Transcript
SIP Trunking for IP PSTN Access
Peter Sakala
[email protected]
© 2009 Cisco Systems, Inc. All rights reserved.
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What is a SIP Trunk? A sampling of Views/Definitions Single IP based interconnect for voice and data using SIP SIP trunking is the IP equivalent of the digital/analog TDM connection that traditionally connected a PBX to the PSTN The logical session or channel established between a carrier and customer – (Porting PSTN Phone number to IP Address) A SIP Trunk service can be either – Managed – SP provides CPE equipment to monitor and guarantee SLAs in addition to basic voice services – “un”Managed – Similar to an analog phone line – provides basic voice services
Any SIP-based “connection” between two applications – Intra-enterprise: Between applications, e.g. MPlace to CUCM, or between different zones or departments within a company – Enterprise to SP: PSTN Access – B2B Inter-Enterprise: Between companies (e.g. Disney and Apple) © 2009 Cisco Systems, Inc. All rights reserved.
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Unified Communications Content Mapping SIP Trunk for PSTN Access CUBE
SMB A CUBE
A
CUBE CUBE
VoIP SP CUBE
Enterprise: Centralized SIP Trunk A
IP-PBX
Enterprise CUBE
Enterprise: Distributed SIP Trunk © 2009 Cisco Systems, Inc. All rights reserved.
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SIP Trunk Industry Update
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Industry Trends in “SIP Trunk for PSTN” Significant uptick in enterprise customer interest in SIP trunking – Numerous trial deployments – Increasing production deployments, mostly on low session counts
Video/SIP trunking for TelePresence offerings becoming available – ATT, TATA, – Increased interest in SIP trunk security features – FW, SRTP/TLS encryption, DOS attack mitigation
Increased interest in SIP normalization/manipulation as industrywide vendor/application interop continues to be problematic – SIP maturity is still some years off – Increasing interest in 3rd party PBX interop with Cisco SIP trunking solution – while we should position CUCM whenever possible, the PBX Interop lab does test CUBE with various IP-PBXs to provide interop info when required
Increased incidences of toll fraud on SIP © 2009 Cisco Systems, Inc. All rights reserved.
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SIP Trunking – Growth and Impeding Factors Growth
Impeding
Can be cheaper Physical access more versatile Capacity changes more dynamic Equipment consolidation Operational consolidation Improved redundancy New rich-media services Vendor/SP advocates Industry hype/pressure
Immature PSTN-equivalent services – – – – –
911 / 112 Fax/Modem MLPP MCID Fault monitoring/isolation
Number portability Poorly understood legal and geographical implications Inconsistent service delivery – Call-ID, recording
Unregulated service – Requires in-depth evaluation – Costs vary significantly based on geography and SP © 2009 Cisco Systems, Inc. All rights reserved.
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Current SP SIP Trunk Services Compared to TDM Services Consideration Basic call completion Suppl. services (Xfer, FWD, Hold, Conf)
SIP Trunk
TDM Trunk
Well defined
Well defined
Requires validation testing
Well defined
Fault Monitoring and Isolation
Options PING monitoring
Yellow/Red Alarms
Emergency Call (911) Handling
Special Handling per SP
Well defined
Not defined
Well defined
Malicious Call-ID (MCID) and Multi-level Priority and Preemption (MLPP) Caller-ID delivery
Inconsistent
Consistent
Voice Band Data
Modems/Baudot TDD ill-defined or unsupported
Well defined
Fax Technology
Industry interop issues
Well defined
SP dependent
Well defined
Deterministic traffic engineering. How are bursts handled? Who sends back equipment busy, enterprise or SP? Who provides announcements?
Porting numbers Geographic and legal dependencies of call routing Future rich media services Cost to enterprise for service Flexibility of call routing; site aggregation Security considerations © 2009 Cisco Systems, Inc. All rights reserved.
