SIP Trunks for PSTN Access

Transcription

1 SIP Trunks for PSTN Access BRKUCT-2001 Tony Mulchrone 1

2 Agenda Technology Overview Deployment Scenarios Issues with SIP Trunks for PSTN Access What's in a SIP Trunk? What services should customers look for in SIP Trunks? What services should Service Providers offer with SIP Trunks? Using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between Service Provider and Customer Issues and configuration options for Cisco Unified Communications Manager Express (CUCME) with SIP Trunks for PSTN Access Issues and configuration options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access General Troubleshooting Issues with SIP Trunks Resources 2

3 SIP Trunks for PSTN Access SIP Trunks can be defined as a method to interconnect different SIPbased networks. SIP Trunks can be used by a wide range of organizations as a method of interconnections. One large application for SIP Trunks is for service providers to offer PSTN interconnects for voice-enabled SIP applications over an IP connection. This gives service providers the ability to offer PSTN services over a combined IP infrastructure, reducing the cost and complexity of the network and providing a single point of interconnect to their users. It also allows them to offer services outside of their geographic regions that have a PSTN footprint and consolidate the interconnect to the PSTN across multiple customers. The benefit for enterprises and smaller businesses is that they can get PSTN interconnect services without needing a PSTN facility at their location. This allows the removal of the PSTN hardware from their equipment and allows connections to the PSTN over a wide variety of mediums supporting IP such as wireless and cable. This session discusses implementation issues and requirements when using SIP Trunks for PSTN access. 3

4 Objectives and Scope of This Seminar Enterprise Owned Cisco Unified Communications Manager or Cisco Unified Communications Manager Express Cisco Unified Border Element H.323/SIP Trunk H.323/SIP Trunk Cisco Unified Border Element Service Provider Owned SIP Trunk Service Provider Enterprise Located Equipment Discuss issues to consider when adding SIP Trunks to solutions Focus will be on deployments with Cisco Unified Communications Manager and Cisco Unified Communications Manager Express Specific Service Provider offering will NOT be evaluated Solution will focus on recommended design solution of using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between SP offering and customer. A Cisco UBE could be provided by the Service Provider and one should be owned by the customer 4

5 Technology Overview 5

6 SIP Trunks Why? SIP Trunks can be cheaper (sometimes) SIP Trunks can be more versatile (i.e. deployed over different physical layer links) SIP Trunks can offer equipment consolidation SIP Trunks can be used for many different purposes Between applications (i.e. Conference bridges to IP PBX, such as MeetingPlace to Cisco Unified Communications Manager) For PSTN Access (Centralized or Distributed) Between different IP Communication Zones within a company (i.e. Europe and USA) or between Companies (i.e. Disney and Apple) NB: This Seminar will focus exclusively on PSTN Access 6

7 SIP Trunks Myth vs. Reality Myth SIP Trunks can be deployed over any media SIP Trunks are always cheaper then PSTN trunks for PSTN Access SIP Trunks provide the exact same experience for the end users SIP Trunks are easy to deploy and just work SIP Trunks should always be used Reality SIP Trunks should only be deployed over media that can provided a guarantee QoS that is acceptable (i.e. it would not be recommended to deploy them across Satellite links if Voice quality is important) Large Enterprise have such low rates for traditional TDM based telephony, rates over SIP Trunks may not save that much in per minute charges for Local or Long Distance voice calls SIP Trunks can provide the same experience In many cases, but some cases (i.e. Baudot connections for Deaf users or V.92 speed modem connections) experience is different SIP is easy to deploy, but interconnection between different vendors implementations of SIP and different Service Providers offering is not yet ironed out In some cases H323 trunks make a better choice and in some cases TDM trunks make a better choice 7

8 SIP Trunks Good Reasons for Implementing Them SIP Trunks offer a roadmap to Enhanced Services WideBand Codecs Calls with SUBJECT lines Exchange of Calendaring information during a call Multimodal communications: Voice, Video, Chat, file sharing, over the same communications pipe SIP Trunks offer the ability to have a voice call over disparate physical links SIP Trunk can be implemented over a wide variety of IP communications trunks (i.e. Metro Ethernet, WiFi, GSM Cellular) SIP Trunks offer the ability to have improved redundancy for communications IP Links can be built with redundancy of communications methods and fast failover that results in quicker time to repair in case of failure 8

