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

  1. 1. Welcome<br />
  2. 2. Outsourcing your TDM Gateways: <br />SIP Trunking as a Service Provider Cloud Service<br />Darryl Sladden,<br />Marketing Manager, SRTG<br />#CNSF2011<br />
  3. 3. ABSTRACT<br />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.  <br />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. <br />Cloud?<br />
  4. 4. Cloud<br /> is often touted as “the next best thing since sliced bread”<br />Game Changer<br />Tremendous Cost Savings<br />“Cloud Computing will cause a radical shift in IT” – CIO Survey<br />
  5. 5. Cloud is a new computing paradigm. In Cloud, IT resources and services are <br />abstracted from the underlying infrastructure and provided on-demand and at <br />scale in a multi-tenant environment. Cloud has several characteristics:<br />• Information technology, from infrastructure to applications, is delivered and <br />consumed as a service over the network<br />• Services operate consistently, regardless of the underlying systems<br />• Capacity and performance scale to meet demandand are invoiced by use<br />• Services are shared across multiple organizations, allowing the same underlying systems and applications to meet the demands of a variety of interests, <br />simultaneously and securely <br />• Applications, services, and data can be accessed through a wide range of connected devices(e.g., smart phones, laptops, and other mobile internet devices)<br /><br />
  6. 6. Is the CLOUD Ready for Essential Services ?<br />Amazon’s Trouble Raises Cloud Computing Doubt<br /><br />Does Cloud Computing Mean More Risks to Privacy<br /><br />Cloud Computing Is for the Birds<br /><br />
  7. 7.
  8. 8. What makes a good CLOUD service ? <br />Better value to end customers vs an in house solution <br /><ul><li>Lower price of SIP Trunks and SBCs then TDM gateways and TDM Trunks</li></ul>Improved redundancy vs an in house solution<br /><ul><li>Higher call completion rate and higher uptime then TDM connections</li></ul>Easy to debug/support technical issues<br /><ul><li>Service must result in less headaches in LONG run then in house solutions </li></ul>SP can make money on offering service<br /><ul><li>Essential to ensure investment level required to maintain quality</li></ul>#CNSF2011<br />
  9. 9. Translating this into SIP Trunking<br />BUYING TDM Trunking Gateways<br />Customizable VoIP protocol<br />Purchase TDM Gateway hardware upfront<br />Single Purpose equipment<br />Mature Technology<br />OUTSOURING your TDM Gateways, and Buying SIP Trunking Service<br />Standard service based on SIP<br />Purchase Enterprise SBC upfront<br />Multipurpose equipment<br />Cutting edge technology compared with TDM interconnect<br />
  10. 10. Is outsourcing TDM Gateway by moving to SIP Trunking a good choice for a CLOUD Service ?<br />YES<br />Service Providers (SP) have a great deal of experience with TDM to IP Gateways<br />The service can be MORE reliable then traditional TDM Gateways on premise<br />The service is technologically more efficient (ie fewer IP translations) , which means quality improves<br />SP can offer additional service on top base service to increase their value add<br />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<br />The function is not core to a high quality Enterprise UC deployment<br />
  11. 11. Migrating to the CLOUD<br />
  12. 12. IP<br /> IP<br />IP<br />IP<br />CUBE<br />CUBE<br />A<br />A<br />A<br />A<br />Rich-media<br />Enabled<br />Rich-media<br />Enabled<br />Rich-media<br />Enabled<br />SBC<br />SBC<br />SIP Trunking: Eventual solution to allow end to end IP CommunicationsEnabling Business-to-Business Collaboration<br />Enterprise Domain 1<br />Enterprise Domain 2<br />Narrowband voice to <br />Rich-media Interconnect<br />Changing Landscapes – VoIP Islands to VoIP Interconnects<br />Unified communications SIP Trunks to destinations beyond the Enterprise <br /><ul><li>Extend rich-media collaboration to vendors, partners and customers