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Within single SP control
Well defined
Independent of geography but not of legislation
Geographically dependent
Great potential
No
Inconsistent
Well defined
Very flexible
SP dependent
IP considerations; toll fraud
Toll fraud
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Future SIP Trunk Services Technology possibilities of new features – – – – – –
Wideband codecs Video and Telepresence Presence SRTP/TLS Calls with subject lines Fixed Mobile Convergence (different endpoints)
Customer requests for additional voice services – Security (SRTP/TLS) – Fax
Industry currently working to get voice established – Most SPs have not discussed or unveiled plans for services beyond voice © 2009 Cisco Systems, Inc. All rights reserved.
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SIP Trunk Deployment Scenarios and Recommendations
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Agenda SIP Trunk Reference Architecture SIP Trunk Enterprise Connection Models SIP Trunk Deployment Topologies Recommended SBC Solutions and Best Practises
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Reference SIP Trunking Architecture SIP Proxy / NMS & Softswitch OSS
Services (Presence, VM etc)
Signaling ITP
PSTN
Media GW
CUBE
CUBE
CUBE
FW/NAT ALG
CUBE
SBC
FW/NAT ALG
CUBE
CUBE
FW/NAT ALG
SP-Managed |
SIP Trunk
A
CUCM
CUCME
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SBCS
IP PBX
TDM PBX
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SP Network | Customer Premise
Bearer
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Agenda SIP Trunk Reference Architecture SIP Trunk Enterprise Connection Models – Levels of Managed Services – Dedicated / Integrated Voice + Data – Centralized / Distributed Trunking
SIP Trunk Deployment Topologies Recommended SBC Solutions and Best Practices
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SIP Trunk SP Service Models SIP Trunk Service with L3 Router Demarc Managed access service providing an IP trunk between the SP network and a customer’s IP-enabled call agent
Service Provider Owned
VoIP SP SBC
SIP Trunk
Enterprise Owned
CUBE
CUBE
Customer Premises
A
A
CUCM
CUCME
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SBCS
IP PBX 13
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SIP Trunk SP Service Models SIP Trunk Service with L7 SBC Demarc Service Provider Owned
Managed access service providing an IP trunk between the SP network and a customer’s IP-enabled call agent
VoIP SP SIP Trunk
SBC
Customer Premises CUBE
CUBE
Enterprise Owned
CUBE
A A
CUCM
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IP PBX 14
SIP Trunk SP Service Models SIP Trunk Managed IP-PBX Service Service in which a customer’s premisebased IP-PBX, UC apps and dial-plan are operated and maintained by the SP
Service Provider Owned
VoIP SP SIP Trunk
SBC
Customer Premises A
Enterprise Owned
Managed CUCM © 2009 Cisco Systems, Inc. All rights reserved.
Managed CME/IP-PBX
Phones
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Security Exposure on Enterprise SIP Trunk Connection Models – Where Should I Firewall? SIP Trunk
A
SIP SP
CUBE
Increased Security Exposure
WAN Data
SIP Trunk
A CUBE
WAN Data
SIP Trunk
A CUBE
Internet Data
WAN SP
SIP + VPN SP
Recommended Deployment Models
SIP SP + Internet
Internet Voice A
Internet CUBE
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Internet Data 16
Cisco Unified Border Element
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Cisco Unified Border Element Architecture Actively involved in the call treatment, signaling and media streams
CUBE
IP
SIP B2B User Agent
Signaling is terminated, interpreted and re-originated Provides full inspection of signaling, and protection against malformed and malicious packets
Media is handled in two different modes
Media Flow-Through Signaling and media terminated by the Cisco Unified Border Element Transcoding and complete IP address hiding require this model
CUBE
IP
Media Flow-Through
Media Flow-Around
Media Flow-Around
