I N D E X

AAA (Auto Attendant)

SRST (Survivable Remote Site Telephony), 390–391

Tcl application, 442aaa preauth command, 109address identifiers, ISDN, 139–140address resolution, 9address signaling, 115

E&M (Ear and Mouth), 126FXS/FXO, 118–119

advanced busyout (AVBO), 346alias command, 398allow-connections command, 65, 110, 397, 556allow-connections h323 to h323 command, 65allow-connections h323 to sip command, 65allow-connections sip to h323 command, 65analog circuits, 116

E&M (Ear and Mouth), 123address signaling, 126Type I signaling, 124Type II signaling, 124Type III signaling, 124Type IV signaling, 125Type V signaling, 125

FXO (Foreign Exchange Office), 117address signaling, 118–119caller ID, 120–121CAMA (Centralized Automated Message

Accounting), 122–123FXS-DID service, 119informational signaling, 119–120power failover, 117supervisory disconnect, 121–122supervisory signaling, 117–118

FXS (Foreign Exchange Service), 117address signaling, 118–119caller ID, 120–121CAMA (Centralized Automated Message

Accounting), 122–123DID service, 119informational signaling, 119–120supervisory disconnect, 121–122supervisory signaling, 117–118

analog connections, trunk circuits, 146analog phones, call flow, 33–35analog ports, local identifier, 28Analog Telephone Adapter (ATA), 13ANI (automatic number identification), 116ani mapping command, 155application MGCPAPP command, 38application mgcpapp command, 39, 49application session command, 98architectures, IPIPGW (Multiservice IP-to-IP

gateway), 553–554ATA (Analog Telephone Adapter), 13audio files, Tcl, 450AuditConnection (AUCX) command, 31authentication

gatekeeper security, 483IPIPGW (Multiservice IP-to-IP gateway),

562–563authentication username command, 109auto attendant. See AAautomatic number identification (ANI), 116AutoQoS, IP WAN QoS, 248–249AVBO (advanced voice busyout), 346

Bback-to-back user agent (B2BUA), SIP (Session

Initiation Protocol), 81bandwidths

control, 9IP WAN design, 221

Basic ACD, Tcl application, 443battery-reversal command, 121BGP (Border Gateway Protocol), 236billing, IPIPGW (Multiservice IP-to-IP gateway),

568bind command, SIP voice service configuration,

101blocking calls, voice translation profiles, 315–316Border Gateway Protocol (BGP), 236BRI interface, MGCP backhaul configuration, 39busyout action graceful command, 337busyout monitor gatekeeper command, 337

600

CCAC (Call Admission Control), 66, 221, 334, 565

call route selection, 334–336local, 336–337measurement-based, 337–346resource-based, 346–348RSVP, 348–359

gatekeepers, 472–475, 496–498IP WAN design, 221IPIPGW (Multiservice IP-to-IP gateway), 565

carrier ID, 565–566maximum connections, 565resource thresholds, 566–567RSVP, 565

Call Admission Control. See CACcall application alternate default command, 39call application voice command, 72, 445, 448call-block disconnect-cause command, 316call-block translation-profile incoming command,

316call command, H.323 voice class option, 62call control agents

CallManager, 14CME (CallManager Express), 14–15EGW (Enterprise Gateway), 15PBX toll bypass, 15SIP proxy server, 15

call control signaling, optional routing, 9call fallback commands, 344–345call flow

H.323 protocol, 56–58MGCP (Media Gateway Control Protocol), 33

analog phones, 33–35fallback, 37ISDN connections with backhaul, 35–36

SIP (Session Initiation Protocol), 88Cisco CallManager 5, 93–95multiple servers, 92–93proxy server, 90–92two SIP gateways, 88–90

call-forward command, 392–393call forwarding, SRST (Survivable Remote Site

Telephony), 392–393call legs, dial peers, 284–285call-manager-fallback command, 383call preservation, remote sites, 398–399

call route selectionCAC (Call Admission Control), 334–336

local, 336–337measurement-based, 337–346

resource-based, 346–348RSVP, 348–359

case study, 360–362hunt groups, 328

digit manipulation, 330–332huntstop command, 329–330preference command, 328–329

POTS-to-POTS considerations, 359–360TEHO (Tail-End Hop-Off), 333–334troubleshooting gatekeepers, 505–508trunk groups, 332–333

call service stop command, SIP voice service configuration, 101

call signaling, ISDN, 136–137call transfers, SRST (Survivable Remote Site

Telephony)consultative transfers, 392transfer-pattern, 391–392

call treatment cause-code command, 567caller ID, 116

FXS/FXO, 120–121manipulation, 317–318

caller-id enable command, 149, 185calling-info command, SIP UA commands, 98calling line identification (CLID), 116calling number identification (CNID), 116calling privileges, COR (Class of Restrictions)

basics, 365–366case study, 374–378implementation, 367–370inbound call restriction, 372–374list assignment with CallManager Express,

