CNSF2011- SIP Trunking as a Service Provider Cloud Service- Darryl Sladden

Welcome

Outsourcing your TDM Gateways:
SIP Trunking as a Service Provider Cloud Service
Darryl Sladden, dsladden@cisco.com
Marketing Manager, SRTG
#CNSF2011

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?

Cloud
is often touted as “the next best thing since sliced bread”
Game Changer
Tremendous Cost Savings
“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 demandand 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

Is the CLOUD Ready for Essential Services ?
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

What makes a good CLOUD service ?
Better value to end customers vs an in house solution
Lower price of SIP Trunks and SBCs then TDM gateways and TDM Trunks
Improved redundancy vs an in house solution
Higher call completion rate and higher uptime then TDM connections
Easy to debug/support technical issues
Service must result in less headaches in LONG run then in house solutions
SP can make money on offering service
Essential to ensure investment level required to maintain quality
#CNSF2011

Translating this into SIP Trunking
BUYING TDM Trunking Gateways
Customizable VoIP protocol
Purchase TDM Gateway hardware upfront
Single Purpose equipment
Mature Technology
OUTSOURING your TDM Gateways, and Buying SIP Trunking Service
Standard service based on SIP
Purchase Enterprise SBC upfront
Multipurpose equipment
Cutting edge technology compared with TDM interconnect

Is outsourcing TDM Gateway by moving to SIP Trunking a good choice for a CLOUD Service ?
YES
Service Providers (SP) have a great deal of experience with TDM to IP Gateways
The service can be MORE reliable then traditional TDM Gateways on premise
The service is technologically more efficient (ie fewer IP translations) , which means quality improves
SP can offer additional service on top base service to increase their value add
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
The function is not core to a high quality Enterprise UC deployment

Migrating to the CLOUD

IP
IP
IP
IP
CUBE
CUBE
A
A
A
A
Rich-media
Enabled
Rich-media
Enabled
Rich-media
Enabled
SBC
SBC
SIP Trunking: Eventual solution to allow end to end IP CommunicationsEnabling Business-to-Business Collaboration
Enterprise Domain 1
Enterprise Domain 2
Narrowband voice to
Rich-media Interconnect
Changing Landscapes – VoIP Islands to VoIP Interconnects
Unified communications SIP Trunks to destinations beyond the Enterprise
Extend rich-media collaboration to vendors, partners and customers
A Cisco Unified Border Element (CUBE) provides b2b interconnectivity for secure rich-media services
Enterprise Domain 1
Enterprise Domain 2
SP VoIP

SP SIP
SP SIP
CVP
CVP
CVP
A
A
A
A
A
A
The Migrations to SIP Trunking in the CLOUD
1. TDM Trunking – Yesterday
2. TDM and IP Trunking – Today
Campus
Contact Center
Branch Offices
3. IP Trunking – Tomorrow
Campus
Contact Center
Branch Offices
Campus
Contact Center
Branch Offices

Today’s Communications Network Challenges
Disparate PBXs
Integration with Applications
PSTN Tolls
PBX
Enterprise apps
Apps
Apps
SBC
PBX
PBX
Mobility
PBX
SBC
PSTN gwy
PBX
Social networking
Apps
Redundant
Complex
Inefficient
Expensive
Limited Features
Expensive
Inflexible
Server Intensive
Apps
SBC
Apps
PSTN gwy
PBX
PSTN gwy
Video

SIP Trunking: Three Simple Steps
$
Set up
for Future
by Extending to Collaborative Services
Save
by Efficiently Interconnecting networks
Simplify
by Streamlining Services Aggregation

Overall Architecture for SIP Trunking
Enterprise SBC based on ISR G2 or ASR
Device Reuse
Device consolidation
Optional Session Manager
Centralization
Application integration
CUBE
Unified SME

Why Now ?

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

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

SIP Trunk: The Benefits
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)

SIP Trunking: Deployment Architectures and Issues

SP VoIP
SP VoIP
PSTN
PSTN
Centralized
Distributed
CUBE
CUBE
CUBE
CUBE
MPLS
MPLS
A
A
A
A
SP VoIP
PSTN
CUBE
CUBE
CUBE
CUBE
CUBE
Site-SP RTP
Site-to-Site RTP
MPLS
Centralized and Distributed SIP Trunk Models
Hybrid

Centralized Deployment Model
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

Distributed Deployment Model
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

A
A
SIP SP
Centralized SIP Trunks: Trade-Offs
Distributed PSTN Trunks
Centralized SIP Trunk

Centralized vs. DistributedSIP Trunks designs
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
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

SIP Trunking Deployment Scenarios
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

Top 5 Issues when adopting a SIP Trunks for PSTN Access Service
Interoperability with IP PBX
Fax Calls
Supplementary Features
Voice Band Data
Quality Control

1. Interoperability Issues with SIP Trunks
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

CUCM/CME and CUBE SIP Trunk Interop Test Plan Outline
Circuit Acceptance Test Cases
SP Layer 2 Connection
SP Layer 3 Connection
SP Reachability and Routing
Connectivity Test Cases
Registration 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 Cases
Caller 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 Cases
Layer 1, 2, 3, 4 failover scenarios
Pg. 243

2. Fax Calls
SIP Trunks can typically use three different methods to supports FAX calls
All calls are sent as G711
Call sends a RE-INVITE to up-speed to G711 when a FAX tone is detected
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

3. Supplementary Services
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
Report findings to Service Provider
Determine 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
CUBE will resolve interopissues
SIP Signaling End-to-End Causes Interop Issues
All Signaling Is Translated Resulting in Fewer Interop Issues
SIPNetwork
PSTN

4. Voice Band Data
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

5. Quality Control
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

Evaluating SPs and Migrating
Tips and Recommendations

Evaluating SIP Trunking for PSTN ServicesService Offerings

SP IP Network
SBC
SIP Trunking also can be used for TDM PBXs
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 PBX
SIP
TDM
CUBE
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.