Digital Signal Processors (DSPs) are required for transcoding (calls with dissimilar codecs) BRKVVT-2305_c1
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Signaling and media terminated by the Cisco Unified Border Element Media bypasses the Cisco Unified Border Element 18
Cisco Unified Border Element Basic Call Flow voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip
Originating Endpoint
Incoming VoIP Call
Terminating Endpoint
Outgoing VoIP Call CUBE
dial-peer voice 1 voip destination-pattern 1000 incoming called-number .T session target ipv4:192.168.10.50 codec g711ulaw
dial-peer voice 2 voip destination-pattern 2000 session protocol sipv2 session target ipv4:192.168.12.25 codec g711ulaw
1. Incoming VoIP setup message from originating endpoint 2. This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD, DTMF method, protocol, etc. 3. Match the called number to outbound VoIP dial peer 2 4. Outgoing VoIP setup message BRKVVT-2305_c1
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H.323 and SIP Layer 5/7 Demarcation
Demarcation
Back-to-Back User Agent Protocol-Independent Memory Structure Holding Call State and Attributes (CLID, Called #, Codec…) H.323/SIP Protocol Stack
H.323/SIP Protocol Stack
Extract Call-Related Parameters from Protocol Message, Discard Message and Update Call Memory
Build New Protocol Message and Insert Call-Related Parameters from Call Memory
Incoming Call Leg
Incoming
CUBE
dial-peer voice 1 voip description Incoming incoming called-number .T session protocol sipv2
BRKVVT-2305_c1
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Outgoing Call Leg
Outgoing dial-peer voice 4 voip description Outgoing destination-pattern 99.T session target ipv4:x.x.x.x session protocol sipv2
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Cisco Unified Border Element—More Than an SBC An Integrated Network Infrastructure Service TDM Gateway Cisco Unified Border Element Address Hiding
Voice and Video TDM Interconnect
H.323 and SIP interworking
PSTN Backup
DTMF interworking SIP security Transcoding
Routing, FW, IPS, QoS
CUBE
Note: An SBC appliance would have only these features
Unified CM Conferencing and Transcoding
WAN Interfaces
RSVP Agent
SRST VXML
GK
Note: Some features/components may require additional licensing BRKVVT-2305_c1
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Key Challenges When Interconnecting UC Networks
Why do I need a session border controller?
Session Mgmt
Demarcation
Real-time session Mgmt Call Admissions Control Ensuring QoS PSTN GW Fallback Statistics and Billing Redundancy/Scalability
Fault isolation Topology Hiding Network Borders L5/L7 Protocol Demarc Statistics and Billing
Interworking
Yours
H.323 and SIP SIP Normalization DTMF Interworking Transcoding Codec Filtering Fax/Modem Support
BRKVVT-2305_c1
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Mine
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Security Encryption Authentication Registration SIP Protection FW Placement Toll fraud
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Session Management
Call Admissions Control CUBE provides various different CAC mechanisms
Total calls, CPU, Memory, GK IP call capacity, max-connections, RSVP High Water Mark Low Water Mark
Total Calls, CPU, Memory CUBE
call threshold global [/mem/cpu] calls low xx high yy
gatekeeper endpoint circuit-id h323id IPIPGW1 AA max-calls 500
GK IP Call Capacity
GK
voice service voip allow-connections h323 to h323 h323 ip circuit max-calls 1500 ip circuit carrier-id AA reserved-calls 1000
CUBE
max-connections
Call #1 dial-peer voice 1 voip max-conn 2
Call #2 Call #3
BRKVVT-2305_c1
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Call #3 Rejected by CUBE 23
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Session Management
Quality of Service (QoS) Requirement Ensure traffic adheres to QoS policies within each network
The Cisco Unified Border Element can remark ToS/DSCP QoS parameters on signaling and media packets between networks dial-peer voice 100 voip ip qos dscp ef media ip qos dscp af31 signaling
Input Interface
Output Interface Police