371–372list assignment with SRST, 370–371operation, 366–367

CallManagercall control agents, 14configuration, MGCP (Media Gateway Control

Protocol), 40–46deployment scenarios

multisite with centralized call control, 17

CAC (Call Admission Control)

601

multisite with distributed call control, 17–18

single site deployment, 16gatekeepers, 481, 508–512H.323 protocol, 67–69registration

MGCP (Media Gateway Control Protocol), 31–32

SIP (Session Initiation Protocol), 102–107RSVP, 352–353SIP (Session Initiation Protocol), 93–95SRST (Survivable Remote Site Telephony), 386versus dial peers, 295–298

CallManager ExpressCOR list assignment, 371–372DSPs transcoding, 434–435

CAMA (Centralized Automated Message Accounting), 122, 146

FXS/FXO, 122–123POTS trunks, PSTN (public switched telephone

network), 154–157signaling format, 146

carrier IDs, CAC (call admission control), 565–566

CAS (channel-associated signaling), 129, 206PBX connection, 206ports, 7T1 circuits, 129

cas-custom command, 170case studies

call route selection, 360–362COR (Class of Restrictions), 374–378dial plans, 298–300digit manipulation, 321–323DSPs, 435–437gatekeeper configuration, 536–545H.323 protocol, 73–75introduction, 18–19IPIPGW (Multiservice IP-to-IP gateway),

570–574MGCP (Media Gateway Control Protocol),

configuration, 49–51PBX connection, 213–217PSTN (public switched telephone network)

connections, 179–180SIP (Session Initiation Protocol), 111–113

SRST (Survivable Remote Site Telephony), 405–408

study site descriptions, 20Tcl, 451–454

CC (country code), 140ccm-compatible command, H.323 voice class

option, 62ccm-manager fallback-mgcp command, 404ccm-manager music-on-hold command, 40ccm-manager redundant-host command, 45Centralized Automated Message Accounting.

See CAMAchannel-associated signaling. See CAScircuits

analog, 116E&M (Ear and Mouth), 123–126FXO (Foreign Exchange Office), 117–123FXS (Foreign Exchange Service), 117–123

digital, 126–127E1, 131echo cancellation, 141–142ISDN, 134–141R2 signaling, 131–133T1, 127–131

signaling methods, 115–116Cisco CallManager, 5

call control agents, 14configuration, MGCP (Media Gateway Control

Protocol), 40–46SIP (Session Initiation Protocol), 93–95

Cisco CallManager Express (CME), 14–15Cisco Enterprise Gateway (EGW), call control

agents, 15Cisco gateways, protocols, 6Cisco proprietary method, sending tones, 70Cisco Voice Portal (CVP), 444class maps

classifying traffic, 222Layer 2, 226–227Layer 3, 225–226Layer 4, 222–225

MPLS, 232–239Class of Restrictions. See CORclear-channel mode, 207CLID (calling line identification), 116, 317clock source internal command, 193, 215clock source line command, 192

clock source line command

602

clock source line primary command, 194, 215clocking, digital trunks, 191–195clustering, gatekeeper redundancy, 516–522CME (Cisco CallManager Express), 14–15CNID (calling number identificatiion), 116codec clear-channel command, 210codec command, 343codec complexity command, 419–420codec G711ulaw command, 557codec-interval, 343codec-numpackets, 343codec preference command, 62codec transparent command, 561codec-size, 343codecs

configuring complexity, 419–420DSP voice termination, 418–419

complexitycodec, 419–420flex, 420–421

components, networks, 54–56compression, IP WAN QoS, 246–248conference bridges, DSPs, 423–424conferencing

DSPsconfiguration, 425–427enhanced, 429–432

SRST (Survivable Remote Site Telephony), 391configuration

gatekeepers, 485CAC (Call Admission Control), 496–498CallManager, 508–512case study, 536–545dial peers, 490–493directory gatekeepers, 500–502dynamic prefix registration, 494–496gateways, 487–489multiple configurations, 498–500redundancy, 513–527security, 528–536technology prefix, 489troubleshooting, 503–508zone prefix, 490–493zones, 486–487

