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

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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 …

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.

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  • Welcome to this year’s Networkers event! I’d like to call out a few house-keeping items for this year’s conference.If you haven’t already, download the Cisco Mobile Events app at bit.ly/cnsfapp. The app can be used on all types of Smart-phones and android devices.You have an opportunity to earn points, win prizes, view program highlights, link to peers, Cisco experts, and partners. When you click on ‘Check-In’ for the Networkers event, you automatically earn a ‘Canada’ badge! Also, if you are engaged in Social Media, join the Networkers 2011 conversation through Twitter, Facebook, and YouTube. Use hash-tag CNSF2011 to be part of the discussion.
  • Sip trunks are more flexible than tdm trunks. Tdm can’t do video, hd audio (high bandwidth). TDM trunks need to add in multiples of 24 or 32. SIP trunks are added as needed with no number block restrictions, so it’s easier to accomodate high traffic spikes or variances with no over-provisioning.With cisco, existing isr gateways can simply be converted to sip gateways, unlike competitors that require the purchase of whole new gateways. Resource re-use (no new boxes and saves rack space – real estate, cooling savings) and no re-training. Multifunction device.Single vendor (cisco) for allCC: can be more efficiently distributed over the organization, more dynamic, more flexible. (can’t change/modify with changing demand), or for disaster recovery, can’t react as easily/quicklyBetter video through pre-arranged connections or ad-hoc from IMENoise cancellation: (DSP technology) allows higher quality SIP trunks. Built-in. Cisco focuses of high (acoustics) quality for enterprises as a result of the superior DSPFEATURES IN GREEN ARE UNIQUE TO CISCOFEATURES in Italic are FUTURE
  • SME 8.0 offers many enhancements to improve the User/AdminExerienceCisco Intercompany Media Engine – Provides rich B2B Voice and Video CallingRSVP w/ SIP Preconditions– Dynamic end to end call admission controlService Advertisement Framework – Call Control Discovery – Automatic provision of SIP routes in SME. Cisco Unified Routing Rules Interface - API for call handling based on policy rulesSME 8.5 EnhancementsVideo Support – Interop with VCS . Framework for enterprise wide video dial plan.SME on UCS – Virtualized offering lowers TCO and simplifies SME deploymentMobility with 3rd party phones – native clients for iPhone, Android, Blackberry, and Nokia can register natively with SME. SIP Early Offer – No media termination point required for interfacing with SIP Trunk. SIP Normalization and Transparency – Modify SIP Headers provide interoperability with multiple PBX/SIP TrunksScalability – 320 calls per second & 72,000 current callsImproved Administration – Increased simplicity/flexibility for load balancing and trunk configuration. SIP OPTIONS Ping – Optimize Call RoutingSIP Call Trace – Improved SIP Troubleshooting tools. Q.SIG over SIP – Traditional Telephony features between PBXs
  • For those interested in copies of this year’s conference presentations, please visit our Event Landing Page at www.networkerssolutionsforum.com. Here you will find the presentations for download.Lastly, we are interested in your feedback. Please take the time to fill-out the Conference Evaluation Form. If you did not receive an Eval Form with today’s Conference Guide, please see one of our Registration Attendants. The Eval Forms will be used for the prize draws at the Cocktail Reception.
  • Transcript