9 Agenda Technology Overview Deployment Scenarios Issues with SIP Trunks for PSTN Access What's in a SIP Trunk? What services should customers look for in SIP Trunks? What services should Service Providers offer with SIP Trunks? Using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between Service Provider and Customer Issues and configuration options for Cisco Unified Communications Manager Express (CUCME) with SIP Trunks for PSTN Access Issus and configuration options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access General Troubleshooting Issues with SIP Trunks Resources 9

10 Deployment Scenarios 10

11 Deployment Scenarios Basically, there are three methods of deploying SIP Trunks today, Centralized where trunks for all regions are centralized and provided only from a central location, distributed, where each regional office has SIP Trunk from the providers and Hybrid models where different solutions are provided for different types of traffic Centralized Location of All SIP Trunks All PSTN Trunks are removed from remote sites and replaced with SIP Trunks that terminate only at the central datacenter or headquarters. This HQ site receives and routes ALL PSTN traffic via SIP Trunks to Service Provider. Distributed Trunks to All Locations SIP Trunks are provided from the Service Provider to all sites. Each site removes their PSTN access and instead replaces it with SIP Trunks from the provider that terminate at the remote sites. Provider needs to route phone calls to remote site via SIP trunk at remote sites. Hybrid Trunk Deployment, Deploy Trunks Based on Function SIP Trunks are added in Headquarters and /or remote sites to complement PSTN trunks. Dialplan is altered so that traffic can flow across most effective trunk, and traffic can be effectively routed via both HQ and remote site Trunks. 11

12 Centralized Deployment Model All Calls routed via a centralized SIP Trunk 12

13 Distributed Deployment Model All Calls routed via a local SIP Trunks 13

14 Hybrid Deployment Model Calls routed via Centralized SIP Trunk Calls Routed via PSTN Calls Routed via Remote site SIP Trunk 14

15 Agenda Technology Overview Deployment Scenarios Issues with SIP Trunks for PSTN Access What's in a SIP Trunk? What services should customers look for in SIP Trunks? What services should Service Providers offer with SIP Trunks? Using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between Service Provider and Customer Issues and configuration options for Cisco Unified Communications Manager Express (CUCME) with SIP Trunks for PSTN Access Issus and configuration options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access General Troubleshooting Issues with SIP Trunks Resources 15

16 Issues with SIP Trunks for PSTN Access Interoperability with IP PBX Voice Band Data Fax Call Supplementary Features Quality Control 16

17 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 that can provide signaling interworking and transcoding This problem can be further reduced by having a Service Provider owned Border Element on Customer premise that acts as a demarcation point for signaling Customer should test / test / test before deployment or first SIP Trunks solution, but replicate after that for scaling 17

18 Voice Band Data Voice Band Data (VBD) is the ability to send information such as slow speed modem calls for credit card transactions or alarm systems (i.e. Telematics) across the voice channel of an PSTN connection Voice Band Data can work reliable up to 56K with PSTN connections With any codecs you cannot maintain a PCM clock sync so 56K connections are not possible; but medium speed modem connections are possible over G711 With compressed codecs (i.e. G729), you cannot reliable send modem tones over VoIP calls, so only low speed connections 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 three most used types of VBD are: Baudot connections for deaf users Credit card validation systems Security systems Sending a Modem Call Over a Codec Is Like Putting It Through a Cheese Grater: the Signal Will Never Be the Same These systems should all be tested before a SIP Trunk for PSTN access is used as a replacement 18

19 Fax Calls SIP Trunks can typically use three different methods to supports FAX calls T.38 FAX capabilities are exchanged All Calls are sent as G711 and best effort fax is done Call sends a RE-INVITE to up-speed to G711 when a FAX tone is detected SIP Service provides also occasionally offer a separate fax to service using T.37 Store and Forward fax Fax Method T.38 Fax Capabilities Exchanged as Part of SIP Messages All Call Sent as G711 Fax Tone Is Detected and RE- INVITE to up-speed to G711 Is sent Pros Highest fax success rates can be achieved Cleanest solution from signaling and media point of view Use less bandwidth than G711 Most widely deployed Simplest solution Provides benefits of least bandwidth with G729 call initially upspeeding to G711 if call is FAX Tone (2100Hz) can be mixed between Modem and Fax Fax Pass-Through Fax and Voice calls differentiated Cons Degree of interoperability Not offered by many Service Providers Consumes a large amount of bandwidth for all calls No ability to distinguish FAX calls from Voice calls in CDRs Each vendors support of RE- INVITEs is different Currently not supported with all Cisco equipment 19