  13. 13. A Cisco Unified Border Element (CUBE) provides b2b interconnectivity for secure rich-media services</li></ul>Enterprise Domain 1<br />Enterprise Domain 2<br />SP VoIP<br />
  14. 14. SP SIP<br />SP SIP<br />CVP<br />CVP<br />CVP<br />A<br />A<br />A<br />A<br />A<br />A<br />The Migrations to SIP Trunking in the CLOUD<br />1. TDM Trunking – Yesterday<br />2. TDM and IP Trunking – Today<br />Campus<br />Contact Center<br />Branch Offices<br />3. IP Trunking – Tomorrow<br />Campus<br />Contact Center<br />Branch Offices<br />Campus<br />Contact Center<br />Branch Offices<br />
  15. 15. Today’s Communications Network Challenges<br />Disparate PBXs<br />Integration with Applications<br />PSTN Tolls<br />PBX<br />Enterprise apps<br />Apps<br />Apps<br />SBC<br />PBX<br />PBX<br />Mobility<br />PBX<br />SBC<br />PSTN gwy<br />PBX<br />Social networking<br />Apps<br /><ul><li>Redundant
  16. 16. Complex
  17. 17. Inefficient
  18. 18. Expensive
  19. 19. Limited Features
  20. 20. Expensive
  21. 21. Inflexible
  22. 22. Server Intensive</li></ul>Apps<br />SBC<br />Apps<br />PSTN gwy<br />PBX<br />PSTN gwy<br />Video<br />
  23. 23. SIP Trunking: Three Simple Steps<br />$<br />Set up<br />for Future<br />by Extending to Collaborative Services<br />Save<br />by Efficiently Interconnecting networks<br />Simplify<br />by Streamlining Services Aggregation<br />
  24. 24. Overall Architecture for SIP Trunking<br /><ul><li>Enterprise SBC based on ISR G2 or ASR
  25. 25. Device Reuse
  26. 26. Device consolidation
  27. 27. Optional Session Manager
  28. 28. Centralization
  29. 29. Application integration</li></ul>CUBE<br />Unified SME<br />
  30. 30. Why Now ? <br />
  31. 31. Go above and beyond IPT, seize up to 53% savings with SIP, SME and IME<br />Capture a 53% cost savings opportunity<br />
  32. 32. Estimate your own savings potential from SIP Trunking: Use the Model<br />
  33. 33. SIP Trunk: The Benefits<br />Areas of cost savings: <br /><ul><li>Reduction in total number of PSTN circuits
  34. 34. Reduction in amount and cost of hardware needed to terminate circuits
  35. 35. Reduction in unused circuits (sites can share capacity)</li></ul>Flexible: <br /><ul><li>New routes/numbers/capacity can be provisioned quickly
  36. 36. Calls can be sent to anywhere that IP network can reach</li></ul>Less Complex:<br /><ul><li>Fewer circuits needed at remote sites (no TDM and IP connection)
  37. 37. Local, LD, WAN, POTS can all be over same link
  38. 38. Potential reduction in number of carriers required
  39. 39. Less conversion needed (Remove’s IP -> TDM conversion at customer site)</li></li></ul><li>SIP Trunking: Deployment Architectures and Issues<br />
  40. 40. SP VoIP<br />SP VoIP<br />PSTN<br />PSTN<br />Centralized<br />Distributed<br />CUBE<br />CUBE<br />CUBE<br />CUBE<br />MPLS<br />MPLS<br />A<br />A<br />A<br />A<br />SP VoIP<br />PSTN<br />CUBE<br />CUBE<br />CUBE<br />CUBE<br />CUBE<br />Site-SP RTP<br />Site-to-Site RTP<br />MPLS<br />Centralized and Distributed SIP Trunk Models<br />Hybrid<br />
  41. 41. Centralized Deployment Model<br />100 ms from HQ to SP<br />100ms from HQ to Branch<br />-------------------------------------<br />Total Delay for Speech 200ms<br />100 ms from HQ to SP<br />100 ms from Branch to HQ<br /><ul><li>CUBE at Headquarters Location
  42. 42. Each site ports Phone numbers to IP address at HQ (Phone numbers often ports out of region)</li></ul>All Calls Routed via a Centralized SIP Trunk<br />
  43. 43. Distributed Deployment Model<br />100 ms from Branch toSP<br />------<br />-------------------------------------<br />Total Delay for Speech 100ms<br />100 ms from Branch to SP<br /><ul><li>CUBE at each regional location
  44. 44. Each site ports Phone numbers to IP address at that SITE</li></ul>All Calls Routed via a Local SIP Trunks<br />
