Design Voice GW

Designing Voice Gateways and Media Resources in Enterprise Networks
VVT-2010
VVT-2010 12625_04_2006_c2
2006 Cisco Systems, Inc. All rights reserved.
Cisco Public
Agenda
GW Platforms and Voice Interface Capabilities GW Design: Features, Protocols and Operation Router-Based Media Services Router-Based Applications
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2006 Cisco Systems, All rights reserved. 2006 Cisco Systems, Inc. Inc. All rights reserved.
Cisco Public Cisco Public
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Scope of This Seminar

Cisco Applications PBX CallManager
PBX
Applications
PSTN
IP WAN
Router/GW Router/GW
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 network
DSP/media services, RSVP, SRST, CME, voice mail
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Platforms and Voice Interface Capabilities
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Cisco Voice Gateways

RouterBased GWs Dedicated GWs SwitchBased GWs Typical Deployment
Enterprise Campus and Beyond Large Branch
Performance and Services
Cisco 3800 Series

Cisco 3700 Series
Cisco 5350XM, 5400XM, 5850
Cisco Catalyst 6500 CMM
Cisco 2800, 2600XM, 2691 Series Cisco VG248
Enterprise Branch SMB/Small Branch Teleworker/SOHO

Cisco Public Cisco Public 5 5
Cisco 2801 1751, 1760
Cisco VG224 Cisco ATA 186
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Voice Gateway Selection Criteria

Platforms Service Provider GWs Enterprise Campus GWs Enterprise Branch Office GWs Analog GWs End User
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Typical Interfaces
T1/E1, T3, STM-1
Common Deployments
SP Networks, Softswitches, GW-to-GW, PSTN GW CCM, PSTN/PBX GW CCM, Enterprise GW-toGW, PSTN/PBX/Key GW CCM Teleworker
Feature Set Focus

Service Provider Voice Enterprise Voice Enterprise Integrated Voice+Data Phone Set Features Phone Set Features
Cisco 5350XM, 5400XM, 5800 Cisco Catalyst 6500 CMM, Cisco 3800 Series Cisco 2600, 3700, 2800, 3800 Series VG224/248 ATA186
T1/E1
Analog, BRI, T1/E1 FXS FXS
Gateway Signaling Protocol Summary

Analog TDM
FXS FXO E&M: wink, immediate, delay dial Analog DID CAMA
Digital TDM
BRI: Q.931, QSIG T1: CAS, FGD, PRI, QSIG, PRI NFAS E1: PRI, QSIG, R2, CAS/MELCAS J1 T3/STM-1
VoIP
SIP H.323 MGCP 1.0: SP call agents MGCP: 0.1+: CCM
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
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GW Interface Card Port Densities

Cisco 2600, 3700, 2800, 3800 Series
NM-HDV*
EVM-HD
VWIC, VWIC2 2
VIC* / VIC2
NM-HD-1V
Station-Side FXS Trunk-Side FXO BRI T1/E1

Note: EOS/EOL For Some of These Cards Have Been Announced
12
24
NM-HD-2VE
NM-HD-2V
4 2
4 2
8 4
12 8
8 4 1 2
NM-HDV2 4 4 2 4
8 8
NM-HDA
Analog/BRI Voice Interface Card (VIC)

Considerations
VIC
Minimum Release Platform Support NM Support Port Density FXO FXS/DID FXO/CAMA
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VIC2
(Including VIC-4FXS/DID and VIC-2DID)
12.0T Cisco VG200, 1700, 2600, 3600, 3700 NM-1V/2V Two-Port Separate FXO cards For Different Geographies Separate FXS and DID Cards (Except the VIC4FXS/DID) Separate FXO and CAMA Cards
12.3T Cisco 1700, 2600XM, 3700, 2800, 3800 NM-HD-1V/2V/2VE, NMHDV2 Two- and Four-Port cards Single FXO Card for All Geographies SW-Configurable Ports on the FXS/DID and DID Cards SW-Configurable Ports on the FXO Cards
T1/E1 Voice/WAN Interface Card (VWIC)

Considerations
VWIC
Minimum Release Platform Support NM Support Drop and Insert Clocking T1/E1 Optional HW ECAN
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VWIC2
12.4
11.3
Cisco VG200, 1700, 2600, Cisco 1700, 2600XM, 3600, 3700, 2800, 3800 3700, 2800, 3800 NM-HDV, NM-HD-2VE, NM-HDV, NM-HD-2VE, NM-HDV2 NM-HDV2 Only Supported Supported on All Cards onDI Cards Single External Clock Dual External Clocks (For Per Card Data Ports) Per Card Separate Cards for T1 SW-Configurable and E1 Operation T1/E1 Ports No Yes
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 DID Plug in zero, one or two expansion modulesin any combination
EM 0
EM 1
Expansion modules:
EM-HDA-8FXS (shared with NM-HDA) EM-4BRI-NT/TE EM-HDA-3FXS/4FXO EM-HDA-6FXO
Use router motherboard DSPs Supported on the 2821, 2851, 3825 and 3845
Not supported on the 2801 or 2811 Max one EVM on the 3825 and max two on the 3845
RJ21 Connector
11 11
EVM-HD Expansion Module Combinations

Combine Individual Components in Any Order
Total Ports Base Board 8 FXS or DID
EM-HDA-8FXS EM-HDA-8FXS EM-HDA-8FXS EM-HDA-8FXS EM-HDA-8FXS EM-HDA-8FXS EM-HDA-3FXS/4FXO EM-HDA-6FXO EM-4BRI-NT/TE EM-HDA-3FXS/4FXO EM-HDA-6FXO EM-4BRI-NT/TE EM-HDA-6FXO EM-4BRI-NT/TE EM-4BRI-NT/TE
EM 0
EM 1
FXS or DID 8 8 8 8 8 8 8 8 8 8 8 8 8 8 8
BRI FXS FXO Ports B-Ch 8 8 16 11 8 8 3 6 3 3 4 8 10 4 6 12 6 4 4 8 8 8 16 4 8 4 6 4 8 16 24 23 22 24 15 22 21 23 14 20 22 16 24

12 12
Module
EVM-HD8FXS/DID
EM-HDA-3FXS/4FXO EM-HDA-3FXS/4FXO EM-HDA-3FXS/4FXO EM-HDA-3FXS/4FXO EM-HDA-6FXO EM-HDA-6FXO EM-HDA-6FXO EM-4BRI-NT/TE EM-4BRI-NT/TE
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DSP Architecture of the ISRs