20 Supplementary Services 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 Testing of Supplementary Services before deployment is only way to ensure success Create a testcase for each service before deployment Report findings to Service Provider Determine if lack of these functionality should effect deployment Typical Supplementary Services test cases Placing call on HOLD Forward on Busy/No Answer to Number within premise Transferring call to another extension Correct billing for forwarded calls All Signaling Is Translated Resulting in Fewer Interop Issues SIP Signaling End-to-End Causes Interop Issues PSTN SIP Network 20

21 Quality Control As customers have deployed SIP Trunks for PSTN access, the experience for users has been inconsistent (i.e. one calls is great, next is not great) A best practice is to create a method of flagging calls (either via CDRs analysis or user feedback) that are very bad Use data from CDRs (i.e. Jitter, Packet Loss) to determine if there are trends and average; these statistics can be gathered from the Customer premise Border Element Try to determine if quality issues co-relate 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 21

22 Agenda Technology Overview Deployment Scenarios Issues with SIP Trunks for PSTN Access What's in a SIP Trunk? What services should customers look for in SIP Trunks? What services should Service Providers offer with SIP Trunks? Using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between Service Provider and Customer Issues and configuration options for Cisco Unified Communications Manager Express (CUCME) with SIP Trunks for PSTN Access Issus and configuration options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access General Troubleshooting Issues with SIP Trunks Resources 22

23 SIP Trunk for PSTN Service Offerings Requirements Unacceptable Offering Good Offering Security None IP Address Validation of SIP INVITES Best Offering TLS Signed SIP INVITES Fax None G711 for all Calls T.38 support Voice Band Data None Offer to work each issue individually Offers SLA for Data speeds for VDB over SIP Trunks Uptime None SLA with 95% uptime offered SLA with % uptime offered and access from customer for reports with refunds for nonconformance Calling Plans N/A Per minute Flat rate with no cost calls between customers; each trunk can configure their own calling plan; billing records provided via WEB interface Redundancy None Ability to route calls to a different phone number or IP address when trunk is down Call re-routed in real time when the SIP Trunk fails; routing is to both secondary IP address and PSTN number Call Number Porting None Call porting of phone numbers can be accomplished for some area codes within 30 days All area codes can have phone number ported with zero lost calls in 48 hours 23

24 Cisco Unified Border Element (Formerly Cisco Multiservice IP-to- IP Gateway) as a Border Element Cisco Unified Border Element as a Border Element for SIP Trunk Solutions 24

25 Cisco Unified Border Element (formerly Cisco Multi-service IP to IP Gateway) Cisco UBE allows the network to provide these services Co-existence with other services such as MTP, SRST, TDM GW CUBE Demarcation Fault Isolation Call Accounting Topology Hiding H323 to SIP SIP to SIP SIP Profiles & Variants Cisco Unified Border Element Session Management Call Admission Control IP QOS/SLA IOS Firewall Integration RTP Media Validation Signaling Protection Inter-working Security 25

26 CUBE Features Cisco Unified Communications Manager Cluster Cisco Unified Border Element H.323/SIP Trunk SBC SIP Trunk Service Provider Network/Topology Hiding for Voice and Video Calls Protocol Support H.323 and SIP Voice Codecs G.711, G.729, G.726, G.723, G.728, Transparent Video Codecs H.261, H.263 and H.264 Codec Filtering Media Media Flow Through and Media Flow Around DTMF Interworking H.245 Alphanumeric, Signal, RFC2833, SIP NOTIFY Fax/Modem T.38, Passthrough, Cisco Fax Relay, Modem Passthrough Security TLS, IPSec with SRTP Signaling Interworking Supplementary Services Transcoding Transport Mode TCP, UDP Number Translation Quality of Service Call Admission Control Call Detail Records TCL/VXML Support Rotary Support 26

