design voice gw
TRANSCRIPT
1© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2
Designing Voice Gateways and Media Resources in Enterprise Networks
VVT-2010
2© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 2© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Agenda
• GW Platforms and Voice Interface Capabilities
• GW Design: Features, Protocols and Operation
• Router-Based Media Services
• Router-Based Applications
3© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 3© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Scope of This Seminar
• Understand PSTN/PBX voice gateway capabilities, choices and design considerations in Cisco CallManager enterprise networks
• Understanding what can be built today, and how to build it• Learning the features that gateways provide to the IP network• Understand router-based IP voice services and capabilities to
build the infrastructure of your IP communications networkDSP/media services, RSVP, SRST, CME, voice mail
Cisco CallManager®
Router/GW Router/GW
Applications Applications
PSTN
IP WAN
PBXPBX
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Platforms and Voice Interface Capabilities
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Cisco Voice Gateways
SMB/Small BranchCisco 2801
1751, 1760
Cisco 3700 Series
Cisco 2800, 2600XM, 2691 Series Cisco VG248
Cisco Catalyst®
6500 CMM
Cisco ATA 186
Cisco VG224
Enterprise Branch
Teleworker/SOHO
Large Branch
Enterprise Campus
and BeyondCisco 3800 Series
Cisco 5350XM, 5400XM, 5850
Router-Based GWs
Dedicated GWs
Switch-Based GWs
Typical Deployment
Perf
orm
ance
and
Ser
vice
s
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Voice Gateway Selection Criteria
Platforms Typical Interfaces
Common Deployments
Feature Set Focus
Service Provider GWs
Cisco 5350XM, 5400XM, 5800
T1/E1, T3, STM-1
T1/E1
Analog, BRI, T1/E1
FXS
End User ATA186 FXS Teleworker Phone Set Features
SP Networks, Softswitches,
GW-to-GW,PSTN GW
Service Provider
Voice
Enterprise Campus GWs
Cisco Catalyst6500 CMM, Cisco 3800
Series
CCM,PSTN/PBX GW
Enterprise Voice
Enterprise Branch Office GWs
Cisco 2600, 3700, 2800, 3800 Series
CCM,Enterprise GW-to-
GW,PSTN/PBX/Key GW
Enterprise Integrated
Voice+Data
Analog GWs VG224/248 CCM Phone Set
Features
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Gateway Signaling Protocol Summary
• Signaling protocol and interface support varies across the GW platforms
• T3/STM-1 capability is only available on the Cisco 5x00 series GWs
• A subset of TDM protocols is supported with MGCP and varies depending on the call agent used with the GW
FXSFXOE&M: wink,
immediate, delay dial
Analog DIDCAMA
BRI: Q.931, QSIGT1: CAS, FGD, PRI,
QSIG, PRI NFASE1: PRI, QSIG, R2,
CAS/MELCASJ1T3/STM-1
SIPH.323MGCP 1.0: SP call
agentsMGCP: 0.1+: CCM
VoIPAnalog TDM Digital TDM
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FXO
T1/E1
BRI
NM
-HD
-2V
8
4
NM
-HD
A
8
NM
-HD
-1V
4
2
NM
-HD
-2VE
8
4
1
4
2
4
NM
-HD
V2
EVM
-HD
12
8
NM
-HD
V*
2
VIC
* / V
IC2
4
2
VWIC
, VW
IC2
2
Station-Side
Trunk-Side
GW Interface Card Port DensitiesCisco 2600, 3700, 2800, 3800 Series
8 124 8 4244FXS
Note: EOS/EOL For Some of These Cards Have Been Announced
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Analog/BRI Voice Interface Card (VIC) Considerations
VIC VIC2(Including VIC-4FXS/DID and VIC-2DID)
Minimum Release 12.0T 12.3T
Cisco 1700, 2600XM, 3700, 2800, 3800
NM-HD-1V/2V/2VE, NM-HDV2
Two- and Four-Port cards
FXO Separate FXO cards For Different Geographies
Single FXO Card for All Geographies
FXS/DIDSeparate FXS and DID Cards (Except the VIC-
4FXS/DID)SW-Configurable Ports on the FXS/DID and DID Cards
FXO/CAMA Separate FXO andCAMA Cards
SW-Configurable Ports on the FXO Cards
Platform Support
Cisco VG200, 1700, 2600, 3600, 3700
NM Support NM-1V/2V
Port Density Two-Port
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T1/E1 Voice/WAN Interface Card (VWIC) Considerations
VWIC VWIC2Minimum Release 11.3 12.4
Cisco 1700, 2600XM, 3700, 2800, 3800
NM-HDV, NM-HD-2VE, NM-HDV2
Drop and Insert
Only Supportedon–DI Cards Supported on All Cards
Clocking Single External ClockPer Card
Dual External Clocks (For Data Ports) Per Card
T1/E1 Separate Cards for T1 and E1 Operation
SW-ConfigurableT1/E1 Ports
Optional HW ECAN No Yes
Platform Support
Cisco VG200, 1700, 2600, 3600, 3700, 2800, 3800
NM Support NM-HDV, NM-HD-2VE, NM-HDV2
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High-Density Analog/BRI Voice (EVM-HD)• Supports high-density FXS, FXO,
Analog-DID and BRI ports• Baseboard: EVM-HD-8FXS/DID
Software-configurable as FXS or DIDPlug in zero, one or two expansion modules—in any combination
• Expansion modules:EM-HDA-8FXS (shared with NM-HDA)EM-4BRI-NT/TEEM-HDA-3FXS/4FXOEM-HDA-6FXO
• Use router motherboard DSPs• Supported on the 2821, 2851, 3825
and 3845Not supported on the 2801 or 2811Max one EVM on the 3825 and max two on the 3845
RJ21 Connector
EM 0 EM 1
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EVM-HD Expansion Module Combinations
Total PortsBase Board BRI
8 FXS or DID Ports B-Ch— — 8 8
EM-HDA-8FXS 8 8 16EM-HDA-8FXS EM-HDA-8FXS 8 16 24EM-HDA-8FXS EM-HDA-3FXS/4FXO 8 11 4 23EM-HDA-8FXS EM-HDA-6FXO 8 8 6 22EM-HDA-8FXS EM-4BRI-NT/TE 8 8 4 8 24
EM-HDA-3FXS/4FXO - 8 3 4 15EM-HDA-3FXS/4FXO EM-HDA-3FXS/4FXO 8 6 8 22EM-HDA-3FXS/4FXO EM-HDA-6FXO 8 3 10 21EM-HDA-3FXS/4FXO EM-4BRI-NT/TE 8 3 4 4 8 23
EM-HDA-6FXO - 8 6 14EM-HDA-6FXO EM-HDA-6FXO 8 12 20EM-HDA-6FXO EM-4BRI-NT/TE 8 6 4 8 22EM-4BRI-NT/TE - 8 4 8 16EM-4BRI-NT/TE EM-4BRI-NT/TE 8 8 16 24
EVM-HD-8FXS/DID
ModuleFXS or DID FXS FXO
EM 0 EM 1
Combine Individual Components in Any Order
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DSP Architecture of the ISRs• Each DSP slot can house one of five
PVDM2 types with increasing voice channel density
PVDM2-8, PVDM2-16, PVDM2-32, PVDM2-48, PVDM2-64
• NM-HDV2s can be used to expand ISR DSP capacity
• DSPs can be shared—to some extent—between motherboard and NM-based interfaces
Platform Onboard PVDM2 slots
NM-Based PVDM2 slots
044816
2801 22811 22821/51 33825 43845 4
Power + 802.3af VPN AIM AIM
HWIC
HWIC
HWIC
HWIC
GE
GE
USB
USB
SFP
NM
NM
NM-HDV2
Onboard DSP Slots (2, 3 or 4), Accessed by the EVM and
Voice Cards in the HWIC Slots
Each NM-HDV2 Provides Four More DSP Slots
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IP Phone Inline Power Support• AC or AC+IP power options, e.g.