  45. 45. A<br />A<br />SIP SP<br />Centralized SIP Trunks: Trade-Offs<br />Distributed PSTN Trunks<br />Centralized SIP Trunk<br />
  46. 46. Centralized vs. DistributedSIP Trunks designs<br />Distributed SIP Trunks<br /><ul><li>Each site has its own SIP Trunk and CUBE for PSTN access
  47. 47. SIP trunk remains active during SRST
  48. 48. RTP path is optimized
  49. 49. Remote site CUBE also acts as local MTP and SRST router
  50. 50. E911 locations tied to local site and hence more accurate
  51. 51. SP and customer need to provision dial plan correctly to ensure optimal call routing
  52. 52. Cost may not decrease as dramatically as centralized solution</li></ul>Centralized SIP Trunk<br /><ul><li>Consolidated PSTN SIP trunks at HQ site
  53. 53. Remote sites have SRST for phone backup, but need TDM access for PSTN backup (1 FXO)
  54. 54. Lower cost of equipment needed for termination
  55. 55. Bandwidth requirement increases as “PSTN” calls from remote site now traverse WAN
  56. 56. QoS concerns as RTP for remote site PSTN calls traverse WAN twice
  57. 57. Requires porting of all DIDs to aggregated SIP trunks – geographic and SP challenges
  58. 58. E911 locations tied to HQ as opposed to phone location </li></li></ul><li>SIP Trunking Deployment Scenarios<br />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<br />
  59. 59. Top 5 Issues when adopting a SIP Trunks for PSTN Access Service<br />Interoperability with IP PBX<br />Fax Calls<br />Supplementary Features<br />Voice Band Data<br />Quality Control<br />
  60. 60. 1. Interoperability Issues with SIP Trunks<br />There is currently no standard for SIP Trunks that can provide the same level of consistency and interoperability of PSTN ISDN Trunks<br />There are efforts underway in the industry to have more interoperability; various efforts are being lead by the SIP forum, ATIS, TISPAN<br />The problem of interoperability is reduced by having a customer owned border element (CUBE) that can provide signaling interworking/normalization and transcoding<br />This problem can be further reduced by having a Service Provider owned Border Element that acts as a demarcation point for signaling<br />Customer should test before deployment of their first SIP Trunks solution, and replicate successful deployment procedure to ensure scaling<br /><br />
  61. 61. CUCM/CME and CUBE SIP Trunk Interop Test Plan Outline<br />Circuit Acceptance Test Cases<br />SP Layer 2 Connection<br />SP Layer 3 Connection<br />SP Reachability and Routing<br />Connectivity Test Cases<br />Registration sequence<br />Session Refresh<br />Basic outbound/inbound call completion<br />Quality of Service<br />Call Admission Control<br />Management Access<br />Call Accounting<br />Voice Quality<br />FAX Quality<br />Non-Standard Calls<br />Stability and Duration<br />Restart<br />SIP Application (Call Flow) Test Cases<br />Caller ID<br />Codec Negotiation<br />Call Hold/Resume<br />Call Forward (Call Forward All to user on PSTN behind SIP Trunk)<br />Call Transfer<br />Ad-Hoc Conference<br />IVR Interaction (Both local and remote IVR)<br />DTMF<br />FAX, Mode, TTY<br />Emergency/911<br />Call types (Local, Long Distance, International)<br />Failover Test Cases<br />Layer 1, 2, 3, 4 failover scenarios<br />Pg. 243<br />
  62. 62. 2. Fax Calls<br />SIP Trunks can typically use three different methods to supports FAX calls<br />All calls are sent as G711<br />Call sends a RE-INVITE to up-speed to G711 when a FAX tone is detected<br />T.38 FAX capabilities are exchanged and fax relay is used<br />SIP Service provides also occasionally offer a separate fax to -mail service using T.37 Store and Forward fax<br />Recommend that your SP support T.38<br />