27 Why Use Cisco UBE for Interconnects? Feature Connect Method Over the Top Managed Router Managed Router Running NAT/PAT Managed Router Running Cisco UBE (IP2IP GW) Voice Calls Possible X X X X QoS Can Be Guaranteed X X X Security X X X IP Address Hiding X X Call Counting Signaling Interworking (H323/SIP) DTMF Interworking (Inband to OOB) Transcoding (Any to Any Codec, etc.) TCL/VxML (Ability to Run Scripts on Calls) Redundancy (HSRP) X X Simple Interconnect SP Configuration with Multiple Endpoints Per Call Voice Quality Statistics CDR Collection Point for Multiple Entities Support for REFER (Note: NOTIFY on DEMAND, Not Subscribed) X X Support for REFER with NSS to Pass Information X X X X X X X X X X 27

28 Media Flow-Through vs. Flow-Around Media Flow-Through Signal Leg: 1 Signal Leg: 2 Media Leg: 1 Media Leg: 2 SBC Media Flow-Around Signal Leg: 1 Signal Leg: 2 SBC Media Bypasses the CUBE Gateway 28

29 Address Hiding CUCM Cluster SBC Site A CUBE Gateway IP WAN /27 Third Party Application Server Site A x/24 Cisco UBE can hide the customer s IP addresses by presenting its own IP address to the public side Cisco UBE can also provide a solution for a customer s multisite deployment in which there are overlapping IP addresses Cisco UBE acts like a Back-to-Back User Agent it would reformulate a request with entirely new From, Via, Contact, Call-ID, etc. RTP headers are changed when configured for media flow-through 29

30 Protocol Interworking H.323-H.323 In Leg Out Leg Support Fast Start Fast Start Bi-Directional Slow Start Slow Start Bi-Directional Fast Start Slow Start Bi-Directional SIP-SIP In Leg Out Leg Support Early Offer Early Offer Bi-Directional 12.5(1 st )T Delayed Offer Delayed Offer Bi-Directional Delayed Offer Early Offer Uni-Directional H.323-SIP In Leg Out Leg Support Fast Start Early Offer Bi-Directional Slow Start Delayed Offer Bi-Directional 30

31 Delayed Offer (DO) Early Offer (EO) 12.5(1 st )T Provide a mechanism to translate a SIP DO to EO Uni-directional only (EO to DO is not required/needed) Designed to enable Service Provider interconnects Allows SP SIP Trunks (that often support only EO) to interconnect to CUCM 5.x/6.x SIP Trunks where DO is preffered CUCM 5.x/6.x supports EO for outbound audio calls only with a G711 MTP - This DO to EO feature removes the MTP requirement CUCM 5.x/6.x doesn t support EO for video calls Release: Audio 12.5(1 st )T, Video 12.5(2 nd )T 31

32 DTMF Interworking H.323 H.245-Alphanumeric H.245-Signal RFC2833 G711 In Band Voice SIP RFC2833 RFC2833 RFC2833 RFC2833 SIP RFC2833 RFC2833 SIP RFC2833 G711 In Band Voice 32

33 Media Transcoding 12.5(1 st )T ilbc, isac, Speex Internet Telephony SP VoIP Network CUBE CUBE Enterprise Network IP Phones: G.711, G.729, G.722 CUBE Transcoding: G.711, G.723.1, G.726, G.728, G.729/a, ilbc, G.722 CUBE supports universal transcoding any voice codec to any other codec Up to 400 sessions on 3845 Re-packetization with different sample sizes not supported Supported Codecs Release G.711 a-law 64Kbps 12.4(11)XW G.711 µlaw 64Kbps 12.4(11)XW G & 6.3 Kbps 12.4(11)XW G.729 (all variants) 8Kbps 12.4(11)XW ilbc 13.3 & 15.2 Kbps 12.4(11)XW G kbps 12.5(1 st )T Transcoding Not Supported on 7200 and

34 IOS Firewall Support SIP ALG 12.5(1 st )T SIP support added to ALG, allowing FW inspection of: Support for RFC3261 OPTIONS, INVITE, REGISTER, ACK, CANCEL, BYE Support for RFC3261 extension methods INFO (RFC2976), PRACK (RFC3262), SUBSCRIBE/NOTIFY (RFC3265), UPDATE (RFC 3311), REFER (RFC3515/RFC3892), MESSAGE (RFC3428) Protocol conformance check: Enforcement of mandatory and forbidden header fields SIP Dialog and transaction enforcement SIP over TCP Media negotiation (RFC3264) Early and Late Media (RFC3960) 34