IPIPGW (Multiservice IP-to-IP gateway), 556–560

connection plar command, 149–151

connectionsIP WANs, 266

business application, 219–220design, 220–221fax services, 249–258modems, 249–250, 258–260QoS (quality of service), 221–249security, 260–266

MGCP (Media Gateway Control Protocol), 28PSTN (public switched telephone network)

case study, 179–180digital trunks, 158–178POTS trunks, 147–157trunk circuits, 145–147

controller E1 command, 159controller T1 command, 159COR (Class of Restrictions), 365

basics, 365–366case study, 374–378implementation, 367–370inbound call restriction, 372–374list assignment with CallManager Express,

371–372list assignment with SRST, 370–371operation, 366–367

corlist incoming command, 370corlist outgoing command, 369country code (CC), 140cptone command, 120cptone GB command, 169cptone locale command, 185CreateConnection (CRCX) command, 30csim start command, 73CVP (Cisco Voice Portal), Tcl application, 444

Ddatabase url command, 400debug cch323 h225 command, 73debug cch323 h245 command, 73debug ccsip command, 111debug commands, IPIPGW (Multiservice IP-to-

IP gateway), 569–570debug ephone command, 405debug fax relay t30 command, 255debug gatekeeper gup asn1 command, 521–522

clock source line primary command

603

debug h225 asn1 command, 506–507debug h225 command, 73debug h245 command, 73debug ip tcp transaction command, 73debug isdn q921 command, 199–200debug isdn q931 command, 200debug voice ccapi inout command, 405debug voice dial peer command, 405debug voice translation command, 316debug voip ccapi inout command, 73debug voip vtsp session command, 190debug vpm all command, 150, 154–156debug vpm signal command, 172, 190debug vpm signaling, 190debug vtsp all command, 172debugging

gatekeepers, 482–483configuration, 528–530troubleshooting, 530–536

gatewaysH.323 protocol, 72MGCP (Media Gateway Control

Protocol), 47IP WANs, 260–261

design, 221firewalls, 265–266NAT, 265V3PN, 264–265voice media, 261–264

IPIPGW (Multiservice IP-to-IP gateway)H.323 network deployment, 562HTTP digest authentication, 562–563TLS (Transport Layer Security), 562

SIP (Session Initiation Protocol), 109–110SRST (Survivable Remote Site Telephony),

399–403delay-start operation, address signaling, 126DeleteConnection (DLCX) command, 30deployment

CallManager scenariosmultisite with centralized call control, 17multisite with distributed call control,

17–18single site deployment, 16

gatekeepers, 476directory gatekeeper, 479

RAI (Resource Availability Indicator), 478–479

redundancy, 476–478designs, IP WANs, 220–221dest-ipaddr command, 343dest-port command, 343dial-peer cor custom name label command, 367dial peer hunt command, 293, 328dial peers, 283–284

call matchinginbound, 285–287operation status, 294–295outbound, 288–289outbound targets, 291–293POTS versus VoIP, 293–294verification, 289–290versus CallManager, 295–298

configuration, SIP (Session Initiation Protocol), 98

gatekeeper configuration, 490–493gateways, 477H.323 configuration, 62–63inbound versus outbound, 284–285virtual, 384–385

dial plans, 275case study, 298–300dial peer matching, 283–284

dial peer verification, 289–290inbound, 284–287operation status, 294–295outbound, 288–289outbound targets, 291–293POTS versus VoIP, 293–294versus CallManager, 295–298

distribution, 283H.323 protocol, 60–61MGCP (Media Gateway Control Protocol), 37number plans, 276

private, 276–277PSTN, 278–281

overlapping number plans, 281–282scalable, 283SIP (Session Initiation Protocol), 96–97SRST (Survivable Remote Site Telephony), 387

configuring patterns, 389–390direct extension dialing, 389H.323 gateways, 388MGCP gateways, 388–389

dial plans

604

dialed number identification service (DNIS), 116dialplan-pattern command, 390, 398DID (direct inward dialing)