CISCO2811—no IP powerCISCO2811-AC-IP—has IP power
• Four to forty eight-port LAN switching• 802.1Q, 802.1P, up to 15 VLANs• Up to 15.4W per switch port • 802.1x port-based authentication• Up to two Etherswitches of any form
factor per platform• Stack through external cable for VLAN
database consistency
HWIC-4ESW-POEHWIC-D-9ESW-POE
NM-16ESW-PWR NMD-36-ESW-PWR
NME-16ES-1GENME-X-24ES-1GENME-XD-24ES-2STNME-XD-48ES-2GE
ChassisCisco POE
Support
802.3af POE
SupportPower (W) Max # POE
SwitchportsMax # IP Phones
(7W)
2801 HWIC HWIC 120 16 162811 HWIC/NM HWIC 160 24 222821 HWIC/NM HWIC 240 32 322851 HWIC/NM HWIC 360 56 513825 HWIC/NM HWIC 360 72 513845 HWIC/NM HWIC 360 96 51
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GW Design: Features, Protocols and Operation
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Agenda
• Gateway Deployment Scenarios
• Which GW Protocol? SIP, H.323 and MGCP
• Gateway Availability Considerations
• Supplementary Services with GWs
• Fax/Modem Capabilities
• Specialized Gateway Capabilities
• Gateway Capacity
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Gateway-to-Gateway Network Deployments
• Late 1990s VoIP and VoFR “toll bypass”transport deployment— before call control became IP-based
• PSTN connection moves from PBX to voice gateway
On-net enterprise traffic leverages VoIP instead of PSTN or inter-PBX TDM TIE-lines
• H.323 networksSmall meshed networksLarge network deployed with gatekeepers
• Dial plan is distributed across GWs, or [optionally] centralized in a GK
• Enterprise: Transparent-CCS transport for carrying proprietary inter-PBX protocols
• SP: Large VoIP transport networks
PBX
Branch Office
Campus
PBX
PSTNIP
H.323
GK
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CCM Network Deployments
• Centralized or distributedCCM scenarios
• ICT: H.323 or SIP• GWs: H.323, SIP or MGCP• Dialplan is primarily
centralizedGW dial-peers point to CCMfor call routingGWs have backup dial-peersfor call routing when CCM is unavailable
• GW platforms often integrate other IP-based voice services
SRST, conferencing,transcoding, RSVP CAC
Branch 1Cluster 1
Branch 2Cluster 1
Branch 1Cluster 2
Branch 2Cluster 2
CCM Cluster 2
GK
ICT
IPPSTN
CCM Cluster 1
H.323 or SIPMGCP, SIP or H.323
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CCM and CME Network Deployments
• One or more CCM andCME sites
• CCM and CMEICT or H.323/SIP trunks
• CCM GWsH.323, SIP or MGCP
• CME GWsH.323 or SIP
• Dialplan is distributedCCM and CME does call routing
Optional GK may help withoverall intersite call routing
CCM Cluster
GK
CME Branch 1 CME Branch 2
ICT
PSTN IP
H.323, SIPMGCP, SIP, or H.323SCCP or SIP
SRSTBranch 1
SRSTBranch 2
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Distributed Contact Center Deployments
• Provides access to centralized IVR applications via branch office• Provides edge queuing/IVR for contact center solutions—keeps
RTP off WAN until agent is selected• Deflects call to agent when necessary• Supported with H.323 (SIP in future)
CVCVPP
Database
dp 1 TCL/VXML
dp 2 H.323
Active Voice Call (speechpath)
GW Control Via
HTTP/VXML
When Agent Available, Active Voice Call Rerouted
Via H.323 to Agent
RTP (VXML controlled)RTP (H.323 controlled)
VXML
IP
PSTN
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Agenda
• Gateway Deployment Scenarios
• Which GW Protocol? SIP, H.323 and MGCP
• Gateway Availability Considerations
• Supplementary Services with GWs
• Fax/Modem Capabilities
• Specialized Gateway Capabilities
• Gateway Capacity
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• Peer-to-peer call setup—IP address servers optional
• GW TDM signaling types local to GW• Resilient over IP connectivity failures• Scalable• Distributed configuration
CCM
• Call agent arbitrates call setup between endpoints
• GW TDM signaling types dependent on call agent
• Dependent on IP connectivity between endpoint and call agent
• Call agent bottleneck• Centralized configuration
Call Control Models
Optional Signaling to Locate IP Address of Peer Call Control Signaling Media (RTP)
GK
GW
H.323 Endpoint
IP
PSTN
SIP Endpoint
IP Address Servers
IPSCCP
Endpoint
Centralized Call Agent
GW PSTN
MGCP GW
Distributed Call ControlPeer-to-Peer Protocols: H.323, SIP
Centralized Call ControlClient-Server Protocols: SCCP, MGCP
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The Power of Cisco IOS Dial-Peers: H.323 and SIP
• Switch calls intelligently if required (interpret the dial plan)• Digit manipulation (called, calling and numbering plan)• Failover (preferences) to alternate destinations• Load balancing• Video ISDN switching• Insert applications into the call path: TCL/VXML
Build support for signaling variations (e.g. CLID on T1 CAS)Hookflash trunk release on FXOVXML call control for call centersRedistribute calls-in-q for CVPAA in the GW
IPPSTN
dp 1 voip
dp 2 voip
dp 3 voip
dp 10 pots
dp 11 pots
dp 12 pots
These Capabilities do Not Exist for MGCP-
Controlled GWs
Dial-Peers Allow You To:
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GW Protocol Features with CCM
Legend:Yes
No
With Caveats
H.323 SIP MGCPCentralized Provisioning
QSIG Supplementary Services
Centralized CDR (DS0 Granularity in CCM CDR)
MLPP (Preemption)
Hookflash Transfer with CCM
ISDN Overlap Sending No GK
ISDN Fac IE Name Display
NFAS
AT&T Megacom NSF Partial Partial
SRTP (CCM to GW) CCM 5.0 Future
Mobility Manager VXML-Based Voice Profile Management Test
Interstar Xmedius T.38 FaxServer Test
Caller ID on FXO Future
TCL/VXML Apps (e.g. CVP Integration) CVP 4.0
Voice+Data Integrated Access
Fractional PRI Workaround
TDM: A-DID, E&M, PRI NFAS, CAMA, T1 FGD
TDM T3 Trunks
ISDN Video Switching on GW
Set numbering Plan Type of Outgoing Calls
H.320 Video Future
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Protocol Deployment Considerations
• Characteristics of larger site(s)—often best served by MGCP
• High-density GWs to PSTN,often PRI
• Dedicated GW platforms• Caller ID/name delivery required• Digital TDM protocol (often PRI)• QSIG connectivity (with
supplementary services) tolegacy PBXs required
• Other considerationsNFAS is H.323/SIP onlyVery high density GWs such as T3 (5x00) are H.323/SIP only
• Characteristics of branch site(s)—often best served by H.323/SIP
• Low-density GW to PSTN, often analog
• GW and router features used on same platform (integrated access)
• Caller ID on analog FXO required• Mixes of PSTN TDM protocols
required (FXO, A-DID, BRI, Frac-PRI—)
• CVP/VXML application control • Other considerations
Can mix H.323 and MGCP on thesame GW (not on same voice port)H.323 dial-peers are needed anywayfor MGCP GW Fallback
Large/Campus Sites
Small/Branch Sites
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Protocol and Platform Summary
Line Side Trunk Side
VG224 Yes Yes Yes YesVG248 Yes No No No
7x00 No Yes Yes No
1751/60 No Yes Yes Yes2600XM, 2691 No Yes Yes Yes
3700 No Yes Yes Yes
Gateway Platform SCCP (FXS) H.323 SIP MGCP (CCM)
Yes Yes
YesNo
Yes
Yes5x00 No Yes Yes
Cat 6K CMM No Yes Yes
2800 Yes Yes
3800 Yes Yes
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Agenda
• Gateway Deployment Scenarios
• Which GW Protocol? SIP, H.323 and MGCP
• Gateway Availability Considerations
• Supplementary Services with GWs
• Fax/Modem Capabilities
• Specialized Gateway Capabilities
• Gateway Capacity
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IP Phone FailoverSurvivable Remote Site Telephony (SRST)
• Phones have list of backup “CCMs” to fail over to in case of no response to keepalives
SRST is the “CCM of last resort” in the phone list
• SRST only controls IP phone connectivity—it does not provide or control GW connectivity or availability
• PSTN GW connectivity during failure modes:POTS/VoIP dial-peersMGCP GWs requires the MGCP GW failover featureCalls from IP phones (under SRST) access the dial-peers to route calls