    • 1. Welcome
    • 2. Outsourcing your TDM Gateways:
      SIP Trunking as a Service Provider Cloud Service
      Darryl Sladden, dsladden@cisco.com
      Marketing Manager, SRTG
      #CNSF2011
    • 3. 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?
    • 4. 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
    • 5. 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
    • 6. 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
    • 7.
    • 8. 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
    • 9. 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
    • 10. 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
    • 11. Migrating to the CLOUD
    • 12. 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
      • 13. A Cisco Unified Border Element (CUBE) provides b2b interconnectivity for secure rich-media services
      Enterprise Domain 1
      Enterprise Domain 2
      SP VoIP
    • 14. 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
    • 15. 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
      Apps
      SBC
      Apps
      PSTN gwy
      PBX
      PSTN gwy
      Video
    • 23. SIP Trunking: Three Simple Steps
      $
      Set up
      for Future
      by Extending to Collaborative Services
      Save
      by Efficiently Interconnecting networks
      Simplify
      by Streamlining Services Aggregation
    • 24. Overall Architecture for SIP Trunking
      • Enterprise SBC based on ISR G2 or ASR
      • 25. Device Reuse
      • 26. Device consolidation
      • 27. Optional Session Manager
      • 28. Centralization
      • 29. Application integration
      CUBE
      Unified SME
    • 30. Why Now ?
    • 31. Go above and beyond IPT, seize up to 53% savings with SIP, SME and IME
      Capture a 53% cost savings opportunity
    • 32. Estimate your own savings potential from SIP Trunking: Use the Model
    • 33. SIP Trunk: The Benefits
      Areas of cost savings:
      • Reduction in total number of PSTN circuits
      • 34. Reduction in amount and cost of hardware needed to terminate circuits
      • 35. Reduction in unused circuits (sites can share capacity)
      Flexible:
      • New routes/numbers/capacity can be provisioned quickly
      • 36. Calls can be sent to anywhere that IP network can reach
      Less Complex:
      • Fewer circuits needed at remote sites (no TDM and IP connection)
      • 37. Local, LD, WAN, POTS can all be over same link
      • 38. Potential reduction in number of carriers required
      • 39. Less conversion needed (Remove’s IP -> TDM conversion at customer site)
    • SIP Trunking: Deployment Architectures and Issues
    • 40. 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
    • 41. 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
      • 42. 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
    • 43. Distributed Deployment Model
      100 ms from Branch toSP
      ------
      -------------------------------------
      Total Delay for Speech 100ms
      100 ms from Branch to SP
      • CUBE at each regional location
      • 44. Each site ports Phone numbers to IP address at that SITE
      All Calls Routed via a Local SIP Trunks
    • 45. A
      A
      SIP SP
      Centralized SIP Trunks: Trade-Offs
      Distributed PSTN Trunks
      Centralized SIP Trunk
    • 46. Centralized vs. DistributedSIP Trunks designs
      Distributed SIP Trunks
      • Each site has its own SIP Trunk and CUBE for PSTN access
      • 47. SIP trunk remains active during SRST
      • 48. RTP path is optimized
      • 49. Remote site CUBE also acts as local MTP and SRST router
      • 50. E911 locations tied to local site and hence more accurate
      • 51. SP and customer need to provision dial plan correctly to ensure optimal call routing
      • 52. Cost may not decrease as dramatically as centralized solution
      Centralized SIP Trunk
      • Consolidated PSTN SIP trunks at HQ site
      • 53. Remote sites have SRST for phone backup, but need TDM access for PSTN backup (1 FXO)
      • 54. Lower cost of equipment needed for termination
      • 55. Bandwidth requirement increases as “PSTN” calls from remote site now traverse WAN
      • 56. QoS concerns as RTP for remote site PSTN calls traverse WAN twice
      • 57. Requires porting of all DIDs to aggregated SIP trunks – geographic and SP challenges
      • 58. 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
    • 59. 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
    • 60. 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
    • 61. 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
    • 62. 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
    • 63. 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
    • 64. 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
    • 65. 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
    • 66. Evaluating SPs and Migrating
      Tips and Recommendations
    • 67. Evaluating SIP Trunking for PSTN ServicesService Offerings
    • 68. 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
      • 69. 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.
    • 70. SIP Trunk providers in Canada(+ others)
    • 71. 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
      • 72. Use the Cost Estimate
      Understand what your WAN costs are
      • Upgrading your IP to “gold” service WAN with your layer 2 providers
      • 73. 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
      • 74. 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
    • 75. 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
    • 76. CiscoUnifiedBorderElement
      Platform and Features
    • 77. 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
    • 78. CUBE
      Cisco Unified Border Element—More Than an SBCAn Integrated Network Infrastructure Service
      TDM Gateway
      • Voice and Video TDM Interconnect
      • 79. PSTN Backup
      Cisco Unified Border Element
      • Address Hiding
      • 80. H.323 and SIP interworking
      • 81. DTMF interworking
      • 82. SIP security
      • 83. 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
    • 84. 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
    • 85. 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
      • 86. Mid Call Codec Renegotiation
      • 87. Dial Peer Level Bind
      • 88. 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
      • 89. SIP Video Calls
      • 90. Scale to 16,000 Calls
      • 91. 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
    • 92. Roadmap
    • 93. SIP Trunking Cloud Service Roadmap
      New Billing Options
      • “Friends and Family” plans between customers
      • 94. Flat rate calling throughout Canada
      New Regions added until all are covered
      • Porting number from all areas to single IP address
      • 95. 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
      • 96. SP will offer support for Enterprise SBC redundancy
      New services on top of SIP Trunking
      • Managed Enterprise SBC service
      • 97. Outsourced call recording
      • 98. Wideband Codec on calls between customers
      • 99. Video Calls
      • 100. Call routing of calls to URLs
      Customizable by each Service Provider
    • 101. Cisco IP Trunking Evolution
      Today
      IT Cost Optimization
      Advanced User Experience
    • 102. 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
      • 103. 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
    • 104. 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)
      • 105. 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.
    • 106. 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
    • 107. 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
      • 108. Contact your Cisco account team and work on a trial of SIP Trunking
      • 109. Read the Whitepaperswww.cisco.com/go/cube
      • 110. Complete a detailed inventory of TDM Trunking
      • 111. Complete a cost model for transitioning from TDM to IP Trunking
      Thank you!
    • 112. Q & A
      #CNSF2011
    • 113. Cisco.com SIP Trunk Design Documents
    • 114. For conference presentations visit:
      www.networkerssolutionsforum.com
      Please take a moment to complete the
      Networkers Conference Event Evaluation Form
      #CNSF2011
    • 115. #CNSF2011