35 IOS FW Support SIP Application Inspection and Control Filter out a black/whitelist of Callers or Callees Filter SIP messages based on SIP Methods, SIP header fields, or content-type! Match SIP methods match request method <method>! Match header fields match {request response} header <header-name> regex <regex-param-map>! Match SIP Status line match response status regex <regex-param-map> Validation of max-forwards Disable IM Example: Block SIP messages coming from a particular proxy parameter-map type regex unsecure_proxy pattern compromised.server.com class-map type inspect sip sip_class match request header Via regex unsecure_proxy policy-map type inspect sip sip_policy class type inspect sip sip_class reset log 12.5(1 st )T 35

36 IOS FW Support AIC Rate-Limiting 12.5(1 st )T Rate-limits application messages Protects against DOS attacks Uses the rich match criteria for AIC classes Example: Limit to 10 SIP INVITE messages per second class-map type inspect sip my_sip_class match request method invite policy-map type inspect sip my_sip_policy class type inspect sip my_sip_class rate-limit 10 Example: Limit to 16 SIP INVITE requests whose header length is greater than 1026 class-map type inspect sip match-all my_sip_class match request method invite match request header length gt 1026 policy-map type inspect sip my_sip_policy class type inspect sip my_sip_class rate-limit 16 36

37 Call Admission Control Call Admission Control Can Be Provided Based on: 1. RSVP (only if IP-to-IP Gateway is used on both ends) 2. Total calls 3. CPU 4. Memory 5. IP call capacity 6. Max-connections 37

38 SIP Trunks for Cisco Unified Communications Manager Express (CUCME) 38

39 Cisco Unified Communications Manager Express Supported on 3.4 on 12.4(4)T1 Supported on 4.0 on 12.4(9)T1 Supported on 4.1 on 12.4(15)T 39

40 Configuration Options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access 40

41 Cisco Unified Communications Manager Release SIP Trunks Gateways Cisco Communications Manager 5.0 CUCME CUBE Carriers/ Other PBXs Cisco Communications Manager 5.0 MeetingPlace/ MP Express Cisco Unified Presence Server Cisco Unity /Cisco Unity Connection CTI Apps Cisco and 3 rd -Party Phones Soft Phones Video Endpoints Cisco Unified Communicator Microsoft LCS IBM Sametime Cisco Communications Manager 5.0 integrates rich, native SIP and SIMPLE support on both line-side and trunk-side interfaces (for both audio and video calls) with integrated presence on phones and applications; KPML and RFC 2833 support for DTMF; TLS and Digest Authentication for security; seamless protocol inter-working between SIP, H.323, MGCP, SCCP, TAPI/JTAPI; RSVP support for topology-aware Call Admission Control, 2008 and Cisco Systems, much Inc. more... All rights reserved. Cisco Public SCCP MGCP H.323 CTI SIP/SIMPLE/KPML 41

42 Cisco Communications Manager Trunks- SIP Media Support SIP Call Flow with Cisco Communications Manager Cisco Communications Manager supports receiving Early Offer and Delayed Offer Cisco Communications Manager sends Delayed Offer to the callee unless MTP required is checked then Early Offer is used Support for Delayed Offer is mandatory in RFC 3261: Concretely, the above rules specify two exchanges for UAs compliant to this specification alone the offer is in the INVITE, and the answer in the 2xx (and possibly in a 1xx as well, with the same value), or the offer is in the 2xx, and the answer is in the ACK. All user agents that support INVITE must support these two exchanges. 42

43 Cisco Communications Manager SIP Trunks Using Early Offer SIP Early Offer Outbound Calls must use an MTP If no MTP resources available, call reverts to Delayed Offer Asymmetric EO DO is supported MTPs are available in three forms: Software based MTPs in Cisco IOS -based gateways (available with any Cisco IOS T-train software and scaling up to 500 sessions (calls) on the Cisco 3845 router platform) Hardware based MTPs in Cisco IOS-based gateways (available with any Cisco IOS T-train software release hardware MTPs use on board DSP resources and scale calls according to the number of DSPs supported on the Cisco router platform) Software based MTPs using the Cisco Communications Manager IP Voice Streaming application on an Cisco MCS server MTPs (and Transcoders) can be controlled by Cisco Communications Manager or the Cisco Unified Border Element 43