A
PSTN
CCM Cluster
WANXSCCP Keepalive to
CCMSCCP Keepalive to SRST if
CCM Does Not Respond
Dial-Peers Control GW Call Routing
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Voice Gateway Failover
• H.323/SIP intrinsically resilientNo special “failover” features, just dial-peer configuration
• Successive VoIP/POTS dial-peers (by preference) attempted on failure
• Each call setup attempt is treated independently
Same sequence of dial-peers
• Flapping IP links do not interfere significantly with call setup operation
• MGCP requires “failover”features
• When call agent is out of contact, MGCP GW fails over to local control
H.323, SIP or POTS dial-peersISDN D-channel is reset to gain control of call state
• Flapping IP links interfere significantly with call setup
Call agent registration and state are affected
PSTN CCM
GK
SIP Proxy
IP123
4 XVoIP Dial-PeersPOTS Dial-Peer
H.323/SIP MGCP
PSTN CCM
IPXGW Switches to
H.323/SIP/POTS Dial-Peers During MGCP Fallback
MGCP Registration, Keepalives and Backhaul
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IOS GW Audio Preservation Summary
• Audio is preserved with no supplementary services• Audio is preserved as long as the RTP stream is not
interrupted by the failure• SIP is still being tested—behaves like H.323
H.323 MGCPPrimary-to-Secondary CCM Failover
ISDNCCM to SRST
Non-ISDN Preserved
Flapping Links 12.4.4XC/12.4.9T, CCM4.1.3-SR2 preserve
Failures Depend on Timing
VXML Calls in Queue TCL Script to Reroute to SRST Hunt-Group
12.4.4XC/12.4.9T, CCM4.1.3-SR2 preserve
N/A
Preserved
FailPreserved(Disable TCP timer)
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Agenda
• Gateway Deployment Scenarios
• Which GW Protocol? SIP, H.323 and MGCP
• Gateway Availability Considerations
• Supplementary Services with GWsQSIG Supplementary Services
“Hookflash” Transfer
Station-and Trunk-Side FXS
• Fax/Modem Capabilities
• Specialized Gateway Capabilities
• Gateway Capacity
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QSIG Supplementary Services
H.323 Only• GW-to-GW• ECMA and ISO QSIG• Router/GW does not interpret
the QSIG supplementary services, it merely tunnels it across IP (H.323) to the far-end PBX for interpretation
MGCP Only• GW-to-CCM• ISO QSIG• Router/GW does not interpret
the QSIG supplementary services, it merely tunnels it across IP (MGCP) to CCM for interpretation
QSIGQSIG
QSIG Supplementary Services between PBXs
QSIG Services Tunneled Via H.323
QSIG
QSIG Supplementary Services between PBX and CCM
QSIG Services Tunneled Via MGCP
“Toll Bypass” QSIG CCM QSIG
IP IP
Note: QSIG Basic Calling and Caller ID Work on SIP GWs to CCM, but Most Supplementary Services Are Not Supported
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“Hookflash” Transfer
• Analog FXO PSTN trunks• Requires PSTN service for HF xfer• PSTN xfers the call, releases trunk to GW• Requires custom-developed
TCL script
• Analog FXS/FXO (or T1 FXS/FXO)• GW: Rx HF on FXS; Tx HF on FXO• “Toll bypass” scenario: CCM does not
support H.323 HF relay• PBX xfers the call, releases trunk to GW• Requires H.245-signal DTMF relay
• Uses PSTN “HF” accesscode service, e.g. *8
• CVP instructs GW to send digits to PSTN inband–T1 CAS/PRI trunks
• PSTN xfers call, releases trunk to GW
2. RedirectPSTN
TCL
1. Call to CUE AA
3. HF
1. Call from PBX to Analog Phone
2. HF on FXS
3. H.323 HF Relay
4. HF on FXO5. Xfer
PSTN
VXML
1. Call from PSTN to IVR
2. Send *8nnn (OOB)
3. *8nnn (inband)
4. Xfer
IP
IP
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“Hookflash” Transfer
• Requires PSTN PRI TBCT (2-b-channel xfer) service
• CVP instructs GW to initiate TBCT to PSTN–PRI trunks
• PSTN xfers call, releases trunk to GW• Requires TCL script
• Xfer calls from TDM IVRs and voice mail systems
• IVR/VM sends HF to GW; GW passes on to CCM in MGCP notification
• CCM xfers call, IVR releases trunk to GW• T1 E&M with MGCP only
IP
IPPSTN
VXML
1. Call From PSTN to IVR
2. Do TBCT4. Xfer TCL
3. TBCT
1. Call to TDM IVR/Vmail system
4. Xfer2. HF on T1 CAS
3. MGCP HF Notification
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Station and Trunk-Side FXS Gateways
• Station-Side FXSSCCP control of FXS ports–ATA, VG224, VG248, IOS GW FXS portswith 12.4.9TUse this for “featured” analog phonesSupplementary features like transfer, conference, call waiting, call park, FAC codes, MWI stutter dial-tone, redial
• Trunk-Side FXSMGCP, H.323, SIP control of FXS ports–IOS GW FXS portsBasic Call, CLID and hookflash blind transfer
ATA
VG224
SCCPSCCP
FXS
FXS
Featured Phone
Station-Side FXS Trunk-Side FXSPOTS Phone
PBX
Key System
FXS
FXS
MGCP, H.323, SIP
IP
36© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 36© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Agenda
• Gateway Deployment Scenarios
• Which GW Protocol? SIP, H.323 and MGCP
• Gateway Availability Considerations
• Supplementary Services with GWs
• Fax/Modem Capabilities
• Specialized Gateway Capabilities
• Gateway Capacity
37© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 37© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Fax over IP Methods
• Fax passthrough: Negotiation and switchover in H.323 and SIP protocolNo negotiation and switchoverin NSE
• Fax-relay Cisco:Switchover in NSE
• Fax-relay T.38 Negotiation and switchover in H.323 and SIP protocolNegotiation in protocol H.323, SIP and MGCP and switchover in NSENo negotiation and switchoverin NSE
Capability Negotiation Using the Protocol Stack
Voice Call Established
Detects 2100Hz CED Tone From Answering Fax machineRequest Switchover
Originating Terminating
Acknowledge Switchover
38© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 38© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Switchover to Fax over IP
• Switchover done via call control protocolIn H.323 the switchover is indicated using the RequestMode H.245message; in SIP the switchover is indicated using a ReInviteBoth gateways involved in the call should use the same protocolused when interoperating with third party devicesCall agent must support the fax method negotiated
• Switchover done via NSENSE events are used to indicate the switchover to passthrough, Cisco fax relay or T.38 fax relayIndependent of the call control protocol, therefore best for mixed protocol environmentNSEs are Cisco specific and can only be used with other Cisco endpointsDoes not require call agent to support the fax method
39© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 39© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Relay vs. Passthrough
Passthrough • Send tones inband• VAD, CNG disabled• ECAN is optionally disabled after
detection of phase reversal tone from the modems
• Very susceptible to packet loss, network delays
• Susceptible to clock skews (clock sync differences between gateways)
Relay• Demodulates/modulates signaling• Send tones in bearer or signaling
path in a special encoding format• Have built in redundancy to combat
network issues
Voice GW
Latency, Jitter, Buffers,
Packet loss
Voice GWFAX/Modem
Over IP
PSTN
PSTN
40© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 40© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Modem over IP Methods
• Modem passthrough: No negotiation and switchover in NSE
• Modem relay signal assisted
Negotiation in Protocol H.323, SIP and MGCP and switchover in NSE
• Modem relay gateway controlled
Negotiation and switchover in NSE
Capability Negotiation
Voice call Established
Switchover MR
ACK MR Switchover
AnsAM
CM Tone
Upspeed RequestedAcknowledge Upspeed
Passthrough Mode(VBD)
PhaseReversal
Ecan Off NSE
Ecan Off NSE ACK
Modem Passthrough
Originating Terminating
41© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 41© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Modem Relay Methods Comparison
• Introduced in 12.2(11)T• Capabilities are exchanged
during call setup