Onboard DSP Slots (2, 3 or 4), Accessed by the EVM and Voice Cards in the HWIC Slots
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
Power + 802.3af
VPN
AIM
AIM
USB USB
NM
HWIC HWIC HWIC GE GE
NM
HWIC
SFP
NM-HDV2s can be used to expand ISR DSP capacity DSPs can be sharedto some extentbetween motherboard and NM-based interfaces
Platform Onboard PVDM2 slots 2 2 3 4 4 NM-Based PVDM2 slots 0 4 4 8 16
NM-HDV2
Each NM-HDV2 Provides Four More DSP Slots

2801 2811 2821/51 3825 3845
IP Phone Inline Power Support

AC or AC+IP power options, e.g.
CISCO2811no IP power CISCO2811-AC-IPhas IP power
HWIC-4ESW-POE HWIC-D-9ESW-POE
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 Chassis
2801 2811 2821 2851 3825 3845
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NM-16ESW-PWR
NMD-36-ESW-PWR
NME-16ES-1GE NME-X-24ES-1GE NME-XD-24ES-2ST NME-XD-48ES-2GE Max # POE Switchports

16 24 32 56 72 96
Cisco POE Support

HWIC HWIC/NM HWIC/NM HWIC/NM HWIC/NM HWIC/NM
802.3af POE Support

HWIC HWIC HWIC HWIC HWIC HWIC
Power (W)
120 160 240 360 360 360
Max # IP Phones (7W)

16 22 32 51 51 51
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

Campus
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
PBX
H.323 networks
Small meshed networks Large network deployed with gatekeepers
GK
IP
PSTN
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
H.323
CCM Network Deployments

CCM Cluster 1 Centralized or distributed CCM scenarios ICT: H.323 or SIP GWs: H.323, SIP or MGCP Dialplan is primarily centralized
GW dial-peers point to CCM for call routing GWs have backup dial-peers for call routing when CCM is unavailable
CCM Cluster 2
GK
ICT
PSTN
IP
GW platforms often integrate other IP-based voice services

SRST, conferencing, transcoding, RSVP CAC
H.323 or SIP MGCP, SIP or H.323

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Branch 1 Cluster 1
Branch 2 Cluster 1
Branch 1 Cluster 2
Branch 2 Cluster 2
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CCM and CME Network Deployments

CCM Cluster One or more CCM and CME sites CCM and CME
ICT or H.323/SIP trunks
CCM GWs
H.323, SIP or MGCP ICT
CME GWs
H.323 or SIP
PSTN
IP
GK
Dialplan is distributed
CCM and CME does call routing Optional GK may help with overall intersite call routing
H.323, SIP MGCP, SIP, or H.323

SCCP or SIP
SRST CME Branch 1 CME Branch 2 Branch 1
SRST Branch 2
Distributed Contact Center Deployments

Active Voice Call (speechpath) GW Control Via HTTP/VXML dp 1 TCL/VXML
PSTN
CV P
IP
dp 2 H.323
Database
When Agent Available, Active Voice Call Rerouted Via H.323 to Agent
Provides access to centralized IVR applications via branch office Provides edge queuing/IVR for contact center solutionskeeps RTP off WAN until agent is selected VXML Deflects call to agent when necessary RTP (VXML controlled) RTP (H.323 controlled) Supported with H.323 (SIP in future)
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Agenda
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Call Control Models

Distributed Call Control
Peer-to-Peer Protocols: H.323, SIP
GK
Centralized Call Control

Client-Server Protocols: SCCP, MGCP
Centralized Call Agent IP
IP Address Servers
IP H.323 Endpoint SIP Endpoint CCM

GW
SCCP Endpoint
PSTN
GW
PSTN
Peer-to-peer call setupIP address servers optional GW TDM signaling types local to GW Resilient over IP connectivity failures Scalable Distributed configuration
MGCP GW 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
Media (RTP)
Optional Signaling to Locate IP Address of Peer

Call Control Signaling
The Power of Cisco IOS Dial-Peers: H.323 and SIP

dp 10 pots dp 1 voip dp 2 voip dp 3 voip
PSTN
dp 11 pots dp 12 pots
IP
Dial-Peers Allow You To:

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 FXO These Capabilities do VXML call control for call centers Redistribute calls-in-q for CVP Not Exist for MGCPAA in the GW Controlled GWs
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GW Protocol Features with CCM

H.323
Centralized Provisioning QSIG Supplementary Services Centralized CDR (DS0 Granularity in CCM CDR) MLPP (Preemption) Hookflash Transfer with CCM ISDN Overlap Sending ISDN Fac IE Name Display NFAS AT&T Megacom NSF SRTP (CCM to GW) Mobility Manager VXML-Based Voice Profile Management Interstar Xmedius T.38 FaxServer Caller ID on FXO TCL/VXML Apps (e.g. CVP Integration) Voice+Data Integrated Access Fractional PRI 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
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SIP
MGCP
No GK
Legend:
Yes No
Partial CCM 5.0
Partial Future Test Test Future CVP 4.0
With Caveats
Workaround
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Protocol Deployment Considerations

Large/Campus Sites
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) to legacy PBXs required Other considerations
NFAS is H.323/SIP only Very high density GWs such as T3 (5x00) are H.323/SIP only
Small/Branch Sites
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 the same GW (not on same voice port) H.323 dial-peers are needed anyway for MGCP GW Fallback
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Protocol and Platform Summary

Gateway Platform
VG224 VG248 1751/60 2600XM, 2691 2800 3700 3800 5x00 7x00 Cat 6K CMM
Line Side SCCP (FXS)

Yes Yes No No Yes No Yes No No No
Trunk Side H.323

Yes No Yes Yes Yes Yes Yes Yes Yes Yes
SIP
Yes No Yes Yes Yes Yes Yes Yes Yes Yes
MGCP (CCM)
Yes No Yes Yes Yes Yes Yes No No Yes
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Agenda
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IP Phone Failover
Survivable 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 connectivityit does not provide or control GW connectivity or availability PSTN GW connectivity during failure modes:
POTS/VoIP dial-peers MGCP GWs requires the MGCP GW failover feature Calls from IP phones (under SRST) access the dial-peers to route calls
CCM Cluster
A
SCCP Keepalive to CCM
SCCP Keepalive to SRST if CCM Does Not Respond
PSTN
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X
WAN
Dial-Peers Control GW Call Routing
Voice Gateway Failover

H.323/SIP
SIP Proxy
POTS Dial-Peer VoIP Dial-Peers
MGCP
GW Switches to H.323/SIP/POTS Dial-Peers During MGCP Fallback
4 PSTN
1 2 3
IP
GK CCM
PSTN
MGCP Registration, CCM Keepalives and Backhaul
IP
H.323/SIP intrinsically resilient

