Design Voice GW

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


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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









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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









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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




• Supplementary Services with GWsQSIG Supplementary Services

“Hookflash” Transfer

Station-and Trunk-Side FXS





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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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• 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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• 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


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Agenda









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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


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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


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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


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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


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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


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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



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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


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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


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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


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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

http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00804794c6.shtml


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

Agenda









12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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.


12625_04_2006_c2`

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


12625_04_2006_c2`

Router-Based Media Services


12625_04_2006_c2`

Agenda

• Media Services Design, Operation and Configuration

• DSP Engineering, Allocation, Sharing

• Case Study


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


• 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


12625_04_2006_c2`

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


12625_04_2006_c2`

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)


• 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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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…


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

Agenda



• Case Study


12625_04_2006_c2`

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


12625_04_2006_c2`

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




12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


12625_04_2006_c2`

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


18

24

18



6 conf (48 parties)

192

96

72

96

72



128

64

48

64

48



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



• 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


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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


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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


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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

http://www.cisco.com/cgi-bin/Support/DSP/cisco_dsp_calc.pl

http://www.cisco.com/go/isr

http://www.cisco.com/en/US/products/ps5855/products_data_sheet09186a0080182d38.html

http://www.cisco.com/en/US/products/ps5855/products_data_sheet09186a0080182d38.html

http://www.cisco.com/en/US/products/ps5855/products_data_sheet0900aecd801b97a6.html

http://www.cisco.com/en/US/products/ps5855/products_data_sheet0900aecd801b97a6.html

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_implementation_design_guide_chapter09186a008044750d.html




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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

http://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124tcg/vcl.htm

http://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124tcg/vcl.htm

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_book09186a00801fcee1.html

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_book09186a00801fcee1.html

http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipc1_c/index.htm

http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipc1_c/index.htm

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_book09186a008020bd29.html

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_configuration_guide_book09186a008020bd29.html

http://www.cisco.com/univercd/cc/td/doc/product/software/ios124/124cg/hsla_c/index.htm

http://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


12625_04_2006_c2`

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

• Drawings will be held in theWorld 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

Documents

Transcript of Design Voice GW