MGCP/SIP uses SDP for MR parameter negotiationH.323 uses H.245 for MRparameter negotiation
• Cisco NSE mechanism is used for media switching from voice to pass-through to relay
• Not Supported with CCM
Signal Assisted• Introduced in 12.4(4)T• MR gateway capabilities are
not exchanged duringcall setup
GW will use configured valuesIn case of configuration mismatch, fallback to modem passthrough
• Cisco NSE mechanism is used for media switching from voice to pass-through to relay
• Supported with CCM
Gateway Controlled
42© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 42© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Agenda
• Gateway Deployment Scenarios• Which GW Protocol? SIP, H.323 and MGCP• Gateway Availability Considerations• Supplementary Services with GWs• Fax/Modem Capabilities• Specialized Gateway Capabilities
Integrated Voice+Data AccessVideo SwitchingDrop and InsertChannel-BankClocking
• Gateway Capacity
43© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 43© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Integrated Voice + Data Access
North-America: Serial Data• Voice: ds0 or pri-group• Data: channel-group (HDLC, FR, PPP)• Timeslot allocation: static• H.323 and SIP only• Has always been supported
Europe/Aus: Pri Data• Voice + data: pri-group• Data: Dialer i/f using PRI channels• Timeslot allocation: dynamic• H.323 and SIP only• Supported as of 12.4.9T
T1
D-Channel (if pri-group is Used), Controls Voice Channels Only
Voice + Data (pri-group)T1/E1
D-Channel Controls All Channels
Voice channels:• Voice bearer CAPData channels:• 64K unrestricted
bearer CAP
Data (Channel-Group: FR, HDLC, PPP, MLPPP)
Voice (ds0, or pri-group)
Single T1/E1
Voice
Data Data
Voice
Single T1/E1
PSTN
WAN
44© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 44© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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PSTN
Integrated Voice + Data PRI
Single T1/E1 PRI
Voice Backup Data Voice
Single T1/E1 PRI
WAN DataData X
Dial Backup Via PSTN Connection During WAN Failure
Voice VoiceDataData
PRI WAN Connection Sharing Same Interface as Voice (SP Offering)
Integrated Service Provider
Data Services
Voice Services
Single T1/E1 PRI
Single T1/E1 PRI
45© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 45© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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IP
Terminating Voice + Data on a Router• Voice and video traffic is terminated on DSPs• Data traffic is terminated on HDLC controllers
Channel-group: need 1 HDLC controller per groupPRI: need 1 HDLC controller per channel (timeslot)
• Requires 12.4.9T minimum
VWIC
DSPs
HDLC Controllers
T1/E1
VoIP Packets
ISR Voice Gateway
TDM Backplane
Data PacketsTDM (Data)
TDM (Voice/Video)Cisco IOS
SW
46© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 46© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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CiscoIOS SW
ISDN Switching: Video• ISDN switching uses
Pri-groupsPOTS dial-peersH.323/SIP for voice termination
ISR Voice Gateway
TDM Backplane
EM-4BRI-NT/TE
NM-HDV2-1T1/E1
EVM
Onboard DSPs
NM-HDV2 DSPs
T1/E1
BRIVideo Media
Signaling
Voice Media
dial-peer voice 3000 voipdestination-pattern 688....session target ipv4:1.1.1.1dtmf-relay h245-alphanumeric
dial-peer voice 21 potsdestination-pattern 9Tport 1/0:15
dial-peer voice 20 potsdestination-pattern 9Tport 2/0/16
IP
For the Full Configuration of Video ISDN Switching, See http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00804794c6.shtml
47© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 47© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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DSP Dropping for Locally Switched Calls
1700 WIC
2800/3800WIC
NM-1V/2V
NM-HDA
NM-HDV
AIM-[ATM]-VOICE-30
NM-HD-1V/2V/2VE
NM-HDV2 EVM
1700 WIC No N/A N/A N/A N/A N/A N/A N/A
2800/3800 WIC Yes N/A No No N/A Yes Yes Yes
NM-1V/2V No No No No No No N/A
NM-HDA No No No No No No
NM-HDV No No No No No
AIM-[ATM-]VOICE-30 No No No N/A
NM-HD-1V/2V/2VE Yes* Yes* Yes
NM-HDV2 Yes* Yes
EVM Yes
*Supported on the 2800 and 3800 Series Routers
48© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 48© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Modem Relay Methods Comparison• D&I is a L1 cross-connect (HW
switch from designated ingress to egress ports)
tdm-groups“connect tdm” CLI
• Statically configured• Cisco IOS CPU/software does not
process the traffic or the signaling• Any kind of traffic can be carried
• Ingress and egress interface must be the same type
Cannot D&I T1/E1 to BRICannot D&I T1 to E1 or vice versaCannot D&I T1 SF-AMI to T1 ESF-B8ZS
ISR Router
TDM Backplane
NM-HDV2-1T1/E1
HWIC
T1/E1
T1/E1
controller T1 1/0/0framing esflinecoding b8zstdm-group 2 timeslots 13-24 type xxx
!controller T1 1/1/0
framing esflinecoding b8zstdm-group 3 timeslots 13-24 type xxxclock source line primary
!connect tdm1 T1 1/0/0 2 T1 1/1/0 3
Cisco IOS SW
VWIC
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Drop and Insert (D&I) vs. TDM SwitchingD&I TDM Switching
Configuration “Connect tdm” CLI POTS Dial-Peers
DSPs Required
Ingress and Egress Interface Are Independent
Traffic Types Not Interpreted—Statically Switched Based on Config
Interpreted—Both Interfaces Are Terminated on the Router
Signaling Types
Signaling Is Not Terminated—Any Traffic Can Be Switched
Signaling is Terminated—Only TDM Signaling Supported by GW Can Be
SwitchedSwitching Flexibility Static Dynamic, Destination Determined on a
Per Call Basis
#Interfaces One Ingress Mapped to One Egress Interface
Any Number of Interfaces Present on the GW
#Channels #Ingress channels = #Egress Channels
Any Number of Ingress/Egress Channels Can Be Switched
ISDN traffic All Channels on the PRIMust Be D&I’d
Individual Channels Can Be Switched to Different Destinations
DSPs No DSPs Needed
Interfaces Ingress and Egress Interface Must Be the Same Type
50© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 50© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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D&I (Cross-Connect) ConfigurationsVoice Gateway
HWIC SlotVWIC
EM-HDA-8FXSVoIPSW Two Analog Phones Cross-
Connected into the PSTN,no VoIP AccessAnalog Phone with VoIP Access
FXS Loopstart
T1 FXO Loopstart T1 FXS Loopstart
EVM-HD
T1 to PSTN with Three ds0-groups:• Two cross-connected FXS/FXO ports, one ds0 each• One ds0-group for PSTN-to-VoIP access-multiple ds0s
Ana
log-
to-D
igita
l Cro
ss-C
onne
ctD
igita
l-to-
Dig
ital C
ross
-Con
nect
EVM-HD-8FXS/DID with 24x FXS
Up to 24 Analog Phones
24-Channel T1VWIC
TDM
PSTN
51© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 51© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Clocking Domains on the ISR Voice Gateways
• Each domain can be clocked independently
• Each domain can only have one external clock if voice (DSP access) is present
• All domains sharing DSP access across the backplane must be sync’d
• Network clock select CLI designates the port that will provide clock to the backplane (there can be primary/secondary ports)
• Network clock participate designates ports deriving clocking from the backplane
• If network clocking is turned off, the domain is clocked independently from other domains
• Network clock participation is required for interfaces accessing motherboard DSPs
• Dual clock sources per domain are only supported when all but one interface are data (NM-HDV2 and VWIC2)
EVM
DSP
NM
HWIC
DSP
NMDSP
ASIC
Motherboard Domain
NM Domain
NM Domain
TDM Backplane
ISR Voice Gateway
HWIC
HWIC
HWIC
52© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 52© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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DSP
DSP
NM
Rx
PLL
Clock Source LineNo Network Clock
TxClock Looped
TDM Back-Plane
Rx
PLL
Clock Source InternalNo Network Clock
Tx
NM
Rx