  63. 63. 3. Supplementary Services<br />Typical Supplementary Services<br />Placing call on HOLD<br />Forward on Busy/No Answer to Number within premise<br />Transferring call to another extension<br />Correct billing for forwarded calls<br />Testing of Supplementary Services before deployment is only way to ensure success<br />Create a test case for each service before deployment<br />Report findings to Service Provider<br />Determine if lack of these functionality should effect deployment<br />The supplementary service invoked over the SIP Trunk is not supported or understood by the far end SIP switch<br />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<br />CUBE will resolve interopissues<br />SIP Signaling End-to-End Causes Interop Issues<br />All Signaling Is Translated Resulting in Fewer Interop Issues<br />SIPNetwork<br />PSTN<br />
  64. 64. 4. Voice Band Data<br />Sending a Modem Call Over a Codec Is Like Putting It Through a Cheese Grater: the Signal Will Never Be the Same<br />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<br />Voice Band Data can work reliably up to 56K with TDM Trunks<br />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)<br />With compressed codecs (i.e. G729), you cannot reliable send modem tones over VoIP calls (G711 required)<br />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:<br />Baudot connections for deaf users<br />Credit card validation systems<br />Security systems<br />Pitney Bowes Postage Machines<br />These systems should all be tested before a SIP Trunk for PSTN access is used as a replacement at <br />
  65. 65. 5. Quality Control<br />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)<br />A “best practice” is to create a method of flagging calls that are very bad (either via CDRs/CMRs analysis or user feedback) <br />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<br />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<br />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<br />
  66. 66. Evaluating SPs and Migrating<br />Tips and Recommendations<br />
  67. 67. Evaluating SIP Trunking for PSTN ServicesService Offerings<br />
  68. 68. SP IP Network<br />SBC<br />SIP Trunking also can be used for TDM PBXs<br />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 <br />TDM PBX<br />SIP<br />TDM<br />CUBE<br />SIP Trunking for TDM PBXs saves money without transitioning TDM PBX to IP PBX<br /><ul><li>Required a Voice Gateway to translate TDM to SIP
  69. 69. Voice Gateways such as (ISR G2) can support both TDM and SIP Trunks with the same equipment</li></ul>Make sure your SP offers SIP Trunking for TDM PBXs via Cisco Voice Gateways.<br />
  70. 70. SIP Trunk providers in Canada(+ others)<br />
  71. 71. Migration PlanHow to Successful adopt a Cloud Service<br />Read and review White papers on Communications Transformation<br /><ul><li></li></ul>Find out who offers Services in your region<br /><ul><li>Ask current provider or VARs, look at who provides Layer 2 connectivity</li></ul>Understand what your Telecom PSTN costs are<br /><ul><li>Both costs of connections (T1/E1/Analog) and per minute costs
  72. 72. Use the Cost Estimate</li></ul>Understand what your WAN costs are<br /><ul><li>Upgrading your IP to “gold” service WAN with your layer 2 providers
  73. 73. IP costs for SIP Trunk are not FREE (as is shown in many ROI calculators), for toll quality voice</li></ul>Deploy trial with some services<br /><ul><li>Outbound is easy as it does not require porting of phone numbers
  74. 74. Inbound does require porting of phone numbers to IP addresses and this may not be as easy as SP promise</li></ul>Monitor quality of deployment and use experience to determine where you want to be in five years—change the world<br />
  75. 75. SBC<br />A<br />IP Network<br />SIP Trunks Move from TDM to IP connection for interconnect between SP and Enterprise for Voice traffic<br />Enterprise UCM Deployment<br />voice<br />Class 4/5<br />Switch<br />TDM-based PSTN<br />Initial Deployments have TDM Gateway to Class 4/5 Switch<br />Enterprise SBC is added and connection to SP SIP Trunk is initiated<br />Phone numbers are ported from TDM trunk to IP Trunks<br />data<br />CUBE<br />SP SBC<br />TDM Trunk Call Path<br />IP Trunk Call Path<br />