FXS (Foreign Exchange Service), 119POTS trunk, PSTN (public switched telephone

network), 152–154signaling format, 146

digest authentication, IPIPGW (Multiservice IP-to-IP gateway), 562–563

digit manipulationbasics, 303–304

prefix, 306stripping, 304–305

caller ID, 317–318case study, 321–323forward digits, 305–306hunt groups, 330–332number expansion, 306–307operation order, 318–320troubleshooting, 320–321voice translation profiles, 307–308

application, 314–315blocking calls, 315–316creating, 314creating voice rules, 308–309regular expressions, 309–314testing rules, 316

digit-strip command, 305digital circuits, 126–127

E1, 131echo cancellation, 141–142ISDN, 134

address identifiers, 139–140call signaling, 136–137IE (Information Elements), 138–139NFAS (Non-Facility Associated

Signaling), 140QSIG (Q Signaling), 141switch configuration, 134–136

R2 signaling, 131inter-register signaling, 132–133line signaling, 131–132

T1, 127–129CAS (channel-associated signaling), 129E&M signaling, 129–130FGD (feature group D), 130–131

digital connections, trunk circuits, 147

digital interfaces, local identifier, 29digital signal level zero, 127Digital Signal Processors. See DSPsdigital trunks

PBX connectionE1/T1 physical layer connections,

190–196ISDN PRI trunks, 196–205tie trunks, 206

PSTN (public switched telephone network), 158E1 R2 trunks, 168–172E1/T1 connection, 158–161ISDN BRI trunks, 175–178ISDN PRI trunks, 162–168T1 CAS trunks, 172–175

direct extension dialing, SRST (Survivable Remote Site Telephony), 389

direct-inward-dial command, 288–289direct inward dialing (DID), FXS (Foreign

Exchange Service), 119directory gatekeepers, 479, 500–502directory services, PSTN numbering plans, 278DNIS (dialed number identification service), 116downloads, Tcl scripts, 444–445ds0-group command, 209dscp-based command, 230dspfarm command, 424DSPs (digital signal processors), 16, 411

CallManager Express transcoding, 434–435case study, 435–437configuring resources, 424–425

enhanced transcoding and conferencing, 429–432

transcoding and conferencing configuration, 425–427

function in gateways, 411–412resource requirement determination, 412

conference bridge, 423–424sharing, 421–422transcoding, 422–423various models, 413–414voice termination, 414–421

DTMF (Dual-tone Multifrequency), 283inter-register signaling, 133IPIPGW (Multiservice IP-to-IP gateway),

563–564

dialed number identification service (DNIS)

605

dtmf h245-alphanumeric command, 557DTMF Relay

configuration, MGCP (Media Gateway Control Protocol), 46

H.323 protocol, 70–71SIP (Session Initiation Protocol), 107–109

dtmf-relay command, 71, 108dtmf-relay rtp-nte command, 557Dual-tone Multifrequency. See DTMFdynamic prefixes, registration, 494–496

EE&M (Ear and Mouth), 116

analog circuits, 123address signaling, 126Type I signaling, 124Type II signaling, 124Type III signaling, 124Type IV signaling, 125Type V signaling, 125

ports, 7signaling

format, 146T1 circuits, 129–130

trunks, PBX connection configuration, 187–190e&m-delay-dial signaling, 130e&m-fgd signaling, 130e&m-immediate-start signaling, 130e&m-wink-start signaling, 130E.164 number resolution, gatekeepers, 469–470

call routing process, 472technology prefixes, 471zone prefixes, 470–471

E1 circuits, 131E1 connections, digital trunks, 190–191

clocking, 191–195verification, 195–196

E1 digital circuit, digital trunks, 158–161E1 R2, signaling format, 147E1 R2 trunks, PSTN (public switched telephone

network) connection, 168–171MGCP, 171verification, 171–172

Ear and Mouth. See E&Mecho cancellation, digital circuits, 141–142

EEM (Embedded Event Manager), Tcl application, 444

EGW (Enterprise Gateway), 15Embedded Event Manager (EEM), 444emergency services, PSTN numbering plans, 278encoding command, H.323 voice class option, 62EndpointConfiguration (EPCF) command, 31endpoints, MGCP (Media Gateway Control

Protocol), 28Enterprise Gateway (EGW), 15events, MGCP (Media Gateway Control

Protocol), 28EVM-HD-8FXS/DID model, 153extension dialing, SRST (Survivable Remote Site

Telephony), 389

Ffair-queue command, 230fallback

call flow, MGCP (Media Gateway Control Protocol), 37

MGCP, 39, 403–404SRST (Survivable Remote Site Telephony),

382–383Fast Start, H.323 protocol, 56, 58fault tolerances, 283Fax Detect script, Tcl application, 443fax-relay ecm disable command, 251fax-relay sg3-to-g3 command, 255faxes