Clock Source LineNetwork Clock Participate
TxClock Source from Another Interface Marked as “select”, or system PLL
Clock Source InternalNetwork Clock Participate
Rx
TxClock Source from Another Interface Marked as “select”, or system PLL
TDM Back-Plane
TDM Back-Plane
TDM Back-Plane
Rx
Tx
Clock Source to Other Interfaces
TDM Back-Plane
Port 0/0/0
PLL
Clock Source LineNetwork Clock Select T1 0/0/0Network Clock Participate
Clocking CLI Configurations
Note: “Network Clock” in These Examples Refers to the Cisco IOS CLI Commands:
• Network clock select—• Network clock participate
DSP
DSP DSP DSP
53© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 53© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Additional Notes on Clocking
• Clocking configurations must be correct:To ensure good voice quality—clock slips impair voice qualityFor correct fax and modem traffic handlingTDM switching and D&I operation
• VWIC cards support a single external clock sourceDual clocking is supported only if both ports are data
• All interfaces using motherboard DSPs must beclocked in sync
• Clocking settings must be correct for conferencing if a TDM endpoint is involved in the conference
If all endpoints are IP, clocking configuration on the GW is notimportant
• Carefully set up clocking for WAN and voice configurations
54© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 54© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Agenda
• Gateway Deployment Scenarios
• Which GW Protocol? SIP, H.323 and MGCP
• Gateway Availability Considerations
• Supplementary Services with GWs
• Fax/Modem Capabilities
• Specialized Gateway Capabilities
• Gateway Capacity
55© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 55© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Cisco Voice Gateway CapacityPhysical Channel (DS0) Connectivity
4802405403001201201201801500000NM/AIM-Based
24024000240240240000240120120Onboard
7204805403003603603601801500240120120E1 Channels Connectivity
3841924322409696961441200000NM/AIM-Based
192
576
16
8
24
64
32
32
24
56
88
3845
192
384
8
8
16
40
20
20
16
36
52
3825
0
432
18
18
32
16
16
16
32
48
3745
0
240
10
10
16
8
8
8
16
24
3725
192
288
4
8
12
40
20
20
12
36
52
2851
192
288
4
8
12
40
20
20
12
36
522821
192
288
4
8
12
24
12
12
12
24
28
2811
0
144
6
6
8
4
4
4
8
12
2691
0
120
5
5
8
4
4
4
8
12
2600XM
0
0
0
0
0
0
0
0
0
0
24
VG224
884Analog-DID
1929696Onboard
1929696T1 Channels Connectivity
000NM/AIM-Based T1/E1 Ports844Onboard T1/E1 Ports844Total T1/E1 Ports
161612BRI Channels886BRI Ports
886E&M16
16
2801
1612FXO/CAMA1612FXS
1760
1751
4802405403001201201201801500000NM/AIM-Based
24024000240240240000240120120Onboard
7204805403003603603601801500240120120E1 Channels Connectivity
3841924322409696961441200000NM/AIM-Based
192
576
16
8
24
64
32
32
24
56
88
3845
192
384
8
8
16
40
20
20
16
36
52
3825
0
432
18
18
32
16
16
16
32
48
3745
0
240
10
10
16
8
8
8
16
24
3725
192
288
4
8
12
40
20
20
12
36
52
2851
192
288
4
8
12
40
20
20
12
36
522821
192
288
4
8
12
24
12
12
12
24
28
2811
0
144
6
6
8
4
4
4
8
12
2691
0
120
5
5
8
4
4
4
8
12
2600XM
0
0
0
0
0
0
0
0
0
0
24
VG224
884Analog-DID
1929696Onboard
1929696T1 Channels Connectivity
000NM/AIM-Based T1/E1 Ports844Onboard T1/E1 Ports844Total T1/E1 Ports
161612BRI Channels886BRI Ports
886E&M16
16
2801
1612FXO/CAMA1612FXS
1760
1751
56© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 56© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Cisco Voice Gateway CapacityVoice Termination Channel Capacity: 12.4
32 38 5070
112
170130
180
290
340
450
18 20 3512 16 24
140
220
290330
100
150
884835
61
12082
102
170
250270
0
50100
150200
250300
350
400450
500550
600
2611XM 2621XM 2651XM 2811 2821 2851 2691 3725 3745 3825 3845
# C
alls
at 7
5% C
PU
FE/GE WAN No-cRTP cRTP
57© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 57© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Cisco Voice Gateway CapacityVXML Sessions: 12.4
Dedicated VXML Service Voice Gateway and VXML
Memory Recommended
Cisco 3725 68 50 50 38 512MBCisco 3745 100 80 77 60 512MB
Platform VXML and DTMF
VXML and ASR/TTS
VXML and DTMF
VXML and ASR/TTS
4203048
729672192192
Cisco 2821 48 36 36 256MBCisco 2851 60 56 56 512MB
AS5400HPX 96 90 90 DefaultAS5350XM 240 192 192 DefaultAS5400XM 240 192 192 Default
Cisco 2801 7 6 6 256MBCisco 2811 30 24 24 256MB
Cisco 3825 120 96 96 512MBCisco 3845 150 144 144 512MB
The Numbers Assume the Only Activities Running on the Gws Are VXML with Basic Routing and IP Connectivity. If Additional Applications Run on the GW, Such as Fax, Security, Normal Business Calls, Etc., Then the Capacity Should Be Prorated Accordingly. These Figures Apply to Cisco IOS 12.4.
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Session Capacity for RSVP Agent• License entitlement
Cisco Unified Survivable Remote Site Telephony: defined by the number of licensed sessions and router performanceCisco IP-to-IP Gateway: limited only by router performance
• Session capacity2611XM 40 sessions 2821 240 sessions2621XM 50 sessions 2851 300 sessions2651XM 65 sessions 3725 250 sessions2691 150 sessions 3745 320 sessions2801 130 sessions 3825 400 sessions2811 180 sessions 3845 536 sessions
• Session capacity is estimated assuming the router is dedicated to the RSVP agent and 75% CPU utilization; addition of concurrent applications will reduce the number of sessions supported
• 12.4.6T
59© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 59© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Router-Based Media Services
60© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 60© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Agenda
• Media Services Design, Operation and Configuration
• DSP Engineering, Allocation, Sharing
• Case Study
61© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 61© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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BusinessVoIP
CiscoCallManagerCluster
IP WAN
...
TranscodingDSPs
IVR
IP-Based Media Services Overview
• ConferencingMixing RTP streams formulti-party conference bridges
• TranscodingSupport multiple codecs onthe same call (e.g., G.711 to G.729)Changing codec neededin presence of single-codec endpoints and applications
• MTPAnchor an RTP stream to change headers or change treatmentof the call
• SBC (session border controller, or IP-IP GW)
Network boundary demarcation (security or billing)Protocol translation (e.g. H.323 to SIP)
Xcod
Conf
PSTN SBC
ConferencingDSPs
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IP WAN
• External caller Xcalls A—no voiceacross WAN
• A conferences B• Three voice streams
across WAN to central mixing
• Conference betweenA, B and X—no voiceacross WAN
• Utilizes DSPs inthe branch router for local mixing
Conference Media ResourcesHQ
Branch
AB
X
DevicePool
DevicePool
AB
X
MRG=HQ1 MRG=HQ2
MRGL
MRG=Br1
Conf Conf
Conf
Conf
Conf
Conf
Branch
HQ
MRG = Media Resource GroupMRGL = Media Resource Group List
MRGL
PSTN
PSTN
IP WAN
1. HQ12. HQ2
1. Br12. HQ13. HQ2
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When Necessary
• Offload central conference bridge software servers• To keep conferences, between participants at a remote site, from
crossing bandwidth-constrained (WAN) links
Where Located • Conf participants at same site—local• Conf participants at different sites—any site
Selection Algorithm
• Based on the location of the initiator of the conference• Conference resources registered with CCM• MRG/MRGL configuration order and device pools
Codecs
• Single-mode: G.711 only conferences• Mixed-mode: G.711 and G.729A participants• Codec chosen for a call leg between an endpoint and a conference
bridge is determined by the regions• CCM will engage transcoder for codec mismatches
Other Considerations