No special failover features, just dial-peer configuration
Successive VoIP/POTS dialpeers (by preference) attempted on failure Each call setup attempt is treated independently
Same sequence of dial-peers
MGCP requires failover features When call agent is out of contact, MGCP GW fails over to local control
H.323, SIP or POTS dial-peers ISDN D-channel is reset to gain control of call state
Flapping IP links do not interfere significantly with call setup operation

Flapping IP links interfere significantly with call setup

Call agent registration and state are affected
IOS GW Audio Preservation Summary

H.323
Primary-to-Secondary CCM Failover ISDN Non-ISDN Flapping Links VXML Calls in Queue
12.4.4XC/12.4.9T, CCM4.1.3-SR2 preserve TCL Script to Reroute to SRST Hunt-Group 12.4.4XC/12.4.9T, CCM4.1.3-SR2 preserve Preserved (Disable TCP timer)
MGCP
Preserved
CCM to SRST
Fail Preserved Failures Depend on Timing N/A
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 testedbehaves like H.323
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Agenda
Gateway Deployment Scenarios Which GW Protocol? SIP, H.323 and MGCP Gateway Availability Considerations Supplementary Services with GWs
QSIG Supplementary Services Hookflash Transfer Station-and Trunk-Side FXS
Fax/Modem Capabilities Specialized Gateway Capabilities Gateway Capacity

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QSIG Supplementary Services

Toll Bypass QSIG
QSIG Supplementary Services between PBXs
QSIG
CCM QSIG
QSIG Supplementary Services between PBX and CCM
QSIG
IP
QSIG Services Tunneled Via H.323
QSIG
IP
QSIG Services Tunneled Via MGCP
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 farend 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
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
TCL
PSTN
3. HF
2. Redirect
1. Call to CUE AA
Analog FXO PSTN trunks Requires PSTN service for HF xfer PSTN xfers the call, releases trunk to GW Requires custom-developed TCL script
1. Call from PBX to Analog Phone
IP
5. Xfer 4. HF on FXO 3. H.323 HF Relay 2. HF on FXS
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
4. Xfer
3. *8nnn (inband)
VXML
2. Send *8nnn (OOB)
PSTN
1. Call from PSTN to IVR
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IP
Uses PSTN HF access code service, e.g. *8 CVP instructs GW to send digits to PSTN inbandT1 CAS/PRI trunks PSTN xfers call, releases trunk to GW
Hookflash Transfer
Requires PSTN PRI TBCT (2-b-channel xfer) service CVP instructs GW to initiate TBCT to PSTNPRI trunks PSTN xfers call, releases trunk to GW Requires TCL script
4. Xfer
TCL
VXML
3. TBCT
2. Do TBCT
PSTN
1. Call From PSTN to IVR
IP
1. Call to TDM IVR/Vmail system
IP
4. Xfer 3. MGCP HF Notification 2. HF on T1 CAS
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
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Station and Trunk-Side FXS Gateways

Station-Side FXS
Featured Phone
ATA FXS SCCP SCCP FXS VG224
Trunk-Side FXS
POTS Phone
FXS
PBX
IP
MGCP, H.323, SIP FXS
Key System
Station-Side FXS
SCCP control of FXS portsATA, VG224, VG248, IOS GW FXS ports with 12.4.9T Use this for featured analog phones Supplementary features like transfer, conference, call waiting, call park, FAC codes, MWI stutter dial-tone, redial
Trunk-Side FXS
MGCP, H.323, SIP control of FXS portsIOS GW FXS ports Basic Call, CLID and hookflash blind transfer
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Agenda
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Fax over IP Methods

Originating Terminating
Fax passthrough:
Negotiation and switchover in H.323 and SIP protocol No negotiation and switchover in NSE
Capability Negotiation Using the Protocol Stack Voice Call Established
Fax-relay Cisco:
Switchover in NSE
Detects 2100Hz CED Tone From Answering Fax machine
Fax-relay T.38
Negotiation and switchover in H.323 and SIP protocol Negotiation in protocol H.323, SIP and MGCP and switchover in NSE No negotiation and switchover in NSE
Request Switchover
Acknowledge Switchover
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Switchover to Fax over IP

Switchover done via call control protocol
In H.323 the switchover is indicated using the RequestMode H.245 message; in SIP the switchover is indicated using a ReInvite Both gateways involved in the call should use the same protocol used when interoperating with third party devices Call agent must support the fax method negotiated
Switchover done via NSE

NSE events are used to indicate the switchover to passthrough, Cisco fax relay or T.38 fax relay Independent of the call control protocol, therefore best for mixed protocol environment NSEs are Cisco specific and can only be used with other Cisco endpoints Does not require call agent to support the fax method
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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)
Latency, Jitter, Buffers, Packet loss
PSTN
FAX/Modem Over IP
Voice GW
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 PSTN
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Modem over IP Methods

Originating Terminating
Modem passthrough:
Capability Negotiation Voice call Established Upspeed Requested Acknowledge Upspeed Passthrough Mode(VBD) Ecan Off NSE Ecan Off NSE ACK Modem Passthrough Switchover MR ACK MR Switchover
No negotiation and switchover in NSE
AnsAM
Modem relay signal assisted

Negotiation in Protocol H.323, SIP and MGCP and switchover in NSE
Phase Reversal
Modem relay gateway controlled

Negotiation and switchover in NSE
CM Tone
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Modem Relay Methods Comparison

Signal Assisted
Introduced in 12.2(11)T Capabilities are exchanged during call setup
MGCP/SIP uses SDP for MR parameter negotiation H.323 uses H.245 for MR parameter negotiation
Gateway Controlled
Introduced in 12.4(4)T MR gateway capabilities are not exchanged during call setup
GW will use configured values In case of configuration mismatch, fallback to modem passthrough
Cisco NSE mechanism is used for media switching from voice to pass-through to relay Not Supported with CCM
Cisco NSE mechanism is used for media switching from voice to pass-through to relay Supported with CCM
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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 Access Video Switching Drop and Insert Channel-Bank Clocking
Gateway Capacity
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Integrated Voice + Data Access

Single T1/E1 Voice Single T1/E1
PSTN
WAN
Voice
Data
Data
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 Data (Channel-Group: FR, HDLC, PPP, MLPPP) Voice (ds0, or pri-group) D-Channel (if pri-group is Used), Controls Voice Channels Only
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 Voice channels: Voice bearer CAP Data channels: 64K unrestricted bearer CAP
T1
T1/E1
Voice + Data (pri-group)
D-Channel Controls All Channels