Mark
Police
Mark
Classify
BRKVVT-2305_c1
Police
Mark
Police
Mark
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Queue Queue
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Shape
Queue
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SIP “Normalization” at the Network Border
Interworking
“Normalize” SIP traffic coming into the SP or Enterprise network at the border Use SIP profiles to translate messages CUBE
Enterprise SBC
VoIP SP 1
CUBE
SP–SP
SBC
VoIP SP 2
CUBE
CUBE CUBE
Small-Medium Business BRKVVT-2305_c1
Residential
IP-PBX
Small-Medium Business
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Interworking
SIP Profiles “Normalization”
SIP profiles is a mechanism to normalize or customize SIP at the network border to provide interop between incompatible devices SIP incompatibilities arise due to:
Add user=phone for INVITEs
A device rejecting an unknown header (value or parameter) instead of ignoring it A device sending incorrect data in SIP A device not implementing (or incorrectly) protocol procedures A device expecting an optional header value/parameter or can be implemented in multiple ways A device sending a value/parameter that must be changed or suppressed (“normalized”) before it leaves/enters the enterprise to comply with policies Variations in the SIP standards of how to achieve certain functions
Incoming INVITE sip:
[email protected]:506 0; SIP/2.0
Outgoing CUBE
INVITE sip:
[email protected]:5060; user=phone SIP/2.0
voice class sip-profiles 100 request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0" request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"
Modify a “sip:” URI to a “tel:” URI in INVITEs Incoming INVITE sip:
[email protected]:5060
Outgoing CUBE
INVITE tel:2222000020
voice class sip-profiles 100 request INVITE sip-header SIP-Req-URI modify "sip:(.*)@[^ ]+" "tel:\1" request INVITE sip-header From modify "
" "" request INVITE sip-header To modify "" ""
More information at www.cisco.com/go/cube > Configure > Configuration Examples and TechNotes BRKVVT-2305_c1
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Delayed Offer to Early Offer Interworking INVITE
Interworking
INVITE (Offer SDP) CUBE
180/183/200 (Offer SDP)
180/183/200 (Answer SDP)
SBC
SP VoIP
ACK/PRACK (Answer SDP)
voice class codec 1 codec preference 1 g711ulaw codec preference 2 … dial-peer voice 4 voip destination-pattern 321.... voice-class codec 1 voice-class sip early-offer forced session target ipv4:x.x.x.x
SP SIP trunk Early Offer (EO) interconnect for enterprise apps that support only Delay Offer (DO) Flow-through required for DE-EO supplementary services
Global Configuration Also Supported:
Early
Delayed
Offer
SDP in INVITE
No SDP in INVITE
Answer
SDP in 180/183
SDP in 200
BRKVVT-2305_c1
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Interworking
Media Transcoding Internet
Enterprise VoIP
SP VoIP CUBE
iLBC, iSAC, Speex
SBC
Cisco Unified Border Element supports universal transcoding
IP Phones: G.711, G.729, G.722
Transcoding: G.711, G.723.1, G.726, G.728, G.729/a, iLBC, G.722
Any voice codec to any other codec e.g. iLBC to G.711 or iLBC to G.729
Supported Codecs*
Voice transcoding only (not video)
G.711 a-law 64 Kbps
Transrating (different packetizations):
Supported: Transrating of different codecs e.g. G.711 a-law 20ms ↔ G.711 µ-law 10ms G.711 20ms ↔ G.729A 30ms
x
Not supported: Transrating of the same codec
Release
G.711 µlaw 64 Kbps G.723—5.3 and 6.3 Kbps G.729, G.729A 8 Kbps
12.4(11)XW and 12.4.20T
G.729B, G.729AB 8 Kbps iLBC—13.3 and 15.2 Kbps G.722—64 Kbps
e.g. G.729A 20ms ↔ G.729A 30ms
12.4(15)XY and 12.4.20T
*Note: Only voice codecs are supported with transcoding—no video codecs BRKVVT-2305_c1
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Demarcation
Demarcation at Network Borders SP UNI Codec Choice/Negotiation Fault Isolation Security
QoS Marking Voice Quality Statistics and Billing
Enterprise H.323
CTS
CUBE
Enterprise SIP Service Provider SBC
Meeting Place CUBE
IP PBX Enterprise/SMB
E-Partner
IP-PBX
IP-PBX BRKVVT-2305_c1