• Densities vary: router model, CPU and DSP vintage• Conference requires dedicated DSPs• No HW conference in SRST (only three-party SW conference)• Ad-hoc vs. Meetme traffic patterns are different• Two-party conf maintains DSP for MeetMe; releases for Ad-hoc
Conferencing Operation and Design
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VoIPWAN
Transcoding Media Services
• Call from HQ to branch remains G.729A when is CFNA/CFB to local branch-basedvoice mail thatrequires G.711
• Utilizes DSPs inthe branch router
• Similar scenario for CME network—maintains G.729A across the WAN
Branch HQ
Xcod
G.729
G.711
G.711
VMail
CME Site B
G.729
CME Site A
VMail
VoIPWAN
PSTN
Xcod
PSTN
65© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 65© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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When Necessary
• Bridge two call legs that use different codecs• To maintain G.729A WAN bandwidth for G.711—only
applications• To enable calls between endpoints with no common
codec supportWhere located • Collocated with the G.711 endpointSelection Algorithm
• Transcoding resources registered with CCM• MRG/MRGL configuration order and device pools
Codecs
• G.711 a/µ-law to/from G.729, G.729A, G.729B, G.729AB• G.711 a-law to/from G.711 µ-law • Different packetizations
G.711: 10ms, 20ms and 30msG.729: 10ms, 20ms, 30ms, 40ms, 50ms and 60ms
Other Considerations
• Densities vary: router model, CPU and DSP vintage• Transcoding shares DSP with voice termination• RFC2833 DTMF detection; but no pass-through yet• Transcoding supported with CCM MRG, as well as IP-IP GW
Transcoding Operation and Design
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IPWAN
IPWAN
Branch HQ
Xcod
VMail
MTP Media Services
• An MTP anchors the RTP stream
• Provides supplementary services for devices that cannot support ECS
• Provides a singleIP address for allendpoints at the site to an outside network connection
Branch HQ
MTP
H.323 Video Device
BusinessVoIP
MTP
PSTN
PSTN
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When Necessary
1)Security demarcation—IP address hiding2) RSVP proxy for CAC3) Anchor point for supplementary services4) CCM 4.0 SIP trunks (no longer required with CCM 5.0)5) DTMF translation
Where Located • For #1–2: collocated at the site• For #3–5: anywhere
Selection Algorithm
• MTP resources registered with CCM• MRG/MRGL configuration order and device pools
Codecs• SW-MTP: same codec, same packetization• HW-MTP: same codec, different packetization• (Transcoding: different codec, same/different packetization)
Other Considerations
• Densities vary: router model, CPU and DSP vintage• SW MTP requires no DSPs; HW MTP does• RSVP-Agent is a special type of SW-MTP• IP-IP GW is an implicit MTP (but not controlled by CCM)
MTP Operation and Design
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Session Border Controller (IP-IP Gateway)
• SBC (IP-to-IP Gateway) can be used forNetwork Privacy/Security DemarcationProtocol InterworkingDTMF InterworkingTranscoding G.711 to G.729Quality of Service – Packet MarkingCall Admission Control – RSVPCall Detail RecordsClass Of RestrictionPoint of Demarcation for TroubleshootingIntegrated TDM functionalityIntegrated Gatekeeper functionality
PBX PBX
SBC
More Information:VVT-2015: Interconnection of Voice and Video Networks using the Cisco Multiservice IP-to-IP Gateway
VoIP SP
PSTN
PSTN
WAN
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DSP Media Services Configuration Example
voice-card 1dsp services dspfarm
!sccp local FastEthernet0/0sccp ccm 10.4.20.24 identifier 1 version 4.0sccp ccm 10.4.20.25 identifier 2 version 4.0sccp ccm 10.4.20.26 identifier 3 version 4.0sccp ip precedence 3sccp!sccp ccm group 988associate ccm 1 priority 1associate ccm 2 priority 2associate ccm 3 priority 3associate profile 10 register CFB123456789966associate profile 6 register MTP123456789988keepalive retries 5switchover method immediateswitchback method immediateswitchback interval 15…
IP Address Used on Packets to CCM Taken From This Interface
Definition of the CCMs the Router Registers with
Priority of CCMs for Failover of Registration
Failover Timers and Behavior
Router Capabilities (Profiles) Registered with CCM
Define the DSPs Used for Media Resources
Multiple [groups of] CCMs can be defined
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DSP Media Services Configuration Example
…dspfarm profile 6 transcodecodec g711ulawcodec g711alawcodec g729ar8codec g729abr8maximum sessions 10associate application SCCP!!!dspfarm profile 10 conferencecodec g711ulawcodec g711alawcodec g729ar8codec g729abr8codec g729r8codec g729br8maximum sessions 6associate application SCCP
Transcoding Profile Definition
Codec Capabilities of this Transcoding Profile
Max Number of Simultaneous Transcoding Sessions (Calls)—Determines How Many DSPs Are Used/Needed
Conference Profile Definition
Codec Capabilities of this Conference Profile By Default All Codecs Are Inserted
Max Number of Simultaneous Conferences (Not Participants)—Determines How Many DSPs Are Used/Needed
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IP
Xcod
Call Agents, Registration, Failover
Conf
1
2
3
sccp local FastEthernet0/0sccp ccm 10.4.20.24 identifier 1 version 4.0sccp ccm 10.4.20.25 identifier 2 version 4.0sccp ccm 10.4.20.26 identifier 3 version 4.0!sccp ccm group 988
associate ccm 1 priority 1associate ccm 2 priority 2associate ccm 3 priority 3…keepalive retries 5switchover method immediateswitchback method immediateswitchback interval 15
CCM
XcodConf
CME, SBC
Xcod
Conf
SRST
XConference, Transcoding and MTP Resources Are Not yet Available During SRST Mode
telephony-serviceip source-address 192.168.1.1 port 2000sdspfarm units 1sdspfarm transcode sessions 16sdspfarm tag 1 MTP000f23cd6100
!sccp local Vlan10sccp ccm 192.168.1.1 identifier 1sccp!sccp ccm group 1
associate ccm 1 priority 1…
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IP
Media Resources on a Single Router with Multiple CCMs
• Each CCM must have aseparate profile
• Each CCM below can do 48 sessions of MTP on this router
sccp ccm x.x.x.x identifier 1 version 4.1 sccp ccm y.y.y.y identifier 2 version 4.1 dspfarm profile 1 mtp
maximum sessions software 48 associate app sccp
dspfarm profile 2 mtp maximum sessions software 48 associate app sccp
sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register MTP123456789001
sccp ccm group 2 associate ccm 2 priority 1 associate profile 2 register MTP123456789002
CCM1: x.x.x.x
CCM2: y.y.y.y
Associate Each Profile with One of the CCMs
Define One or More Profiles for Each CCM
Define the CCMs
Share Router Resources Among Multiple CCMs
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sccp local FastEthernet0/0sccp ccm 10.9.20.24 identifier 1 version 4.0!sccp ccm group 1
associate ccm 1 priority 1 bind interface FastEthernet0/0
Controlling IP Addresses
interface Loopback1 ip address 10.1.1.3 255.255.255.255
! interface FastEthernet0/0
ip address 10.9.68.100 255.255.255.0 ! sccp local Loopback1sccp ccm 10.9.64.138 identifier 10 version 4.1
interface FastEthernet0/0 ip address 10.9.68.100 255.255.255.0
! sccp local FastEthernet0/0sccp ccm 10.9.20.24 identifier 1 version 4.0
• Typical configuration: associate CCM with a router interface
• If the interface is down, media resources are unusable regardless of IP reachability
• Associate CCM with a loopback interface
• This is always up and media resources remain available as long as a routing path is available (independent of the status of any specific interface)
• The “sccp local” command controls the IP address on signaling messages—global
• The “bind interface” commands controls the IP address on media packets—per CCM group
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WAN
WAN
SPVoIP
• Transcoder control is in CCM—based on regions and endpoint capabilities
• CCM is aware of both legs of the call
• DSPs required on router
• Transcoder control is in IP-IP GW—based on mismatched codecs on ingress/egress call legs