Integrated Voice + Data PRI

Dial Backup Via PSTN Connection During WAN Failure
Single T1/E1 PRI Voice Single T1/E1 PRI Voice
PSTN
Backup Data
Data
WAN
Data
PRI WAN Connection Sharing Same Interface as Voice (SP Offering)

Single T1/E1 PRI Voice Data
Data Services Voice Services
Single T1/E1 PRI Voice Data
Integrated Service Provider

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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 group PRI: need 1 HDLC controller per channel (timeslot)
Requires 12.4.9T minimum
ISR Voice Gateway

VoIP Packets
DSPs
TDM (Voice/Video) VWIC TDM (Data)
IP
Cisco IOS SW Data Packets
T1/E1
TDM Backplane
HDLC Controllers
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ISDN Switching: Video

dial-peer voice 20 pots destination-pattern 9T port 2/0/16 dial-peer voice 3000 voip destination-pattern 688.... session target ipv4:1.1.1.1 dtmf-relay h245-alphanumeric
ISDN switching uses

Pri-groups POTS dial-peers H.323/SIP for voice termination TDM Backplane Signaling
Video Media
EVM
EM-4BRINT/TE
BRI
Cisco IOS SW
Voice Media
IP
dial-peer voice 21 pots destination-pattern 9T port 1/0:15
Onboard DSPs
NM-HDV2-1T1/E1 T1/E1
ISR Voice Gateway
NM-HDV2 DSPs
For the Full Configuration of Video ISDN Switching, See http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00804794c6.shtml

46 46
DSP Dropping for Locally Switched Calls

1700 WIC 2800/ 3800 WIC NM1V/2V NMHDA NM- AIM-[ATM]- NM-HDNMHDV VOICE-30 1V/2V/2VE HDV2
EVM
1700 WIC 2800/3800 WIC NM-1V/2V NM-HDA NM-HDV AIM-[ATM]VOICE-30 NM-HD1V/2V/2VE NM-HDV2 EVM
No Yes
N/A N/A No
N/A No No No
N/A No No No No
N/A N/A No No No No
N/A Yes No No No No Yes*
N/A Yes No No No No Yes* Yes*
N/A Yes N/A No No N/A Yes Yes Yes
*Supported on the 2800 and 3800 Series Routers

47 47
Modem Relay Methods Comparison

D&I is a L1 cross-connect (HW switch from designated ingress to egress ports)
tdm-groups connect tdm CLI
Ingress and egress interface must be the same type

Cannot D&I T1/E1 to BRI Cannot D&I T1 to E1 or vice versa Cannot D&I T1 SF-AMI to T1 ESF-B8ZS
Statically configured Cisco IOS CPU/software does not process the traffic or the signaling Any kind of traffic can be carried
ISR Router
Cisco IOS SW
TDM Backplane
HWIC
VWIC
NM-HDV2-1T1/E1
controller T1 1/0/0 framing esf linecoding b8zs T1/E1 tdm-group 2 timeslots 13-24 type xxx ! controller T1 1/1/0 framing esf linecoding b8zs tdm-group 3 timeslots 13-24 type xxx clock source line primary T1/E1 ! connect tdm1 T1 1/0/0 2 T1 1/1/0 3
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Drop and Insert (D&I) vs. TDM Switching

D&I
Configuration DSPs Interfaces Traffic Types Signaling Types Switching Flexibility #Interfaces #Channels ISDN traffic
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TDM Switching
POTS Dial-Peers DSPs Required Ingress and Egress Interface Are Independent InterpretedBoth Interfaces Are Terminated on the Router Signaling is TerminatedOnly TDM Signaling Supported by GW Can Be Switched Dynamic, Destination Determined on a Per Call Basis Any Number of Interfaces Present on the GW Any Number of Ingress/Egress Channels Can Be Switched Individual Channels Can Be Switched to Different Destinations
Connect tdm CLI No DSPs Needed Ingress and Egress Interface Must Be the Same Type Not InterpretedStatically Switched Based on Config Signaling Is Not Terminated Any Traffic Can Be Switched Static One Ingress Mapped to One Egress Interface #Ingress channels = #Egress Channels All Channels on the PRI Must Be D&Id
D&I (Cross-Connect) Configurations

Analog-to-Digital Cross-Connect
Voice Gateway
HWIC Slot
VWIC
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
PSTN
VoIP SW
T1 FXO Loopstart T1 FXS Loopstart
EM-HDA-8FXS EVM-HD
FXS Loopstart
Two Analog Phones CrossConnected into the PSTN, no VoIP Access Analog Phone with VoIP Access
Digital-to-Digital Cross-Connect
24-Channel T1
VWIC
TDM
Up to 24 Analog Phones
EVM-HD8FXS/DID with 24x FXS

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50 50
Clocking Domains on the ISR Voice Gateways

ISR Motherboard Domain Voice HWIC Gateway
DSP TDM Backplane ASIC EVM
HWIC HWIC HWIC
NM Domain
DSP NM
NM Domain
DSP NM
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 syncd 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)
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Clocking CLI Configurations

Clock Source Line NoDSP Network Clock DSP
Rx
TDM BackPlane
Clock Source Line Network Clock Participate DSP

Rx
TDM BackPlane
NM
Clock Looped PLL
Tx
Tx Clock Source from Another Interface Marked as select, or system PLL
Note: Network Clock in These Examples Refers to the Cisco IOS CLI Commands: Network clock select Network clock participate
Clock Source Internal No Network Clock DSP

Rx
TDM BackPlane
Clock Source Internal Network Clock Participate DSP

Rx
TDM BackPlane
Clock Source Line Network Clock Select T1 0/0/0 Network Clock Participate DSP
Clock Source to Other Interfaces
NM
Tx PLL
Tx Clock Source from Another Interface Marked as select, or system PLL
TDM BackPlane
Rx
Port 0/0/0
Tx PLL
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Additional Notes on Clocking

Clocking configurations must be correct:
To ensure good voice qualityclock slips impair voice quality For correct fax and modem traffic handling TDM switching and D&I operation
VWIC cards support a single external clock source

Dual clocking is supported only if both ports are data
All interfaces using motherboard DSPs must be clocked 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 not important
Carefully set up clocking for WAN and voice configurations

53 53
Agenda
VVT-2010 VVT-2010 12625_04_2006_c2` 12625_04_2006_c2
54 54
Cisco Voice Gateway Capacity