  76. 76. CiscoUnifiedBorderElement<br />Platform and Features<br />
  77. 77. ASR 1004/6 RP2<br />ASR 1001<br />3900E ISR G2<br />ASR 1002<br />3900 ISR G2<br />AS5000XM<br />800/1861E<br />Cisco Unified Border Element (Enterprise Edition) Portfolio<br />50-150+<br />50-100<br />20-35<br />New Platform<br />17<br />CPS<br />2900 ISR G2<br />8-12<br /><5<br />Even Higher Capacity<br />New Platform<br />10-12K<br />12-16K+<br />5-30<br />200-600<br />600-800<br />900-1000<br />1500-1700<br />2000-2500<br />Active Voice Call (Session) Capacity<br />
  78. 78. CUBE<br />Cisco Unified Border Element—More Than an SBCAn Integrated Network Infrastructure Service<br />TDM Gateway<br /><ul><li>Voice and Video TDM Interconnect
  79. 79. PSTN Backup</li></ul>Cisco Unified Border Element<br /><ul><li>Address Hiding
  80. 80. H.323 and SIP interworking
  81. 81. DTMF interworking
  82. 82. SIP security
  83. 83. Transcoding</li></ul>Routing, FW, IPS, QoS<br />Note: An SBC appliance would have only these features<br />Unified CM Conferencing and Transcoding<br />WAN Interfaces<br />SRST<br />RSVP Agent<br />VXML<br />GK<br />Note: Some features/components may require additional licensing<br />
  84. 84. Mine<br />Yours<br />Cisco Unified Border Element Key Features<br />Session Mgmt<br />Demarcation<br />Fault isolation<br />Topology Hiding<br />Network Borders<br />L5/L7 Protocol Demarc<br />Statistics and Billing<br />Real-time session Mgmt<br />Call Admissions Control<br />Ensuring QoS<br />PSTN GW Fallback<br />Statistics and Billing<br />Redundancy/Scalability<br />Security<br />Interworking<br />Encryption<br />Authentication<br />Registration<br />SIP Protection<br />FW Placement<br />Toll fraud<br />H.323 and SIP<br />SIP Normalization<br />DTMF Interworking<br />Transcoding<br />Codec Filtering<br />Fax/Modem Support<br />H.323 and SIP<br />SIP Normalization<br />DTMF Interworking<br />Transcoding<br />Codec Filtering<br />Fax/Modem Support<br />
  85. 85. Cisco Unified Border Element (CUBE) Features delivered in 2010<br />Continue rich feature development on SIP Interworking and Media Optimizing<br />CUBE 8.5 Enhancements<br /><ul><li>Call Preservation with Box to Box Redundancy
  86. 86. Mid Call Codec Renegotiation
  87. 87. Dial Peer Level Bind
  88. 88. RAI in SIP Messages</li></ul>CUBE 8.6 Enhancements<br />Registration Proxy support<br />Full support for UPDATE method<br />Conditional SIP Profiles<br />CUBE(Ent) on ASR (RLS 3.2)<br /><ul><li>H323 to SIP Voice Calls
  89. 89. SIP Video Calls
  90. 90. Scale to 16,000 Calls
  91. 91. Full Stateful failover with Box to Box Redundancy</li></li></ul><li>Cisco Unified Border Element (CUBE)CUBE 8.8 on ISR G2 and CUBE on ASR (RLS 3.3)<br />Enables rich applications. Affordable for the small branch. Enhanced interoperability. <br />New Capabilities <br />Media forking for call recording on ISR G2<br />CUBE functionality extended to 88x/892 platforms<br />Improved interoperability including; sRTP-RTP supplementary services; Support for Multi-cast music on hold; Domain based routing; and dynamic REFER handling<br />ASR IPv6 improvements: RTCP Pass through and T.38<br />Customer Benefits<br />Enables a simplified, lower cost architecture for call recording<br />Makes SIP trunking more cost effective for the small branch/ business<br />Improved interworking with SIP trunk service providers and endpoints<br />Partner Benefits<br />Expands the partner business opportunities into recording <br />Creates the ability to position CUBE into small deployments<br />
  92. 92. Roadmap<br />
  93. 93. SIP Trunking Cloud Service Roadmap<br />New Billing Options<br /><ul><li>“Friends and Family” plans between customers
  94. 94. Flat rate calling throughout Canada</li></ul>New Regions added until all are covered<br /><ul><li>Porting number from all areas to single IP address