SIP Trunk providers in Canada(+ others)

Migration PlanHow to Successful adopt a Cloud Service
Read and review White papers on Communications Transformation
http://www.cisco.com/en/US/products/sw/voicesw/ps5640/prod_white_papers_list.html
Find out who offers Services in your region
Ask current provider or VARs, look at who provides Layer 2 connectivity
Understand what your Telecom PSTN costs are
Both costs of connections (T1/E1/Analog) and per minute costs
Use the Cost Estimate
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
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
Monitor quality of deployment and use experience to determine where you want to be in five years—change the world

SBC
A
IP Network
SIP Trunks Move from TDM to IP connection for interconnect between SP and Enterprise for Voice traffic
Enterprise UCM Deployment
voice
Class 4/5
Switch
TDM-based PSTN
Initial Deployments have TDM Gateway to Class 4/5 Switch
Enterprise SBC is added and connection to SP SIP Trunk is initiated
Phone numbers are ported from TDM trunk to IP Trunks
data
CUBE
SP SBC
TDM Trunk Call Path
IP Trunk Call Path

CiscoUnifiedBorderElement
Platform and Features

ASR 1004/6 RP2
ASR 1001
3900E ISR G2
ASR 1002
3900 ISR G2
AS5000XM
800/1861E
Cisco Unified Border Element (Enterprise Edition) Portfolio
50-150+
50-100
20-35
New Platform
17
CPS
2900 ISR G2
8-12
<5
Even Higher Capacity
New Platform
10-12K
12-16K+
5-30
200-600
600-800
900-1000
1500-1700
2000-2500
Active Voice Call (Session) Capacity

CUBE
Cisco Unified Border Element—More Than an SBCAn Integrated Network Infrastructure Service
TDM Gateway
Voice and Video TDM Interconnect
PSTN Backup
Cisco Unified Border Element
Address Hiding
H.323 and SIP interworking
DTMF interworking
SIP security
Transcoding
Routing, FW, IPS, QoS
Note: An SBC appliance would have only these features
Unified CM Conferencing and Transcoding
WAN Interfaces
SRST
RSVP Agent
VXML
GK
Note: Some features/components may require additional licensing

Mine
Yours
Cisco Unified Border Element Key Features
Session Mgmt
Demarcation
Fault isolation
Topology Hiding
Network Borders
L5/L7 Protocol Demarc
Statistics and Billing
Real-time session Mgmt
Call Admissions Control
Ensuring QoS
PSTN GW Fallback
Statistics and Billing
Redundancy/Scalability
Security
Interworking
Encryption
Authentication
Registration
SIP Protection
FW Placement
Toll fraud
H.323 and SIP
SIP Normalization
DTMF Interworking
Transcoding
Codec Filtering
Fax/Modem Support
H.323 and SIP
SIP Normalization
DTMF Interworking
Transcoding
Codec Filtering
Fax/Modem Support

Cisco Unified Border Element (CUBE) Features delivered in 2010
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

Cisco Unified Border Element (CUBE)CUBE 8.8 on ISR G2 and CUBE on ASR (RLS 3.3)
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

Roadmap

SIP Trunking Cloud Service Roadmap
New Billing Options
“Friends and Family” plans between customers
Flat rate calling throughout Canada
New Regions added until all are covered
Porting number from all areas to single IP address
SP will start to offer service across multiple countries
New redundancy options
SP offer the ability to send calls to multiple devices that can be changed in real time
SP will offer support for Enterprise SBC redundancy
New services on top of SIP Trunking
Managed Enterprise SBC service
Outsourced call recording
Wideband Codec on calls between customers
Video Calls
Call routing of calls to URLs
Customizable by each Service Provider

Cisco IP Trunking Evolution
Today
IT Cost Optimization
Advanced User Experience

Cisco or Non-Cisco
Contact Center
SIP or TDM Trunk
CUBE
SBC based Noise Reduction with CUBE
1
Media is processed to improve quality
2
Caller in Noise environment
DSP
SIP
RTP
CUBE Noise Reduction
3
Called Party hears voice of caller with background noise removed
Enhance Feature statically configured based on phone numbers
Parameters can be dynamically changed to support different environments

SP IP Network
CUBE
A
SBC
DSPs will change the way SBC are deployed and used – Be ready for these advancements
Enterprise SBC, such a CUBE will add more capabilities to improve Voice and Video communications
SIP/H.323
SIP
DSP
DSPs ENHANCE AUDIO and VIDEO
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
Shipping now or soon

CUBE
CUBE
CUBE Media Forking
Destination – Can be any SIP device or Trunk
Enterprise -B
Enterprise -A
SIP
SIP
SIP
B
A
WAN
Source CUBE
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
CSeq: 1 INVITE
Contact: sip:123@9.44.44.9:7654
Max-Forwards: 70
Call-Info: <sip:10.10.100.200:5060>;purpose=X-cisco-enableforking
Subject: Performance Test
Content-Type: application/sdp
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
a=ptime:20
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.

CUBE Top of Mind for
2011-2012
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

Call to Action
Available at Amazon.ca
Headline
Headline
Headline
Know the $$$ impact
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
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!

Q & A
#CNSF2011

Cisco.com SIP Trunk Design Documents

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www.networkerssolutionsforum.com
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CNSF2011- SIP Trunking as a Service Provider Cloud Service- Darryl Sladden

by Cisco Canada on May 17, 2011

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CNSF2011- SIP Trunking as a Service Provider Cloud Service- Darryl Sladden — Presentation Transcript