44 CUCM SIP Trunking SIP Delayed Offer, G711 & G729 Regions - No MTP XXX SIP Signaling 5001 SIP 7776 RTP Media Stream 5555 Voice Gateway RTP Media Stream CUBE SP SIP Switch 7777 SIP Delayed Offer G711 & G729 Regions between devices and SIP Trunk For CUCM outbound calls Codec Preference G711 For Inbound Calls to Cisco Communications Manager SIP Delayed Offer calls SIP switch selects codec SIP Early Offer calls CUCM selects codec G711 Fax calls Fax Pass-through and T.38 Fax Relay Inbound and Outbound Re-Invites supported 44

45 CUCM Designs SIP Delayed Offer, G711 and G729 Regions No MTP Remote Site IP WAN G729 Central Site G729 Inter Region Codec CUBE SP WAN SIP Switch G711 Inter Region Codec All Fax Machines are in G711 Region All remote branch phones are in G729 Region Voice calls, Fax pass-through, T.38 Fax Relay, G711 MOH For Outbound Calls CUCM Codec Preference region dependent For Inbound Calls For G711 regions SP SIP switch Codec Preference 45

46 CUCM SIP Trunking Cube DO to EO CUCM Delayed Offer, G711 & G729 Regions Service Provider requires Early Offer 2.7XXX 5000 SIP Delayed Offer SIP Early Offer 5001 SIP MTP RTP Media Stream 7776 RTP Media Stream CUBE 5555 Voice Gateway SP SIP Switch 7777 SIP Early Offer (meaning SDP included in INVITE) required by Service Provider No MTP required in RTP path for Outbound CUCM calls G711 & G729 Regions between devices and SIP Trunk For CUCM outbound calls SP selects codec For Calls Inbound to Cisco Communications Manager SIP Delayed Offer calls SP selects codec SIP Early Offer calls CUCM selects codec Fax calls Fax Pass-through and T.38 Fax Relay Inbound and Outbound Re-Invites supported 46

47 CUCM SIP Trunking Cube DO to EO CUCM Delayed Offer, G711 & G729 Regions Service Provider requires Early Offer SIP Delayed Offer SIP Early Offer Remote Site IP WAN G729 Central Site G729 Inter Region Codec MTP G711 CUBE SP WAN SIP Switch G711 Inter Region Codec All Fax Machines are in G711 Region All remote branch phones are in G729 Region Voice calls, Fax pass-through, T.38 Fax Relay, G711 MOH For Outbound Calls SP selects codec For Calls Inbound to Cisco Communications Manager SIP Delayed Offer calls SP selects codec SIP Early Offer calls CUCM selects codec 47

48 Current SIP Trunk Recommendations and Strategy Cisco Communications Manager 5.X preferred and recommended over CM4.X implementations Cisco Unified Border Element (CUBE) is recommended as an Enterprise owned Border Element SIP Delayed Offer is preferred over SIP Early Offer Where SIP Early Offer is required by SP use CUBE DO to EO to avoid MTP usage CM7.0 Introduces: G729 MTP for SIP Early Offer Support for + character Privacy Asserted Identity (PAI) 48

49 Implementation Options CUCM to CUBE CUCM CUBE redundancy Call Volume CUBE Redundancy CUBE Gateway to Service Provider CUBE CUBE CUBE SIP Routing Entity 49

50 CUCM - CUBE Redundancy H.323 or SIP Define each CUBE device by its IP address SIP: Each CUBE GW is an IP address destination for SIP trunk H.323: Each CUBE GW is an H.323 Gateway Define one route group is defined for each CUBE Gateway; then, define one route list to encompass all route groups Use TCP for fast failover on Trunk failure or tune UDP timers CUBE Gateway 1 CUCM Cluster Route Pattern xxxx Route List Route Group 1 Route Group N CUBE CUBE CUBE Gateway N 50

51 Call Volume Add the number of CUBEs required CUBE CUBE CUBE SIP Proxy Server of Service Provider 51

52 Performance Capacity Recommendations for CUBE Gateway Platforms Platform CUBE GW Only (VAD ON) CUBE GW (VAD OFF) Multiple Features CUBE GW with Software MTP AS5000XM N/A