Physical Channel (DS0) Connectivity
2600XM VG224 2691 2811 2821 2851 1751 1760 2801 3725 3745 3825 3845
FXS FXO/CAMA E&M Analog-DID BRI Ports BRI Channels Total T1/E1 Ports Onboard T1/E1 Ports NM/AIM-Based T1/E1 Ports T1 Channels Connectivity Onboard NM/AIM-Based E1 Channels Connectivity Onboard NM/AIM-Based
12 12 6 4 6 12 4 4 0 96 96 0 120 120 0
16 16 8 8 8 16 4 4 0 96 96 0 120 120 0
16 16 8 8 8 16 8 8 0 192 192 0 240 240 0
24 0 0 0 0 0 0 0 0 0 0 0 0 0 0
12 8 4 4 4 8 5
12 8 4 4 4 8 6
28 24 12 12 12 24 12 8
52 36 12 20 20 40 12 8 4 288 192 96 360 240 120
52 36 12 20 20 40 12 8 4 288 192 96 360 240 120
24 16 8 8 8 16 10
48 32 16 16 16 32 18
52 36 16 20 20 40 16 8
88 56 24 32 32 64 24 8 16 576 192 384 720 240 480

55 55
5 120 0 120 150 0 150
6 144 0 144 180 0 180
4 288 192 96 360 240 120
10 240 0 240 300 0 300
18 432 0 432 540 0 540
8 384 192 192 480 240 240

Voice Termination Channel Capacity: 12.4
FE/GE
600 550
WAN No-cRTP cRTP
# Calls at 75% CPU
500 450 400 350 300 250 200 150 100 50 0

2611XM 2621XM 2651XM
450
340 290 220 180 170 150 140 130 120 102 100 82 170 290 250
330 270
32 18
38
12
2016
5035
70
24
48 35
112 88 61
2811
2821
2851
2691
3725
3745
3825
3845
56 56
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VXML Sessions: 12.4
Dedicated VXML Service Voice Gateway and VXML VXML and DTMF 6 24 36 56 50 77 96 144 90 192 192 VXML and ASR/TTS 4 20 30 48 38 60 72 96 72 192 192 Memory Recommended 256MB 256MB 256MB 512MB 512MB 512MB 512MB 512MB Default Default Default VXML and DTMF 7 30 48 60 68 100 120 150 96 240 240 VXML and ASR/TTS 6 24 36 56 50 80 96 144 90 192 192
Platform
Cisco 2801 Cisco 2811 Cisco 2821 Cisco 2851 Cisco 3725 Cisco 3745 Cisco 3825 Cisco 3845 AS5400HPX AS5350XM AS5400XM
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.
57 57
Session Capacity for RSVP Agent

License entitlement
Cisco Unified Survivable Remote Site Telephony: defined by the number of licensed sessions and router performance Cisco IP-to-IP Gateway: limited only by router performance
Session capacity
2611XM 40 sessions 2621XM 50 sessions 2651XM 65 sessions 2691 2801 2811 150 sessions 130 sessions 180 sessions 2821 2851 3725 3745 3825 3845 240 sessions 300 sessions 250 sessions 320 sessions 400 sessions 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
58 58
Router-Based Media Services
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59 59
Agenda
Media Services Design, Operation and Configuration DSP Engineering, Allocation, Sharing Case Study
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60 60
IP-Based Media Services Overview

IVR
Conferencing
Mixing RTP streams for multi-party conference bridges
Cisco CallManager Cluster
Transcoding
Support multiple codecs on the same call (e.g., G.711 to G.729) Changing codec needed in presence of single-codec endpoints and applications
Transcoding Xcod DSPs
Conferencing Conf DSPs
MTP
Anchor an RTP stream to change headers or change treatment of the call
Business VoIP
PSTN
IP WAN
SBC
SBC (session border controller, or IP-IP GW)

Network boundary demarcation (security or billing) Protocol translation (e.g. H.323 to SIP)
...
Conference Media Resources

HQ Branch A B MRGL
1. Br1 2. HQ1 3. HQ2
PSTN
IP WAN
Conf
External caller X calls Ano voice across WAN A conferences B Three voice streams across WAN to MRGL central mixing
1. HQ1 2. HQ2
HQ
PSTN
A B
Device Pool
Conf
IP WAN
Conf Conf Conf Conf
MRG=Br1
Device Pool
Conference between A, B and Xno voice across WAN Utilizes DSPs in the branch router for local mixing
MRG = Media Resource Group MRGL = Media Resource Group List
Branch
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MRG=HQ1 MRG=HQ2
Conferencing Operation and Design

When Necessary Where Located Selection Algorithm
Offload central conference bridge software servers To keep conferences, between participants at a remote site, from crossing bandwidth-constrained (WAN) links Conf participants at same sitelocal Conf participants at different sitesany site Based on the location of the initiator of the conference Conference resources registered with CCM MRG/MRGL configuration order and device pools 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 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
Codecs
Other Considerations
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Transcoding Media Services

VMail
PSTN
G.711 G.729
VoIP WAN
Xcod
Branch
VMail
HQ
Call from HQ to branch remains G.729A when is CFNA/CFB to local branch-based voice mail that requires G.711 Utilizes DSPs in the branch router
G.729 G.711
Xcod
VoIP WAN
Similar scenario for CME network CME Site A maintains G.729A across the WAN
CME Site B
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PSTN
64 64
Transcoding Operation and Design

When Necessary Bridge two call legs that use different codecs To maintain G.729A WAN bandwidth for G.711only applications To enable calls between endpoints with no common codec support Transcoding resources registered with CCM MRG/MRGL configuration order and device pools G.711 a/-law to/from G.729, G.729A, G.729B, G.729AB G.711 a-law to/from G.711 -law Different packetizations
G.711: 10ms, 20ms and 30ms G.729: 10ms, 20ms, 30ms, 40ms, 50ms and 60ms
Where located Collocated with the G.711 endpoint Selection Algorithm
Codecs
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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
MTP Media Services

VMail
PSTN IP WAN
Xcod
MTP
Branch
PSTN Business VoIP IP WAN
MTP
An MTP anchors the RTP stream Provides supplementary services for devices H.323 Video that cannot support Device HQ ECS
Provides a single IP address for all endpoints at the site to an outside network connection HQ
Branch
66 66
MTP Operation and Design