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Demarcation
Topology/Address Hiding 192.168.10.10
192.168.10.50
172.16.10.5
172.16.10.6
192.168.10.50
192.168.10.10
IP Inside
CUBE
CUBE
Inside
Outside
Site A—192.168.10.x/24
172.16.10.x/24
Site B—192.168.10.x/24
Requirements Maintain connectivity without exposing the IP network details Interconnect networks that have overlapping IP Addresses
B2BUA provides complete topology hiding on signaling and media Maintains security and operational independence of both networks Provides implicit NAT service by substituting Cisco Unified Border Element IP addresses on all traffic
BRKVVT-2305_c1
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Security
CUBE Security Protection Points
Identity / Service Theft
DOS B2BUA – L7 Inspection Call Volume/BW Limiting (CAC) Call Codec Limiting SIP Malformed Inspection SIP Listen Port Configuration RTP Malformed Topology Hiding Co-resident IOS: ACLs, FW, IPS
SIP Digest Authentication SIP Hostname Validation SIP Trunk Register CDR Toll Fraud Co-resident IOS: ACLs, COR
Voice Application Code L7 Protocol-independent memory structures holding call state and attributes (CLID, Called #, Codec…) Dial-peer
Dial-peer DTMF xlation Codec Filtering Xcoding Control
SIP/H.323 Protocol Stack
SIP/H.323 Protocol Stack
RTP Library
RTP Library
DSP API DSP Hardware
TCP UDP TLS
TCP UDP TLS
IOS Infrastructure (ACLs, FW, IPS, VPN) HW LAN/WAN Interfaces
Ingress I/F
Signaling BRKVVT-2305_c1
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Privacy SIP Header Manipulation Authentication and encryption (media) – SRTP Authentication and encryption (signaling) – TLS Co-resident IOS: All VPN features
Egress I/F
Media 32
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Security
SIP Protection Digest Authentication
CUBE
Invite [From< [email protected]>]
sip-ua authentication username xxx password yyy
SIP Proxy challenges INVITEs from the Cisco Unified Border Element to check endpoint validity with 401 Unauthorized The Cisco Unified Border Element responds with INVITE including credentials
100 Trying 401 Unauthorized Invite [Authorization: name, passwd] 100 Trying 200 OK
Hostname Validation Initial INVITEs with a hostname URI are compared to a configured list of up to 10 hostnames If there is no a match to the INVITE, the Cisco Unified Border Element returns a "400 Bad Request—Invalid Host" BRKVVT-2305_c1
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sip-ua permit hostname dns:example1.sip.com permit hostname dns:example2.sip.com permit hostname dns:example3.sip.com permit hostname dns:example4.sip.com
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Security
SIP Protection SIP Listening Port Default SIP Listen ports are 5060 (UDP/TCP) and 5061 (TLS) These ports are well-known and can be the target of attacks
voice service voip sip shutdown voice service voip sip listen-port non-secure 2000 secure 2050
Change the SIP Listen port to a different setting that is not well-known
Registration The Cisco Unified Border Element can send SIP REGISTER messages with credentials to a proxy Register statically on behalf of endpoints behind the Cisco Unified Border Element that do not register BRKVVT-2305_c1
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x(config)#sip-ua x(config-sip-ua)#credentials username 1001 password cisco realm cisco.com sip-ua registrar ipv4:172.16.193.97 expires 3600 credentials username 1001 password 0822455D0A16 realm cisco.com
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Toll Fraud—ACLs, Dial-Peers Use ACLs to allow/deny explicit sources of calls Apply explicit incoming and outgoing dial-peers to both CUBE interfaces to control the types and parameters of calls allowed on the network Use explicit destination-patterns on dial-peers (not .T) to block out disallowed offnet call destinations Use translation rules to ensure only valid calling/called numbers allowed Use Tcl/VXML scripts to do database lookups or additional checks to allow/deny call flows Change SIP port to something other than 5060 Is this a valid call flow to allow?