53 CUBE Gateway Redundancy HSRP 1:N redundancy HSRP 1:N CUBE CUBE CUBE CUBE Broadsoft AS/NS MCI NS/RS 53

54 CUBE Gateway to Service Provider Use DNS SRV _sip._udp.bsas IN SRV bsas1.vzb.com IN SRV bsas2.vzb.com bsas1 IN A bsas2 IN A DNS Server FQDN: bsas.vzb.com INVITE @bsas.vzb.com From: cluster_ipipgw.cisco.com CUBE CUBE INVITE @clusteripipgw.cisco.com From: bsas.vzb.com CUBE FQDN: cluster_ipipgw.cisco.com _sip._udp.cluster-ipipgw IN SRV ipipgw1.cisco.com IN SRV ipipgw2.cisco.com IN SRV ipipgw3.cisco.com ipipgw1 IN A ipipgw2 IN A Ipipgw3 IN A

55 Configuration 55

56 Basic CUBE Configuration 1. Enabling the IP-to-IP Calls SBC3845#config t SBC3845(config)# voice service voip SBC3845(conf-voi-serv)#allow-connections h323 to h323 SBC3845(conf-voi-serv)#allow-connections h323 to sip SBC3845(conf-voi-serv)#allow-connections sip to h323 SBC3845(conf-voi-serv)#allow-connections sip to sip 2. Mandatory to have Incoming and Outgoing VoIP Dial-peers with required parameters like Protocol, Transport, Codec, CAC, QoS, etc Incoming VoIP Call Outgoing VoIP Call dial-peer voice 1 voip destination-pattern 1000 incoming called-number.t session target ipv4: codec g711ulaw SBC dial-peer voice 2 voip destination-pattern 2000 session protocol sipv2 session target ipv4: codec g711ulaw 56

57 Agenda Technology Overview Deployment Scenarios Issues with SIP Trunks for PSTN Access What's in a SIP Trunk? What services should customers look for in SIP Trunks? What services should Service Providers offer with SIP Trunks? Using the Cisco Unified Border Element (formerly Cisco Multiservice IP-to-IP Gateway) as a Border Element between Service Provider and Customer Issues and configuration options for Cisco Unified Communications Manager Express (CUCME) with SIP Trunks for PSTN Access Issus and configuration options for Cisco Unified Communications Manager (CUCM) with SIP Trunks for PSTN Access General Troubleshooting Issues with SIP Trunks Resources 57

58 Cisco CUBE/SBC Portfolio Platform Support Cisco CUBE supports 1. SIP/H CUCM interworking 3. Demarcation 4. Security 5. Transcoding AS5000XM Cisco 7301 Cisco 7600 Cisco XR12000 Cisco 3800 ISR Cisco 7200 VXR Cisco 2600XM Cisco 2800 ISR Cisco 3700 Cisco SBC supports 1. NAT & FW traversal 2. Security 3. CAC & Policies 4. Media & Protocol interworking 5. IMS support(c-bgf, I-BGF, P-CSCF) Scale 58

59 CUBE Licensing 1 1 FL-CUBE-25 USD $2900 Cisco Unified Border Element License for up to 25 Sessions FL-CUBE-100 USD $9900 Cisco Unified Border Element License for up to 100 Sessions New Feature Licenses will be available to order on Voice bundles FL-CUBE-25 FL-CUBE-100 Licenses are additive, increments of 25 sessions Orderable on IOS images of IP Voice and up 59

60 Cisco Unified Border Element Enhanced Functionality with IOS Image Security Images IP Voice FL-CUBE License will be available as an option on several different IOS images. More powerful/expensive IOS images provide additional functionality. H323 H323 Voice Calls H323 to SIP SIP SIP Voice Calls Features Enhanced with IOS Image -IVS- Image SIP TLS Secure RTP SIP ALG and FIREWALL H.323 Gatekeeper Video Call Flow Flow Around Media 60

61 Additional Resources Cisco Multiservice IP-to-IP Gateway General Information (Datasheet, Q and A) Cisco Multiservice IP-to-IP Gateway Configuration Guide cts_installation_and_configuration_guides_list.html 61

62 Q and A 62

63 Recommended Reading Continue your Networkers at Cisco Live learning experience with further reading from Cisco Press Suggested books: Cisco Voice Gateways and Gatekeepers [ X] Cisco IP Communications Express: Cisco Communications Manager Express with Cisco Unity Express [ X] Available Onsite at the Cisco Company Store 63

64 Complete Your Online Session Evaluation Win fabulous prizes; give us your feedback Receive ten Passport Points for each session evaluation you complete Go to the Internet stations located throughout the Convention Center to complete your session evaluation Winners will be announced daily at the Internet stations 64

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