• CCM is aware of only one leg of the call (single-codec call from CCM’s point of view)
• DSP required on IP-IP GW
Using the IP-IP GW as a Transcoder
IP-to-IP
Xcod
Xcod
G.711
G.711
G.729
G.729
CCM Transcoding
IP-IP GW Transcoding
PSTN
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Agenda
• Media Services Design, Operation and Configuration
• DSP Engineering, Allocation, Sharing
• Case Study
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How Many DSPs Do I Require?PVDM2 DSP Basic Rules
• Each DSP is treated individually—PVDM2 boundaries are irrelevant
• ConferencingRequires a dedicated DSPUp to eight participants per conferenceSingle-mode: all participants are G.711 a-law/µ-law; eight* fixed conferences (up to 64 participants) per DSP—static configurationMixed-mode: at least one participant is G.729A; two* fixed conferences(up to 16 participants) per DSP—static configurationConferences cannot span multiple DSPs
• Voice Termination, Transcoding, MTPThese functions can share a DSPG.711 (low complexity): 16 sessions (calls, xcoding or MTP sessions)G.729A (medium complexity): eight sessions (calls, xcoding or MTP sessions)G.729 (high complexity): six sessions (calls, xcoding or MTP sessions)
*Note: The Max Number of Conferences Per DSP Is Fixed, Regardless of How Many Participants There Are per Conference
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http://www.cisco.com/cgi-bin/Support/DSP/cisco_adv.pl
Configure RouterCards and Voice
Termination channels
Configure OptionsConf, Xcod, MTP, Analog
Reservation, IP SLA
ResultsDSP Cards and # DSPs
How Many DSPs Do I Require?DSP Calculator Tool
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Where Should the DSPs Be Located?DSP Architecture on the ISRs
• Two, three, four onboard PVDM2 slots (depending on router platform); four PVDM2 slots on each NM-HDV2
• HWIC/EVM interfaces slots use onboard DSPs• NM interfaces use NM DSPs• Analog/BRI can only use DSPs local DSPs• T1/E1 interfaces can optionally share DSPs from
another domainSearch order: local domain first, then remote domains starting with slot 0
• Conf/Xcod/MTP can use DSPs from any domain
EVM
NM-HDV2DSP
NM-HDV2
Motherboard Domain
NM Domain
NM Domain
ASIC
Additional NMs
Name #DSPs G.711 Channels
G.729A Channels
G729 Channels
“Half” 46121824
1234
48
162432
PVDM2-8 8PVDM2-16 16PVDM2-32 32PVDM2-48 48PVDM2-64 64
PVDM2s
DSP
DSP
HWICHWIC
HWICHWIC
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PVDM2 DSP Allocation—Example
PVDM2-64 (4 DSPs) Conferencing: Two DSPs• Up to 16 G.711 Conferences (16*8 =
128 participants), or• Up to four G.711/G.729A
Conferences (4*8 = 32 participants)
Voice Termination, Transcoding, MTP: Four DSPs• Flex complexity(FC): up to 64 (4*16) G.711-only
sessions, or between 24–64 mixed codec sessions
• Med complexity (MC): up to 32 (4*8) sessions• High complexity (HC): up to 24 (4*6) sessions
PVDM2-32 (2 DSPs)
• No CLI/manual control over individual DSP allocation• Conf/Xcod DSP availability checked at configuration time• Voice termination:
Signaling DSPs assigned/checked at configuration timeMedia DSPs assigned at runtime (oversubscription)
• Order of DSP allocation may be different after router reboot
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How Are the DSPs Allocated to Services?DSP Channel Allocation Algorithm
123456789
101112
G.711G.729A
G.729AG.711
G.729A
Conf One
Conf Two
Conf Three
Conf Four
Conf Five
Conf Six
Conf Seven
Conf Eight
Conf One
Conf Two
Voice Termination, Xcoding, MTP G.711 Only Conference
G.711/G.729A
Up to Sixteen Signaling Channels Assigned per DSP
Media Channels—#Channels Depends on Combination of Codecs: Six to Sixteen Channels per DSP
Up to Eight Participants
Up to Eight Participants
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How Are the DSPs Shared?DSP Access with DSP Sharing
• DSP sharing pools together all the PVDM2 DSPs present inthe chassis
• DSP sharing is done only for T1/E1 digital ports, not for analog or BRI
• Default is no sharing• Recommendation
Set codec complexity to be the same on all cards that share DSPsTurn on network clocking for all cards that share DSPs
Cisco 2800/3800 RouterTDM
BackplaneVWIC
NM-HDV2-1T1/E1
NM-HDV
DSP Access
VIC
HWIC
Onboard DSP Slots
EM
EVM-HD
PVDM-12s
PVDM2s
PVDM2s
FXS, FXO, E&M, BRI
T1/E1
T1/E1
FXS/DID
FXS, FXO or BRI
T1/E1
VWIC
VIC
HWIC
FXS, FXO, E&M, BRI
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PVDM2 DSP CodecsComplexity and Channel Support
CodecChannels
Per DSP in High Compl.
(HC)
Channels Per DSP in
Med Compl. (MC)
Channels Per DSP in
Flex Compl. (FC)
G.711 (µ-law, a-law) 6 8 16Fax/Modem Passthrough 6 8 16
Clear-Channel Codec 6 8 16G.726 (32K, 24K, 16K) 6 8 8
G.728 6 Not Supported 6
Fax Relay 6 8 8G.729A, G.729AB 6 8 8G.729, G.729B, G.728 6 Not Supported 6
G.723.1 (5.3K, 6.3K), G.723.1A (5.3K, 6.3K) 6 Not Supported 6
Modem Relay 6 Not Supported 6
Low Complexity Codecs
Medium Complexity Codecs
High Complexity Codecs
Reference
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PVDM2 DSP Capacity Summary
Onboard Domain NM Domain
Application 2801/2811(1–8 DSPs)
2821/2851(1–12 DSPs)
3825/45(1–16 DSPs)
NM-HD-2VE (3 DSPs)
NM-HDV2(1–16 DSPs)
256
128
96
128
96
50 conf*(400 parties)
48
32 conf(256 parties)
24
256
128
96
128
96
50 conf*(400 parties)
18
24
18
24 conf(192 parties)
32 conf(256 parties)
6 conf (48 parties)
192
96
72
96
72
50 conf*(400 parties)
24 conf(192 parties)
128
64
48
64
48
50 conf*(400 parties)
16 conf(128 parties)
G.711a/µlaw
G.729A (MC Codecs)
Voice Termi-nation
G.729 (HC Codecs)
G.711a/µlaw ↔G.729A /G.729AB
Trans-coding G.711a/µlaw ↔
G.729 /G.729B
G.711 Conference(Single-Mode)
Con-ference G.729 Conference
(Mixed Mode)
Reference
*The Maximum Number of Conference Sessions Are Limited to 50 (with 400 Participants by HW/IO
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Summary: DSPs, Platforms, Interfaces
Voice Channels Per DSPPlatform/Interface DSP #Fixed
DSPs#Expandable
DSPs HC MC FCXCod/Conf
6 Yes
Yes
Yes
Yes
NoNoYes
AIM-VOICE-30, AIM-ATM-VOICE-30 TI-5421 4 4 8 No
YesYesYes
6-16
6-16
6-16
6-166-166-16
4
8
8
8
84
888
2
6
6
6
242
666
TI-549
TI-5510
TI-5510
TI-5510
TI-542TI-5421TI-549
TI-5510TI-5510TI-5510
1751/1760 Onboard 102801, 2811 (Onboard PVDM2) 8
NM-HD-1V/2V 1
2821, 2851 (Onboard PVDM2) 12
3825, 3845 (Onboard PVDM2) 16
NM-1V/2V 1NM-HDA 2 2NM-HDV 15
NM-HD-2VE 3NM-HDV2 (PVDM2) 16
Reference
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Agenda
• Media Services Design, Operation and Configuration
• DSP Engineering, Allocation, Sharing
• Case Study
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Fax Machines
Branch Office
Campus
Users
FXOPRI
Branch Office Requirements
• Centralized CCM at another site• Use G.711 within the site, and G.729A to
other sites• Branch site: 50 users
12 PSTN channels—fractional PRI66% terminate on local phones (G.711)33% of calls terminate at another site (G.729A),perhaps after a transfer
Four FXO lines for backupSix fax machinesSix to ten people in two conferences at any one timeFive transcoding channels for calls from other sites into local voicemail
CUE VoiceMail
Cisco 2821 Router with
SRST
WAN
PSTN
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Router Choice
• A 2801 router is sufficient purely for the voice needs, but all its slots would be populated (i.e. no room for expansion)
• To ensure capacity for the data and security needs of the office, at least a 2811 is required
• To optimize slot use of the FXO and FXS ports, and to leave HWIC and NM slots open for data applications and future growth needs, an EVM-HD is required, and therefore a 2821 router is selected
HWIC–Empty HWIC–Empty
FXOT1 PRIT1
6x Fax Machines
WANPSTN
NM–Data
EVM-HD-8FXS/DIDEM-HDA-6FXOVWIC2-1MFT-T1/E1VWIC2-1MFT-T1/E1
2821
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How Many DSPs Do We need?