  95. 95. SP will start to offer service across multiple countries </li></ul>New redundancy options<br /><ul><li>SP offer the ability to send calls to multiple devices that can be changed in real time
  96. 96. SP will offer support for Enterprise SBC redundancy</li></ul>New services on top of SIP Trunking<br /><ul><li>Managed Enterprise SBC service
  97. 97. Outsourced call recording
  98. 98. Wideband Codec on calls between customers
  99. 99. Video Calls
  100. 100. Call routing of calls to URLs</li></ul>Customizable by each Service Provider<br />
  101. 101. Cisco IP Trunking Evolution<br />Today<br />IT Cost Optimization<br />Advanced User Experience<br />
  102. 102. Cisco or Non-Cisco<br />Contact Center<br />SIP or TDM Trunk<br />CUBE<br />SBC based Noise Reduction with CUBE<br />1<br />Media is processed to improve quality<br />2<br />Caller in Noise environment<br />DSP<br />SIP<br />RTP<br />CUBE Noise Reduction<br />3<br />Called Party hears voice of caller with background noise removed<br /><ul><li>Enhance Feature statically configured based on phone numbers
  103. 103. Parameters can be dynamically changed to support different environments</li></li></ul><li>SP IP Network<br />CUBE<br />A<br />SBC<br />DSPs will change the way SBC are deployed and used – Be ready for these advancements<br />Enterprise SBC, such a CUBE will add more capabilities to improve Voice and Video communications<br />SIP/H.323<br />SIP<br />DSP<br />DSPs ENHANCE AUDIO and VIDEO<br />Transcoding<br /> Input Gain <br /> Noise Cancellation <br /> Acoustic Shock<br />Media Forking / Recording<br />Synthetic Traffic Generation<br /> Video Mixing<br /> Acoustic Echo Cancellation<br /> Text Overlay<br /> Audio Transcribing<br /> Video improvement/ enhancement<br />Shipping now or soon<br />
  104. 104. CUBE<br />CUBE<br />CUBE Media Forking<br />Destination – Can be any SIP device or Trunk<br />Enterprise -B<br />Enterprise -A<br />SIP<br />SIP<br />SIP<br />B<br />A<br />WAN<br />Source CUBE<br />RTP<br />RTP<br />RTP<br />C<br /><ul><li>Media Forking results in 2 INVITES and RTP packets from (A) to (B) and (C)
  105. 105. INCOMING INVITE (A)</li></ul>INVITE sip:11111@ SIP/2.0<br />Via: SIP/2.0/UDP;branch=z9hG4bK-23006-1-0<br />From: sipp <sip:123@>;tag=23006SIPpTag001<br />To: sut <sip:11111@><br />Call-ID: 1-23006@<br />CSeq: 1 INVITE<br />Contact: sip:123@<br />Max-Forwards: 70<br />Call-Info: <sip:>;purpose=X-cisco-enableforking<br />Subject: Performance Test<br />Content-Type: application/sdp<br />Content-Length: 172<br />v=0<br />o=user1 53655765 2353687637 IN IP4<br />s=SIP Call<br />c=IN IP4<br />t=0 0<br />m=audio 6768 RTP/AVP 8 19<br />a=rtpmap:8 PCMA/8000<br />a=rtpmap:19 CN/8000<br />a=ptime:20<br />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.<br />
  106. 106. CUBE Top of Mind for <br />2011-2012<br />Feature equivalence on ASR and ISR G2<br />Media Forking on ISR G2<br />Mid Call REINVITE consumption<br />Noise Cancellation <br />Support for MMOH on SIP Trunks<br />SME+CUBE Management and Operation<br />Acoustic Shock Prevention<br />CUBE on 800 Series<br />Advanced SRTP to RTP interworking<br />
  107. 107. Call to Action<br />Available at<br />Headline<br />Headline<br />Headline<br />Know the $$$ impact<br />Run a trial<br /><ul><li>Learn how to configure SIP
  108. 108. Contact your Cisco account team and work on a trial of SIP Trunking
  109. 109. Read the
  110. 110. Complete a detailed inventory of TDM Trunking
  111. 111. Complete a cost model for transitioning from TDM to IP Trunking</li></ul>Thank you!<br />
  112. 112. Q & A<br />#CNSF2011<br />
  113. 113. SIP Trunk Design Documents<br />
  114. 114. For conference presentations visit: <br /><br />Please take a moment to complete the <br />Networkers Conference Event Evaluation Form<br />#CNSF2011<br />
  115. 115. #CNSF2011<br />