registration sequence
Ad-Hoc Conference
Session Refresh
IVR Interaction (Both local and remote IVR)
Basic outbound/inbound call completion
DTMF
Quality of Service
FAX, Mode, TTY
Call Admission Control
Emergency/911
Management Access
Call types (Local, Long Distance, International)
Call Accounting
Voice Quality • Failover Test Cases
FAX Quality Layer 1, 2, 3, 4 failover scenarios
Non-Standard Calls
Stability and Duration
Restart Pg. 243
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 29

30. • SIP Trunks can typically use three different methods to supports FAX calls
1. All calls are sent as G711
2. Call sends a RE-INVITE to up-speed to G711 when a FAX tone is detected
3. 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
T.38 Fax Capabilities Exchanged Fax Tone Is Detected and RE-INVITE to
Fax Method as Part of All Call Sent as G711 up-speed to G711
SIP Messages Is sent
Pros § Highest fax success rates can be § Most widely deployed § Provides benefits of least bandwidth with
achieved § Simplest solution G729 call initially upspeeding to G711 if
§ Cleanest solution from signaling call is FAX
and media point of view § Tone (2100Hz) can be mixed between
§ Use less bandwidth Modem and Fax
than G711 § Fax Pass-Through
§ Fax and Voice calls differentiated
Cons § Degree of interoperability § Consumes a large amount of bandwidth for all § Each vendors support of RE-INVITEs is
§ Not offered by all Service calls different
Providers § No ability to distinguish FAX calls from Voice § Currently not supported with all Cisco
calls in CDRs equipment
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 30

31. • Typical Supplementary Services
Placing call on HOLD
Forward on Busy/No Answer to Number within premise
Transferring call to another extension
Correct billing for forwarded calls
• Testing of Supplementary Services before deployment is only way to
ensure success
Create a test case for each service before deployment CUBE will
Report findings to Service Provider
resolve interop
Determine if lack of these functionality should effect deployment
issues
• 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
SIP Signaling End-to-End Causes
All Signaling Is Translated Interop Issues
Resulting in Fewer Interop Issues
SIP
PSTN
Network
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 31

32. Sending a Modem Call
Over a Codec Is Like
Putting It Through a
Cheese Grater: the Signal
Will Never Be
the Same
• 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
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 32

33. • 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
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 33

34. © 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 34

35. Service Offerings
Requirements Unacceptable Offering Good Offering Best Offering
IP Address Validation of
Security None TLS Signed SIP INVITES
SIP INVITES
Fax None G711 for all Calls T.38 support
Offer to work each Offers SLA for Data speeds for VDB over SIP
Voice Band Data None
issue individually Trunks
SLA with 99.999% uptime offered and access
Uptime None SLA with 95% uptime offered from customer for reports with refunds for
nonconformance
Flat rate with no cost calls between
customers; each trunk can configure their
Calling Plans N/A Per minute
own calling plan; billing records provided via
WEB interface
Ability to route calls to a Call re-routed in real time when the SIP
Redundancy None different phone number or IP Trunk fails; routing is to both secondary IP
address when trunk is down address and PSTN number
Porting of phone numbers can
All area codes can have phone numbers
Number Porting None be accomplished for some area
ported with zero lost calls in 48 hours
codes within 30 days
Video Calls N/A Plan to offer video calls Current offering Video calls over SIP Trunks
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 35

36. • 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
TDM SIP
SP IP Network
TDM
CUBE SBC
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.
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 36

37. © 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 37

38. 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
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 38

39. Enterprise UCM Deployment
A
voice
Class 4/5
SBC V Switch
CUBE
TDM-based PSTN
1. Initial Deployments have TDM
data
Gateway to Class 4/5 Switch V
2. Enterprise SBC is added and
connection to SP SIP Trunk is
IP Network
initiated
3. Phone numbers are ported from
TDM trunk to IP Trunks
SP SBC
TDM Trunk Call Path
IP Trunk Call Path
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 39

40. © 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 40

41. ASR 1004/6 RP2
50-150+
ASR 1001
50-100
3900E ISR G2
20-35 New Platform
ASR 1002
17
3900 ISR G2
CPS
AS5000XM
2900 ISR G2
8-12
<5
800/1861E
New Platform
Even Higher Capacity
5-30 200-600 600-800 900-1000 1500-1700 2000-2500 10-12K 12-16K+
© 2010 Cisco and/or its affiliates. All rights reserved.
Active Voice Call (Session) Capacity Cisco Confidential 41

42. An Integrated Network Infrastructure Service
TDM Gateway
Cisco Unified Border Element § Voice and Video TDM Interconnect
§ Address Hiding § PSTN Backup
§ H.323 and SIP interworking
§ DTMF interworking
§ SIP security
§ Transcoding
Routing, FW, IPS,
CUBE QoS
Note: An SBC appliance would
have only these features
Unified CM Conferencing
and Transcoding
WAN Interfaces
RSVP SRST
Agent
VXML GK
Note: Some features/components may require additional licensing
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 42

43. Session Mgmt Demarcation
Real-time session Mgmt Fault isolation
Call Admissions Control Topology Hiding
Ensuring QoS Network Borders
PSTN GW Fallback L5/L7 Protocol Demarc
Statistics and Billing Statistics and Billing
Redundancy/Scalability
Security Yours Interworking
Encryption H.323 and SIP
Authentication Mine SIP Normalization
Registration DTMF Interworking
SIP Protection Transcoding
FW Placement Codec Filtering
Toll fraud H.323 and SIP Fax/Modem Support
SIP Normalization
DTMF Interworking
Transcoding
© 2010 Cisco and/or its affiliates. All rights reserved.
Codec Filtering Cisco Confidential 43
Fax/Modem Support