Close unused H.323/SIP ports Disable secondary dial-tone on TDM ports A
Incoming CUBE
192.168.10.10
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Incoming
172.16.10.6
access-list 1 permit 192.168.10.0 0.0.0.255 access-list 100 deny … (everything else) Explicit inc and outg dial-peers BRKVVT-2305_c1
SP VoIP
Outgoing
Outgoing
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access-list 2 permit 172.16.10.0 0.0.0.255 access-list 200 deny … (everything else) Explicit inc and outg dial-peers 35
Deployment Options
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Centralized/Aggregated SIP Trunk Model CUBE at central location Single SIP trunk IP address to SP All remote site calls hairpin through the campus site where SIP trunk terminates SP VoIP
PSTN
SBC
A
MPLS
CUBE
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HQ-SP RTP Branch-HQ RTP
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Distributed SIP Trunk Model CUBE at each site SIP trunk IP address per site Calls flow directly from site to SP SP VoIP
PSTN
SBC
A
Site-SP RTP
MPLS
CUBE
CUBE
CUBE
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CUBE
CUBE
CUBE
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Agenda SIP Trunk Reference Architecture SIP Trunk Enterprise Connection Models SIP Trunk Deployment Topologies – SMB – Enterprise
Recommended SBC Solutions and Best Practises
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SMB Deployment Models SIP Managed Voice Services SP-owned
Commercial Managed Voice Services
Customer-owned
FXS
VoIP SP 1
TDM PBX Interconnect
SIP Trunk Model – Managed Services (transparent to end customer) – Distributed (every site has a connection) – Redundancy: None – Capacity: <50 sessions
IP PBX
IP-PBX Interconnect
Border Element – – – –
SIP TDM GW IAD with FW/NAT IAD with CUBE-“light” CME with integrated SIP trunking
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Small Enterprise Deployment Models CME and CUCM
CME Centralized
CUCM Centralized
CME
A CUBE
CUBE
SRST
VoIP SP 1 CME Distributed
CUCM Distributed
CME
A CUBE
CUBE
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SRST
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Small Enterprise Deployment Models CME or CUCM SIP Trunk Model – Centralized – typically used when: • Cost benefits can be shown • SIP SP is different from WAN provider – Distributed – typically used when: • Survivability is important • SIP SP is the same as WAN (often MPLS) provider – Redundancy: None – Capacity: <200 sessions
Border Element – CME with integrated SIP trunking – Medium-range standalone CUBE or integrated SRST/CUBE
SRNDs: – www.cisco.com/go/interoperability > CUBE
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Medium and Large Enterprise Deployment Models Multi-Site and Multi-Cluster CUCMs
Multi-cluster, Multi-site CUCM A SBC
A A
Multi-site CUCM Centralized (hybrid)
A
SBC
A
SBC
A A
A
SBC
SRST SRST
VoIP SP 1
Centralized (hybrid) A
Multi-site CUCM Distributed SBC
A A
A
SBC
A
SBC
A SBC
A A
SBC: SRST
SRST CUBE
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Other Enterprise Deployment Models IP-PBX and TDM-PBX
IP-PBX Centralized
TDM-PBX Centralized GK CUBE CUBE
VoIP SP 1 IP-PBX Distributed
TDM-PBX Distributed GK CUBE CUBE
CUBE
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CUBE
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Medium and Large Enterprise Deployment Models SIP Trunk Model – Centralized – typically used when: • Cost benefits can be shown • SIP SP is different from WAN provider – Distributed – typically used when: • Survivability is important • SIP SP is the same as WAN (often MPLS) provider • Geographic considerations – Redundancy: Generally must-have – Capacity: • Medium Enterprise: 500-1500 sessions at campus/data center sites • Large Enterprise: 1500-5000 sessions at campus/data center sites • Very Large Enterprise: 5000+ sessions at campus/data center sites • 10-100 in remote sites