IP WANs, 249–255Cisco fax relay, 251–252passthrough configuration, 256–258

IPIPGW (Multiservice IP-to-IP gateway), 564feature group D (FGD), T1 circuits, 130–131FGD (feature group D), T1 circuits, 130–131fgd-eana signaling, 130fgd-os signaling, 130firewalls, IP WAN security, 265–266flex complexity, voice termination, 420–421Foreign Exchange Office. See FXOForeign Exchange Station. See FXSforward digits, manipulation, 305–306forward-digits 4 command, 67

forward-digits 4 command

606

forward-digits command, 305forward-digits extra inband command, 395forwarding calls, SRST (Survivable Remote Site

Telephony), 392–393frame forwarding mode, 207frame relay

FRF.12 fragmentation, 242–244MLPoFR fragmentation, 244–245

framing command, 159FRF.12 fragmentation, 242–244FXO (Foreign Exchange Office), 116, 146

analog circuits, 117address signaling, 118–119caller ID, 120–121CAMA (Centralized Automated Message

Accounting), 122–123FXS-DID service, 119informational signaling, 119–120power failover, 117supervisory disconnect, 121–122supervisory signaling, 117–118

POTS trunk, PSTN (public switched telephone network), 147–151

signaling format, 146FXO ports (Foreign Exchange Office ports), 7,

184–186FXS (Foreign Exchange Station), 116, 146

analog circuits, 117address signaling, 118–119caller ID, 120–121CAMA (Centralized Automated Message

Accounting), 122–123DID service, 119informational signaling, 119–120supervisory disconnect, 121–122supervisory signaling, 117–118

signaling format, 146FXS ports (Foreign Exchange Station ports), 7,

184–186

Ggatekeeper command, 486Gatekeeper Transaction Message Protocol

(GKTMP), 469

Gatekeeper Update Protocol (GUP), 468–469gatekeepers

basics, 459–461CAC (Call Admission Control), 472–475CallManager, 481CallManager deployment scenarios

multisite with centralized call control, 17multisite with distributed call control,

17–18single site deployment, 16

clustering, 478configuration, 485

CAC (Call Admission Control), 496–498CallManager, 508–512case study, 536–545dial peers, 490–493directory gatekeepers, 500–502dynamic prefix registration, 494–496gateways, 487–489multiple configurations, 498–500redundancy, 513–527security, 528–536technology prefix, 489troubleshooting, 503–508zone prefix, 490–493zones, 486–487

deployment models, 476directory gatekeeper, 479RAI (Resource Availability Indicator),

478–479redundancy, 476–478

E.164 number resolution, 469–470call routing process, 472technology prefixes, 471zone prefixes, 470–471

H.323 protocol, 12, 55role, 8–9security, 482–483signaling, 461

GKTMP (Gatekeeper Transaction Message Protocol), 469

GUP (Gatekeeper Update Protocol), 468–469

RAS signaling, 461–468tokenless call authentication, 483

gateway command, 487

forward-digits command

607

gatewayscall control agents

CallManager, 14CME (CallManager Express), 14–15EGW (Enterprise Gateway), 15PBX toll bypass, 15SIP proxy server, 15

CallManager deployment scenariosmultisite with centralized call control, 17multisite with distributed call control,

17–18single site deployment, 16

dial peers, 477gatekeeper configuration, 487–489H.323 protocol, 12, 61

CallManager, 67–69case study, 73–75DTMF Relay, 70–71network components, 55redundancy, 69–70security, 72toll bypass, 65–67troubleshooting, 72–73voice class configuration, 62–64voice service commands, 64–65

MGCP (Media Gateway Control Protocol)CallManager configuration, 40–42, 45–46configuration, 38configuring fallback, 39DTMF Relay configuration, 46music on hold enabling, 40PRI and BRI backhaul configuration, 39security, 47source IP address assignment, 39

role, 5–7SIP (Session Initiation Protocol)

call flow, 88–90CallManager registration, 102–107dial peer configuration, 98DTMF Relay, 107–109security, 109–110SIP UA configuration, 98–100toll bypass, 102voice service configuration, 101–102

SRST (Survivable Remote Site Telephony), 383–385

types, 7routers, 7standalone voice gateways, 7–8switch modules, 8

GKTMP (Gatekeeper Transaction Message Protocol), 469

ground-start signaling, 118group command, 109GUP (Gatekeeper Update Protocol), 468–469