• Four total DSPsThree DSPs for voice channels and xcodingOne DSP for conference
• PVDM2-64• Installed on router motherboard• CISCO2821-SRST/K9 (includes a
PVDM2-32)
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Router Configuration for Voice
voice-card 0dsp services dspfarm
!controller T1 1/0/0framing esflinecode b8zspri-group timeslots 1-12,24!interface Loopback0ip address 10.9.100.1 255.255.255.255
!interface Serial0/0/0ip address 10.9.101.6 255.255.255.252clock rate 2000000!interface Serial1/0/0:23no ip addressencapsulation hdlcisdn switch-type primary-niisdn incoming-voice voice
!voice-port 1/0/0:23
PSTN Fractional PRI with 12 Channels
Loopback Interface Used to Maintain CCM Connectivity
WAN Connectivity
D-Channel for the Voice PRI
Configure the DSPs for Use as Media Resources
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Router Configuration for Voice (Cont.)sccp local Loopback0sccp ccm 10.9.64.108 identifier 3 version 5.0.1 sccp!sccp ccm group 2
bind interface Loopback0associate ccm 3 priority 1associate profile 1 register XCODE123456associate profile 2 register CONF12345
!dspfarm profile 1 conferencecodec g711ulawcodec g711alawcodec g729ar8codec g729abr8codec g729r8codec g729br8maximum sessions 2associate application SCCP
!dspfarm profile 2 transcodecodec g711ulawcodec g711alawcodec g729ar8codec g729abr8maximum sessions 5associate application SCCP
Definition of the CCM the Router Registers with, Associate the Conf and Xcod Profiles with This CCM; Define the Device Names Used in the Registration
Define Two Conferences of Mixed Codecs
Define up to Five Transcoding Sessions
IP Address Used on Packets to CCM Taken from This Interface
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Router-Based Applications
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Agenda
• Secure Voice
• RSVP Agent
• IP Service Level Agreements
• Circuit Emulation over IP
• Network Analysis
• Cisco CallManager Express/Cisco Unity® Express
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IP
• Phone-to-phone and phone-to-GW calls protected
MGCP and H.323 GWs
• SRTP protects end-to-end media
• TLS or IPSec protects end-to-end signaling
• H.323 encryption supported for CCM and toll bypass configurations
Voice Encryption: SRTP
Site A
Central Site
Site B
SRTP (Media)TLS or IPSec (Signaling)
PSTN
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Location BLocation A
RSVP Agent
• CCM inserts a pair of RSVP agents in the media path whenever RSVP is needed (based on the CCM locations configuration)
• RSVP agent creates RSVP paths (reservations) on behalf of the endpoints between locations
WAN
HQ
RSVPRTP
Media Resource Control
SCCP SCCP
RSVP Agent RSVP Agent
RTP
Media Resource Control
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San Diego 1Headquarters
SeattleSales Office
L.A.Sales Office
San JoseSales Office
ChicagoRegional Office
New YorkSales Office
BostonSales Office
San Diego 2Data Center
CiscoCallManager
Cluster
Cleveland Detroit
SAA End-to-End Measurements• Edge-to-edge• Hop-by-hop• Edge-to-server
Responder
SAA
IP SLA: Measurements with SAA
SAASAA
SAA
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Radar
Encryption
SCADA
Telemetry
H.320 Video
OtherApplications
Applications that Require Bit-Transparent Circuits Can’t Be Natively Integrated into a Packet Network, but They Can Be “Tunneled” Across the IP Network Using Circuit Emulation
Cell Site Backhaul
T1/E1 LeasedLine Emulation
VoIPIP Telephony
IP, ATM, FR— Data Data
Circuit Emulation over IP
CPE
TDM TDM
CPE
IP
CESovIP CESovIP
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Traffic Monitoring (NM-NAM)
• Web-based traffic analyzer GUI• Analyzes traffic-flows for applications,
hosts, conversations, and QoS/VoIP services
• Collects NetFlow Data Export to provide application-level visibility
• Tracks response times to isolate application performance
Security
IDS
IP L2/L3
AAA
Firewall IDS
Video Surv.
Operations
Content
6K-NAM
NM-NAM
NM-NAM Available for Cisco 2800/3800 Series Routers
6K-NAM Available for Cisco Catalyst 6500 Switches and Cisco 7600 Series Routers
NetFlow Data Export to 6K-NAM
Monitoring Remote Sites Through Web Based Traffic Analyzer
IP
6K-NAM
6K-NAM
NM-NAM
NM-NAM
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Cisco Unified CallManager Express and Cisco Unity Express
Cisco Survivable Remote Site Telephony and Cisco Unity Express
Redundant Call Processing Localized or Central MessagingNetworked
Branch A
Branch B
Applications(UM, IVR, ICD—)
Central Site
Cisco CallManagerCluster
Centralized Call Processing
Cisco CallManagerExpress and Cisco Unity Express
Localized Call Processing Localized Messaging
Fat Pipe
Small Pipe “Loosely Coupled”
IP
PSTN
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Q and A
100© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 100© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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In Conclusion:
• The Cisco voice gateway platforms offer a wide variety of TDM and IP-based voice services in the network infrastructure
• The protocol used to a Cisco voice gateway can be H.323, SIP or MGCP (or SCCP for FXS)
The selection of the appropriate protocol for a particular deployment is a key network design decision
• The use of media resources should be carefully considered—placing them in the right quantities and locations directly affect the optimization of the network design
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Related Sessions
• RST-2454: Cisco ISR Architecture• CRT-2203: GWGK Exam Preparation—Implementing
Gateways• CRT-2204: GWGK Exam Preparation—Implementing
Gatekeepers and IP-to-IP Gateways• TEC-VVT1: Enterprise IP Telephony Design and
Deployment• TEC-VVT2: Session Initiation Protocol• VVT-1001: Intro to IP Telephony or VoIP for the Enterprise• VVT-2000: Intermediate Voice and Video Control
Protocols: H.323• VVT-2008: Understanding CallManager Dial Plan
Functionality
102© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 102© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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Related Sessions
• VVT-2015: Interconnection of Voice and Video Networks using the Cisco Multiservice IP-to-IP Gateway
• VVT-2101: Designing and Deploying IP-Based Audio and Web Conferencing Solutions
• VVT-2105: Call Admission Control Design for the Enterprise Wide Area Network
• VVT-2014: Designing CallManager Express andUnity Express Network Architecture
• VVT-2106: Deploying CallManager Express andUnity Express: Advanced Deployment Scenarios, Management and Security
103© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 103© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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References—General
• DSP Calculatorhttp://www.cisco.com/cgi-bin/Support/DSP/cisco_dsp_calc.pl
• Integrated Services Router General Informationhttp://www.cisco.com/go/isr
• Cisco Voice Gateway Router Interoperability with CallManagerhttp://www.cisco.com/en/US/products/ps5855/products_data_sheet09186a0080182d38.html
Gateway Channel Density (Table 5)
• Router Conferencing and Transcoding Density (table 2)http://www.cisco.com/en/US/products/ps5855/products_data_sheet0900aecd801b97a6.html
• Media Resources (chapter in the IP Tel SRND)http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guide_chapter09186a008044750d.html
104© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-201012625_04_2006_c2 104© 2006 Cisco Systems, Inc. All rights reserved. Cisco PublicVVT-2010
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References—Cisco IOS Voice Features
• Cisco IOS Voice Configuration Libraryhttp://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124tcg/vcl.htm
• Cisco IOS H.323 Configuration Guidehttp://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_book09186a00801fcee1.html
• Cisco IOS SIP Configuration Guidehttp://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipc1_c/index.htm
• Cisco IOS MGCP and Related Protocols Configuration Guidehttp://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_book09186a008020bd29.html
• Cisco IOS IP SLAs Configuration Guidehttp://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124cg/hsla_c/index.htm
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Recommended Reading
• Continue your Cisco Networkers learning experience withfurther reading from Cisco Press
• Check the Recommended Reading flyerfor suggested books
Cisco Voice Gateways and Gatekeepers[1-58705-258-X]—available August 2006Cisco CallManager Fundamentals, 2nd Ed.[1-58705-192-3]Voice over IP Fundamentals, 2nd Ed.[1-58705-257-1]Cisco IP Communications Express:CallManager Express with Cisco Unity Express[1-58705-180-X]
Available on Site at the Cisco Company Store
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Complete Your Online Session Evaluation
• Win fabulous prizes; Give us yourfeedback
• Receive ten Passport Points for each session evaluation you complete
• Go to the Internet stations located throughout the Convention Center tocomplete your session evaluation
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