When Necessary 1)Security demarcationIP address hiding 2) RSVP proxy for CAC 3) Anchor point for supplementary services 4) CCM 4.0 SIP trunks (no longer required with CCM 5.0) 5) DTMF translation For #12: collocated at the site For #35: anywhere MTP resources registered with CCM MRG/MRGL configuration order and device pools SW-MTP: same codec, same packetization HW-MTP: same codec, different packetization (Transcoding: different codec, same/different packetization) 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)
Where Located Selection Algorithm Codecs
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Session Border Controller (IP-IP Gateway)

PBX PBX
PSTN
WAN PSTN SBC (IP-to-IP Gateway) can be used for

Network Privacy/Security Demarcation Protocol Interworking DTMF Interworking Transcoding G.711 to G.729 Quality of Service Packet Marking Call Admission Control RSVP Call Detail Records Class Of Restriction Point of Demarcation for Troubleshooting Integrated TDM functionality Integrated Gatekeeper functionality
VoIP SP
SBC
More Information: VVT-2015: Interconnection of Voice and Video Networks using the Cisco Multiservice IP-to-IP Gateway
68 68
DSP Media Services Configuration Example

Define the DSPs Used for Media Resources voice-card 1 dsp services dspfarm ! IP Address Used on Packets sccp local FastEthernet0/0 to CCM Taken From This sccp ccm 10.4.20.24 identifier 1 version 4.0 Interface sccp ccm 10.4.20.25 identifier 2 version 4.0 sccp ccm 10.4.20.26 identifier 3 version 4.0 Definition of the CCMs the sccp ip precedence 3 Router Registers with sccp ! Multiple [groups of] CCMs can be defined sccp ccm group 988 associate ccm 1 priority 1 Priority of CCMs for Failover of associate ccm 2 priority 2 associate ccm 3 priority 3 Registration associate profile 10 register CFB123456789966 associate profile 6 register MTP123456789988 Router Capabilities keepalive retries 5 (Profiles) Registered switchover method immediate with CCM switchback method immediate switchback interval 15 Failover Timers and Behavior
69 69
DSP Media Services Configuration Example

dspfarm profile 6 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 10 associate application SCCP ! ! ! dspfarm profile 10 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 6 associate 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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Call Agents, Registration, Failover CCM

IP
Conf
1 2 3
Xcod
sccp local FastEthernet0/0 sccp ccm 10.4.20.24 identifier 1 version 4.0 sccp ccm 10.4.20.25 identifier 2 version 4.0 sccp ccm 10.4.20.26 identifier 3 version 4.0 ! sccp ccm group 988 associate ccm 1 priority 1 associate ccm 2 priority 2 associate ccm 3 priority 3 keepalive retries 5 switchover method immediate switchback method immediate switchback interval 15
CME, SBC
Conf
Xcod
telephony-service ip source-address 192.168.1.1 port 2000 sdspfarm units 1 sdspfarm transcode sessions 16 sdspfarm tag 1 MTP000f23cd6100 ! sccp local Vlan10 sccp ccm 192.168.1.1 identifier 1 sccp ! sccp ccm group 1 associate ccm 1 priority 1
SRST
X
Conf
Xcod
Conference, Transcoding and MTP Resources Are Not yet Available During SRST Mode
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Media Resources on a Single Router with Multiple CCMs

Share Router Resources Among Multiple CCMs
Each CCM must have a separate 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
IP
CCM2: y.y.y.y
Define the CCMs Define One or More Profiles for Each CCM
Associate Each Profile with One of the CCMs

Controlling IP Addresses
interface FastEthernet0/0 ip address 10.9.68.100 255.255.255.0 ! sccp local FastEthernet0/0 sccp ccm 10.9.20.24 identifier 1 version 4.0 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 Loopback1 sccp ccm 10.9.64.138 identifier 10 version 4.1 sccp local FastEthernet0/0 sccp ccm 10.9.20.24 identifier 1 version 4.0 ! sccp ccm group 1 associate ccm 1 priority 1 bind interface FastEthernet0/0
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 messagesglobal The bind interface commands controls the IP address on media packetsper CCM group
Using the IP-IP GW as a Transcoder

CCM Transcoding
G.711
PSTN
WAN
G.729
Xcod
Transcoder control is in CCM based on regions and endpoint capabilities CCM is aware of both legs of the call DSPs required on router
IP-IP GW Transcoding
G.711
SP VoIP
IP-to-IP Xcod
WAN
G.729
Transcoder control is in IP-IP GWbased on mismatched codecs on ingress/egress call legs CCM is aware of only one leg of the call (single-codec call from CCMs point of view) DSP required on IP-IP GW
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Agenda
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75 75
How Many DSPs Do I Require?

PVDM2 DSP Basic Rules
Each DSP is treated individuallyPVDM2 boundaries are irrelevant Conferencing
Requires a dedicated DSP Up to eight participants per conference Single-mode: all participants are G.711 a-law/-law; eight* fixed conferences (up to 64 participants) per DSPstatic configuration Mixed-mode: at least one participant is G.729A; two* fixed conferences (up to 16 participants) per DSPstatic configuration Conferences cannot span multiple DSPs
Voice Termination, Transcoding, MTP

These functions can share a DSP G.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
76 76
How Many DSPs Do I Require?

DSP Calculator Tool
http://www.cisco.com/cgibin/Support/DSP/cisco_adv.pl
Configure Router
Cards and Voice Termination channels
Configure Options
Conf, Xcod, MTP, Analog Reservation, IP SLA
Results
DSP Cards and # DSPs
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77 77
Where Should the DSPs Be Located?

DSP Architecture on the ISRs
Motherboard Domain
HWIC HWIC HWIC HWIC
DSP
ASIC EVM NM Domain

DSP NM-HDV2
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 domain
Search order: local domain first, then remote domains starting with slot 0
Conf/Xcod/MTP can use DSPs from any domain
PVDM2s
NM Domain
DSP NM-HDV2
Name PVDM2-8 PVDM2-16 PVDM2-32 PVDM2-48 PVDM2-64
#DSPs Half 1 2 3 4
G.711 Channels 8 16 32 48 64
G.729A Channels 4 8 16 24 32
G729 Channels 4 6 12 18 24
Additional NMs
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PVDM2 DSP AllocationExample

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 2464 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 time Media DSPs assigned at runtime (oversubscription)
Order of DSP allocation may be different after router reboot

79 79
How Are the DSPs Allocated to Services?