44. Continue rich feature development on SIP Interworking and Media Optimizing
CUBE 8.5 Enhancements
CUBE 8.6 Enhancements
• Call Preservation with Box to Box Redundancy
• Registration Proxy support
• Mid Call Codec Renegotiation
• Full support for UPDATE method
• Dial Peer Level Bind
• Conditional SIP Profiles
• RAI in SIP Messages
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
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 44

45. 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
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 45

46. © 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 46

47. 1. New Billing Options
q “Friends and Family” plans between customers
q Flat rate calling throughout Canada
2. New Regions added until all are covered
q Porting number from all areas to single IP address
q SP will start to offer service across multiple countries
3. New redundancy options
q 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 Trunking
q Managed Enterprise SBC service
q Outsourced call recording
q Wideband Codec on calls between customers
q Video Calls
q Call routing of calls to URLs
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 47

48. Phase 1 Phase 2 Phase 3 Phase 4 Phase 5
UC Applications
IP Trunk Enterprise Trunk Collaborative Collaborative
Interconnects via
interconnect via SIP Trunking Experiences
IP
Call Rich audio:
H.323 and now SP connect Real-time
Processing: wide-band,
migrating to and PSTN video/voice
CME, UCM, noise
SIP access translation
etc. cancellation
Enterprise to
Applications: Auto Video Bandwidth
Enterprise
Webex, Unity, Format Adapting
direct
CVP Adaptation Collaboration
connections
Collaboration:
Management Device
Presence,
parity with Adapting
calendaring.
TDM Collaboration
Rich caller ID
Mobility: SNR, Auto
Today B2B Connection
Telepresence Discovery
IT Cost Optimization
Advanced User Experience
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 48

49. 1 Media is
Caller in Noise processed to 2
environment improve quality
DSP
SIP
CUBE RTP
SIP or TDM Trunk
Cisco or Non-Cisco
CUBE Noise Contact Center
Reduction
Called Party hears
3 voice of caller with
background noise
removed
• Enhance Feature statically configured based on phone numbers
• Parameters can be dynamically changed to support different environments
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 49

50. • Enterprise SBC, such a CUBE will add more capabilities to improve Voice and Video
communications
SIP/H.323 SIP SP IP Network
A
CUBE SBC
DSP
DSPs ENHANCE AUDIO and VIDEO
Shipping now or soon
Transcoding
Input Gain
Noise Cancellation
Acoustic Shock
Media Forking / Recording
Synthetic Traffic Generation
Video Mixing
Acoustic Echo Cancellation
Text Overlay
Audio Transcribing
Video improvement/ enhancement
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 50

51. CUBE Media Forking Destination – Can be any SIP
device or Trunk
Enterprise -B
Enterprise -A SIP SIP
SIP
A B
CUBE CUBE
Source CUBE WAN
RTP RTP
RTP
C
•Media Forking results in 2 INVITES and RTP packets from (A) to (B) and (C)
•INCOMING INVITE (A)
INVITE sip:11111@9.42.30.33:5060 SIP/2.0
Via: SIP/2.0/UDP 9.44.44.9:7654;branch=z9hG4bK-23006-1-0
From: sipp <sip:123@9.44.44.9:7654>;tag=23006SIPpTag001
To: sut <sip:11111@9.42.30.33:5060>
Call-ID: 1-23006@9.44.44.9
CUBE will provide the functionality for NEW
CSeq: 1 INVITE
Contact: sip:123@9.44.44.9:7654
RECORDING ARCHITECTURES on SIP Trunks,
Max-Forwards: 70
Call-Info: <sip:10.10.100.200:5060>;purpose=X-cisco-enableforking
recording can be done either on premise or as an
Subject: Performance Test
Content-Type: application/sdp outsourced CLOUD Service.
Content-Length: 172
v=0
o=user1 53655765 2353687637 IN IP4 9.44.44.9
s=SIP Call
c=IN IP4 9.44.44.9
t=0 0
m=audio 6768 RTP/AVP 8 19
a=rtpmap:8 PCMA/8000
a=rtpmap:19 CN/8000
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 51
a=ptime:20

52. CUBE Top of Mind for
2011-2012
• Feature equivalence on ASR and ISR G2 • SME+CUBE Management and Operation
• Media Forking on ISR G2 • Acoustic Shock Prevention
• Mid Call REINVITE consumption • CUBE on 800 Series
• Noise Cancellation • Advanced SRTP to RTP interworking
• Support for MMOH on SIP Trunks
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 52

53. Available at
Amazon.ca
Headline
Headline Headline
Know the $$$ impact
Run a trial
• Read the Whitepapers • Learn how to configure SIP
www.cisco.com/go/cube Trunking
• Complete a detailed inventory of www.cisco.com/go/interoperability
TDM Trunking • Contact your Cisco account team
• Complete a cost model for and work on a trial of SIP Trunking
transitioning from TDM to IP
Trunking
Thank you!
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 53

54. #CNSF2011
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 54

55. Document Coverage Location
CUCM 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 Contact Center: CVP + CUBE http://cisco.com/en/US/docs/voice_ip_comm/unified_co
Configuration Guide mmunications/cubecc.html
SP SIP Trunk Interop CUCM/CUBE Validation testing cisco.com/go/interoperability
with specific SP Offerings: Cisco Unified Border Element
- AT&T TollFree, FlexReach, VoEVPN
- Allstream
- Verizon
- Paetec
…
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 55

56. #CNSF2011
© 2010 Cisco and/or its affiliates. All rights reserved. Cisco Confidential 56

57. Thank you.
#CNSF2011