Border Element – Medium-Large Campus/Data Center: CUSP+CUBE cluster or CUBE on ASR – Large-Very Large Campus/Data Center: CUBE on ASR – Remote sites: High-end standalone CUBE; integrated SRST/CUBE
SRNDs: – www.cisco.com/go/interoperability > CUBE
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Agenda SIP Trunk Reference Architecture SIP Trunk Enterprise Connection Models SIP Trunk Deployment Topologies Recommended SBC Solutions and Best Practises – SBC Product Positioning – Determining an SBC Recommendation – SBC Redundancy Options – CUCM Best Practises – SBC Best Practises
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Cisco Unified Border Element Portfolio Cisco Unified Border Element (Enterprise Edition) provides SBC features for enterprise implementations CUBE (Ent)
CUBE (SP)
Cisco Unified Border Element (Service Provider Edition) provides SBC features for carrier class service provider implementations
CPS
7600
ASR 1000 Series
50,000/Blade 250,000/System
AS5000XM
3800 ISR 2800 ISR 7200VXR 7201, 7301
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Session Capacity
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CUBE (Enterprise Edition) Portfolio ASR 1000 Series
CPS
50+
AS5000XM
8-12
3800 ISR 7200VXR 7201, 7301 2800 ISR
<5
<250
500-800
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Session Capacity
5000+ 48
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CUBE ASR
CUBE + CUSP
CUBE ISR
Large-Scale SIP Trunks SP SIP
CUCM SIP Trunk
SP SIP Trunk CUBE
CUCM A
CUBE
SBC
CUBE CUBE
CUBE Cluster
SP SIP Trunk
SP SIP
CUCM SIP Trunk CUCM A
SBC
CUBE CUBE CUBE CUBE
CUBE Cluster
SP SIP SP SIP Trunk
CUCM SIP Trunk
CUCM A
SBC
CUBE (Ent) © 2009 Cisco Systems, Inc. All rights reserved.
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CUBE (Ent) Solution Advantages (subject to change)
ISR
5350XM/5400XM
ASR1002/4/6
• Collocated features • TDM GW • Tcl / VXML • Add’al collocated features • SRST • MTP • IOS FW • T.38 fax • H.323 • Video/TP • DSP features • Transcoding • In-band tone DTMF • Transrating (upcoming) • Voice quality scoring (upcoming) • GK Support • Cost-effective geographic (1+1 and N+1) redundancy
• Collocated features • TDM GW • Tcl / VXML • T.38 fax • H.323 Support • Video/TP • DSP features • Transcoding • In-band tone DTMF • Transrating (upcoming) • Voice quality scoring (upcoming) • GK Support • Cost-effective geographic (1+1 and N+1) redundancy • Footprint (5350XM 1RU)
• Scalability • Inbox redundancy • ASR1002/4: SW failover with media preservation • ASR1006: HW failover with media preservation • Footprint (2/4/6 RU)
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Summary SIP Trunk Sizing Recommendations (subject to change)
Enterprise Size
SIP Trunk Sessions
Redundancy Recommendation
<100
None
Single 2811, 2901
100-200
None
Single 2851, 2911
200-500
None
Single 3845, 2951
500-1000
Optional
1000-2000
Must-have
Inbox redundancy: Single ASR1002 Geo redundancy: Dual ASR1002 or future ISR G2*
Large
2000-4000
Must-have
Inbox redundancy: Single ASR1004/6 RP2 Geo redundancy: Dual ASR1004/6 RP2
Very Large
4000+
Must-have
Inbox redundancy: Single ASR1006 RP2 Geo redundancy: Dual ASR1006 RP2
Small
Medium
Platform Recommendation
No redundancy: Single 3945 Redundancy: Dual 3945
*Future: 1H 2010 3945 with new SPE-xxx © 2009 Cisco Systems, Inc. All rights reserved.
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Redundancy Options: None A A
SIP SP
SIP SP
Aggregate SBC capacity is equal to trunk capacity – E.g. 4 boxes @ 500 each = 2000 session SIP trunk – Full trunk capacity guaranteed only when ALL boxes are up
Failure impact – single-box solution: – All connections dropped; SIP trunk out of service – No new calls until recovery
Failure impact – multiple box solution: – % of connections dropped – New calls handled with reduced SIP trunk capacity © 2009 Cisco Systems, Inc. All rights reserved.