DSP Channel Allocation Algorithm
Voice Termination, Xcoding, MTP
Media Channels#Channels Depends on Combination of Codecs: Six to Sixteen Channels per DSP
G.711 Only Conference

Conf One Conf Two Conf Three Conf Four Conf Five Conf Six Conf Seven Conf Eight Conf Two
Up to Eight Participants Up to Eight Participants
G.711/G.729A
1 2 3 4 5 6 7 8 9 10 11 12
G.711 G.729A G.729A G.711 G.729A
Conf One
Up to Sixteen Signaling Channels Assigned per DSP

80 80
How Are the DSPs Shared?

DSP Access with DSP Sharing
Cisco 2800/3800 Router
TDM Backplane HWIC DSP Access
VWIC
T1/E1 FXS, FXO, E&M, BRI

FXS, FXO or BRI
Onboard DSP Slots

PVDM2s
VIC
HWIC
EM
EVM-HD NM-HDV2-1T1/E1
FXS/DID T1/E1
DSP sharing pools together all the PVDM2 DSPs present in the 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 DSPs Turn on network clocking for all cards that share DSPs
PVDM2s
VIC
FXS, FXO, E&M, BRI T1/E1
PVDM-12s
NM-HDV
VWIC
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PVDM2 DSP Codecs

Complexity and Channel Support
Channels Per DSP in High Compl. (HC)
6 6 6 6 6 6 6 6 6 6
Reference
Codec
G.711 (-law, a-law) Fax/Modem Passthrough Clear-Channel Codec G.726 (32K, 24K, 16K) Fax Relay G.729A, G.729AB G.729, G.729B, G.728 G.728 G.723.1 (5.3K, 6.3K), G.723.1A (5.3K, 6.3K) Modem Relay
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Channels Per DSP in Med Compl. (MC)

8 8 8 8 8 8 Not Supported Not Supported Not Supported Not Supported
Channels Per DSP in Flex Compl. (FC)

16 16 16 8 8 8 6 6 6 6
Low Complexity Codecs Medium Complexity Codecs
High Complexity Codecs
Reference
PVDM2 DSP Capacity Summary

Onboard Domain Application
G.711a/law
NM Domain
2801/2811 2821/2851 3825/45 NM-HD-2VE NM-HDV2 (18 DSPs) (112 DSPs) (116 DSPs) (3 DSPs) (116 DSPs) 128 64 48 64 48
50 conf* (400 parties)
192 96 72 96 72
50 conf* (400 parties) 24 conf (192 parties)
256 128 96 128
48 24 18 24
256 128 96 128
Voice Termination
G.729A (MC Codecs) G.729 (HC Codecs) G.711a/law G.729A /G.729AB
Transcoding
G.711a/law G.729 /G.729B G.711 Conference (Single-Mode)
96 50 conf* (400 parties) 32 conf (256 parties)
18 24 conf (192 parties) 6 conf (48 parties)
96 50 conf* (400 parties) 32 conf (256 parties)
ConG.729 Conference 16 conf ference (Mixed Mode) (128 parties)
*The Maximum Number of Conference Sessions Are Limited to 50 (with 400 Participants by HW/IO
83 83
Reference
Summary: DSPs, Platforms, Interfaces

Platform/Interface
1751/1760 Onboard 2801, 2811 (Onboard PVDM2) 2821, 2851 (Onboard PVDM2) 3825, 3845 (Onboard PVDM2) NM-1V/2V NM-HDA NM-HDV
DSP
TI-549 TI-5510 TI-5510 TI-5510 TI-542 TI-5421 TI-549
#Fixed #Expandable Voice Channels Per DSP XCod/ DSPs DSPs Conf HC MC FC
10 8 12 16 1 2 2 6 6 6 2 4 2 4 6 6 6 4 8 8 8 6 6-16 6-16 6-16 Yes Yes Yes Yes No No Yes No 6-16 6-16 6-16 Yes Yes Yes
2 15
8 4 8 8 8 8
AIM-VOICE-30, AIM-ATMTI-5421 VOICE-30 NM-HD-1V/2V NM-HD-2VE NM-HDV2 (PVDM2) TI-5510 TI-5510 TI-5510
4 1 3 16
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Agenda
VVT-2010 VVT-2010 12625_04_2006_c2` 12625_04_2006_c2
85 85
Branch Office Requirements

PSTN
PRI CUE VoiceMail FXO
Campus
WAN
Centralized CCM at another site Use G.711 within the site, and G.729A to other sites Branch site: 50 users
12 PSTN channelsfractional PRI 66% terminate on local phones (G.711) 33% of calls terminate at another site (G.729A), perhaps after a transfer Four FXO lines for backup Six fax machines Six to ten people in two conferences at any one time Five transcoding channels for calls from other sites into local voicemail
Fax Machines
Cisco 2821 Router with SRST
Users
Branch Office
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 PSTN WAN
T1
VWIC2-1MFT-T1/E1 HWICEmpty NMData
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T1 PRI
VWIC2-1MFT-T1/E1 HWICEmpty
FXO
EM-HDA-6FXO EVM-HD-8FXS/DID
2821
6x Fax Machines
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How Many DSPs Do We need?
Four total DSPs

Three DSPs for voice channels and xcoding One DSP for conference
PVDM2-64 Installed on router motherboard

CISCO2821-SRST/K9 (includes a PVDM2-32)
88 88
Router Configuration for Voice

voice-card 0 dsp services dspfarm ! controller T1 1/0/0 framing esf linecode b8zs pri-group timeslots 1-12,24 ! interface Loopback0 ip address 10.9.100.1 255.255.255.255 ! interface Serial0/0/0 ip address 10.9.101.6 255.255.255.252 clock rate 2000000 ! interface Serial1/0/0:23 no ip address encapsulation hdlc isdn switch-type primary-ni isdn incoming-voice voice ! voice-port 1/0/0:23
Configure the DSPs for Use as Media Resources PSTN Fractional PRI with 12 Channels Loopback Interface Used to Maintain CCM Connectivity
WAN Connectivity
D-Channel for the Voice PRI

Router Configuration for Voice (Cont.)

sccp local Loopback0 sccp ccm 10.9.64.108 identifier 3 version 5.0.1 sccp ! sccp ccm group 2 bind interface Loopback0 associate ccm 3 priority 1 associate profile 1 register XCODE123456 associate profile 2 register CONF12345 ! dspfarm profile 1 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP ! dspfarm profile 2 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 5 associate application SCCP
IP Address Used on Packets to CCM Taken from This Interface
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

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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Voice Encryption: SRTP

Site A PSTN Phone-to-phone and phone-to-GW calls protected
MGCP and H.323 GWs
Site B
IP Central Site
SRTP protects end-toend media TLS or IPSec protects end-to-end signaling H.323 encryption supported for CCM and toll bypass configurations
SRTP (Media) TLS or IPSec (Signaling)

93 93
RSVP Agent
SCCP
Media Resource Control
HQ
SCCP
Media Resource Control
RTP
RSVP
RTP
RSVP Agent
WAN
RSVP Agent
Location A
Location B
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
94 94
IP SLA: Measurements with SAA

San Diego 2
Data Center
Cisco CallManager Cluster
SAA End-to-End Measurements Edge-to-edge Hop-by-hop Edge-to-server
SAA
San Diego 1
Headquarters
SAA
Responder
SAA SAA
Chicago
Regional Office
Seattle
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L.A.
San Jose Cleveland Detroit
New York
Boston
95 95
Sales Office Sales Office Sales Office

Sales Office Sales Office

Circuit Emulation over IP

Radar Encryption SCADA Telemetry
TDM TDM
Applications that Require BitTransparent Circuits Cant Be Natively Integrated into a Packet Network, but They Can Be Tunneled Across the IP Network Using Circuit Emulation
H.320 Video Cell Site Backhaul T1/E1 Leased Line Emulation Other Applications VoIP IP Telephony IP, ATM, FR
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CESovIP
CPE
CESovIP
CPE
IP
Data Data
96 96
Traffic Monitoring (NM-NAM)

Security
Operations
NM-NAM
IP L2/L3
Firewall IDS
Monitoring Remote Sites Through Web Based Traffic Analyzer
6K-NAM Content
IDS
Video Surv.
NM-NAM Available for Cisco NM-NAM 2800/3800 Series Routers 6K-NAM 6K-NAM Available for Cisco
Catalyst 6500 Switches and Cisco 7600 Series Routers
IP
6K-NAM NetFlow Data Export AAA to 6K-NAM
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
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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 Messaging Cisco CallManager Cluster Centralized Call Processing Applications (UM, IVR, ICD)
Networked Branch A
PSTN
Fat Pipe
IP
Small Pipe Loosely Coupled Cisco CallManager Express and Cisco Unity Express Localized Call Processing Localized Messaging
Central Site
Branch B
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Q and A
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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 consideredplacing 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 PreparationImplementing Gateways CRT-2204: GWGK Exam PreparationImplementing 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
101 101
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 and Unity Express Network Architecture VVT-2106: Deploying CallManager Express and Unity Express: Advanced Deployment Scenarios, Management and Security
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ReferencesGeneral
DSP Calculator http://www.cisco.com/cgi-bin/Support/DSP/cisco_dsp_calc.pl Integrated Services Router General Information http://www.cisco.com/go/isr Cisco Voice Gateway Router Interoperability with CallManager http://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/prod ucts_implementation_design_guide_chapter09186a008044750 d.html
103 103
ReferencesCisco IOS Voice Features

Cisco IOS Voice Configuration Library
http://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124tcg/ vcl.htm
Cisco IOS H.323 Configuration Guide

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_confi guration_guide_book09186a00801fcee1.html
Cisco IOS SIP Configuration Guide

http://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 Guide

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_confi guration_guide_book09186a008020bd29.html
Cisco IOS IP SLAs Configuration Guide

http://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124cg/h sla_c/index.htm
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Recommended Reading
Continue your Cisco Networkers learning experience with further reading from Cisco Press Check the Recommended Reading flyer for suggested books
Cisco Voice Gateways and Gatekeepers [1-58705-258-X]available August 2006 Cisco 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

105 105
Complete Your Online Session Evaluation

Win fabulous prizes; Give us your feedback Receive ten Passport Points for each session evaluation you complete Go to the Internet stations located throughout the Convention Center to complete your session evaluation Drawings will be held in the World of Solutions
Tuesday, June 20 at 12:15 p.m. Wednesday, June 21 at 12:15 p.m. Thursday, June 22 at 12:15 p.m. and 2:00 p.m.
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Design Voice GW

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Designing Voice Gateways and Media Resources in Enterprise Networks

2006 Cisco Systems, Inc. All rights reserved.

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Cisco Public Cisco Public

Scope of This Seminar

Cisco Public Cisco Public

Platforms and Voice Interface Capabilities

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Cisco Public Cisco Public

Cisco Voice Gateways

Performance and Services

Cisco 3800 Series

Cisco 5350XM, 5400XM, 5850

Cisco Catalyst 6500 CMM

Cisco 2800, 2600XM, 2691 Series Cisco VG248

Enterprise Branch SMB/Small Branch Teleworker/SOHO

Cisco 2801 1751, 1760

Cisco VG224 Cisco ATA 186

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Voice Gateway Selection Criteria

Feature Set Focus

Analog, BRI, T1/E1 FXS FXS

Gateway Signaling Protocol Summary

Cisco Public Cisco Public

GW Interface Card Port Densities

Station-Side FXS Trunk-Side FXO BRI T1/E1

Cisco Public Cisco Public

Analog/BRI Voice Interface Card (VIC)

T1/E1 Voice/WAN Interface Card (VWIC)

High-Density Analog/BRI Voice (EVM-HD)

Cisco Public Cisco Public

EVM-HD Expansion Module Combinations

BRI FXS FXO Ports B-Ch 8 8 16 11 8 8 3 6 3 3 4 8 10 4 6 12 6 4 4 8 8 8 16 4 8 4 6 4 8 16 24 23 22 24 15 22 21 23 14 20 22 16 24

EM-HDA-3FXS/4FXO EM-HDA-3FXS/4FXO EM-HDA-3FXS/4FXO EM-HDA-3FXS/4FXO EM-HDA-6FXO EM-HDA-6FXO EM-HDA-6FXO EM-4BRI-NT/TE EM-4BRI-NT/TE

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Cisco Public Cisco Public

DSP Architecture of the ISRs

Each NM-HDV2 Provides Four More DSP Slots

2801 2811 2821/51 3825 3845

IP Phone Inline Power Support

NME-16ES-1GE NME-X-24ES-1GE NME-XD-24ES-2ST NME-XD-48ES-2GE Max # POE Switchports

Cisco POE Support

802.3af POE Support

Max # IP Phones (7W)

GW Design: Features, Protocols and Operation

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Gateway-to-Gateway Network Deployments

CCM Network Deployments

GW platforms often integrate other IP-based voice services

H.323 or SIP MGCP, SIP or H.323

Cisco Public Cisco Public

CCM and CME Network Deployments

H.323, SIP MGCP, SIP, or H.323

SRST CME Branch 1 CME Branch 2 Branch 1

Distributed Contact Center Deployments

Cisco Public Cisco Public

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Call Control Models

Centralized Call Control