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Redundancy Options: 1+1 A
SIP SP
A
SIP SP
Active/Active
Active/Standby
Active/Standby (HSRP) – HSRP can work for intra-enterprise solutions, but is not recommended for SP SIP trunks
Active/Active (Load balancing) – Special case of N+1 redundancy (next slide) – SP SIP trunks usually offer only 2 IP addresses – if more than 2 boxes are needed to guarantee SIP trunk session capacity, then a CUSP+CUBE solution is recommended
Local/Geographic Considerations – HSRP provides local redundancy only – Load-balancing Act/Act can provide local or geographic redundancy
Failure impact: – All existing connections on failed box are dropped; no stateful failover – New calls are immediately handled with full SIP trunk capacity © 2009 Cisco Systems, Inc. All rights reserved.
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Redundancy Options: Inbox ASR1006
ASR1002/4 Active OS
Standby OS
Dual Forwarding plane HW Dual Control plane HW (CPU)
A
SIP SP A
SIP SP
SW Inbox redundancy: ASR1002 + ASR1004 HW Inbox redundancy: ASR1006 – Control plane (CPU or RP) – Data/Forwarding plane (packet forwarding)
Failure impact – CUBE (Ent): – Media preservation for existing calls – New calls handled immediately with full SIP trunk capacity © 2009 Cisco Systems, Inc. All rights reserved.
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N+1 and N+M Redundancy Options No, or 1+1 redundancy on CUSP
A
SIP SP
A
SIP SP
… …
N routers to guarantee session capacity M routers to protect against M simultaneous failures
CUBEs can be ISRs or ASRs Local or Geographic redundancy – CUBEs can be distributed across sites as needed
Use a load balancing algorithm in the attached call agent (or use DNS or CUSP) to distribute calls over pool of CUBEs Failure impact: – New calls are handled immediately with full SIP trunk capacity © 2009 Cisco Systems, Inc. All rights reserved.
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CUCM Best Practises CUCM 6.x or 7.x recommended for SIP trunking – H.323 CUCM interconnects to SIP trunks not recommended
H.323 or SIP SBC interconnects with non-Cisco IP-PBX or TDMPBXs can be used CUCM Configuration – Delayed Offer (no MTP) for CUCM outbound calls – Early Offer (no MTP) for CUCM inbound calls to CUCM – SBC Delayed Offer to Early Offer interworking
Configure alternate PSTN routing if SIP trunk is down – Recommend not to remove TDM PSTN GWs until after a SIP trunk has been proven in
If xcoding is required – CUCM-controlled xcoding is the more flexible option for SBC engineering purposes – SBC xcoding is more flexible in codec combinations © 2009 Cisco Systems, Inc. All rights reserved.
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CUBE (Ent) Best Practises (1) Always discuss the trade-offs of centralized and distributed SIP trunk design Always try to do a POC of a SIP trunk connection MTPs: – Avoid MTP designs if possible; if not, collocate MTPs with CUBE (Ent) to optimize the media path
Integration or dedicated CUBE (Ent) – At low end (<500), MTP, VXML, FW, SRST easily integrated with CUBE (Ent) – At >1000 sessions, it’s often better to dedicate platforms to each function
CUBE (Ent) Performance Engineering – H.323-SIP vs. SIP-SIP makes no significant difference – DTMF interworking or DO-EO adds no significant extra load – SIP profiles and Tcl tend to be fairly “light” on the CPU, but is configuration dependent – MTP, Xcoding and SRTP-RTP conversion are CPU-intensive © 2009 Cisco Systems, Inc. All rights reserved.
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CUBE (Ent) Best Practises (2) Use SIP registration on the trunk if offered by the SP, it offers better security Define explicit incoming and outgoing dial-peers Deploy IOS UC features and techniques to mitigate tollfraud
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CUBE (Ent) Best Practises (3) CUBE (Ent) and FW placement – Campus/Data center sites: Place CUBE (Ent) behind the FW – Remote/small sites: Enable IOS FW integrated on CUBE (Ent)
Redundancy Best Practises – Centralized SIP trunking: Redundancy always recommended, regardless of session capacity – Distributed SIP trunking: Redundancy recommended at sites with >1000 sessions
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