HH.225 specification, H.323 protocol, 53H.235 specification, H.323 protocol, 54H.245 Alphanumeric method, sending tones, 70H.245 Signal method, sending tones, 70H.245 specification, H.323 protocol, 54H.261 specification, H.323 protocol, 54H.263 specification, H.323 protocol, 54H.320 specification, H.323 protocol, 54H.323 networks

gatekeeper signaling, 466–468IPIPGW (Multiserver IP-to-IP gateway), 555,

562H.323 protocol, 6, 12, 53, 61

call flow, 56–58CallManager, 67–69case study, 73–75cons, 59dial plan, 60–61DTMF Relay, 70–71fax services passthrough, 257–258gateway security, 72network components, 54

gatekeepers, 55gateways, 55MCU (multipoint control units), 56proxy servers, 56terminals, 55

pros, 58–59redundancy, 69–70SIP (Session Initiation Protocol) connections,

110specifications, 53–54SRST (Survivable Remote Site Telephony), 388toll bypass, 65–67

H.323 protocol

608

troubleshooting, 72–73voice class configuration, 62–64voice service commands, 64–65when to use, 59–60

H.450 specification, H.323 protocol, 54H225 command, H.323 voice class option, 62H245 command, H.323 voice class option, 62h323-gateway voip bind srcaddr command, 68h323-gateway voip interface command, 487hierarchical gatekeeper implementation, 479hierarchical numbering plans, 283Hot Standby Routing Protocol. See HSRPHSRP (Hot Standby Routing Protocol ), 477,

513–516HTTP, digest authentication, 562–563hunt groups, call route selection, 328

digit manipulation, 330–332huntstop command, 329–330preference command, 328–329

huntstop channel command, 391huntstop command, hunt groups, 329–330

IIE (Information Elements), ISDN, 138–139immediate-start operation, address signaling, 126inbound calls, COR restriction, 372–374inbound dial peers, 284–287incoming called-number command, 63Information Elements (IE), ISDN, 138–139information services, PSTN numbering plans,

278informational signaling, 116, 119–120interface bri command, 176interleave, IP WAN QoS, 239–240

frame relay FRF.12 fragmentation, 242–244frame relay MLPoFR, 244–245MLPoATM, 245–246multilink PPP, 240–242

international calls, PSTN numbering plans, 278Internet Protocol Device Control (IPDC), 25Internet Telephony Service Provider. See ITSPinter-register signaling, R2 signaling, 132–133IP addresses

IPIPGW (Multiserver IP-to-IP gateway), 561MGCP source assignment, 39

ip sla responder command, 341ip source address command, 402IP Telephony Service Providers (ITSP), 10, 80IP WANs, 266

business application, 219–220design, 220–221fax services, 249–251, 254–255

Cisco fax relay, 251–252passthrough configuration, 256–258T.38 for MGCP gateways, 252

modems, 249–250, 258modem relay, 258–259passthrough, 259–260

QoS (quality of service), 221–222AutoQoS, 248–249classifying traffic using class maps,

222–227compression, 246–248link fragmentation and interleave,

239–246MPLS class mapping, 232–239policy maps, 227–232

security, 260–261firewalls, 265–266NAT, 265V3PN, 264–265voice media, 261–264

IPDC (Internet Protocol Device Control), 25IPIPGW (Multiservice IP-to-IP gateway), 551

basics, 551–552architecture, 553–554H.323 networks, 555media-handling modes, 554–555protocol interworking, 556SIP networks, 555

case study, 570–574configuration, 556–560supported features, 560

billing, 568CAC (call admission control), 565–567debug commands, 569–570DTMF interworking, 563–564faxes, 564hidden IP addresses, 561QoS (quality of service), 565security, 562–563show commands, 569

H.323 protocol

609

Tcl scripts, 568transcoding, 568video support, 560–561VSML (voice XML), 568

IPsec, IP WAN security, 263IP-to-IP gateways, role, 10–11ISDN

digital circuits, 134address identifiers, 139–140call signaling, 136–137IE (Information Elements), 138–139NFAS (Non-Facility Associated

Signaling), 140QSIG (Q Signaling), 141switch configuration, 134–136

PRI trunks, 196–198MGCP, 201–202QSIG protocol, 204–205troubleshooting, 203–204verification, 198–201

isdn bchan-number-order command, 163isdn bind-l3 ccm-manager command, 39, 165, 201ISDN BRI, signaling format, 147ISDN BRI ports, 7ISDN BRI trunks, PSTN (public switched

telephone network) connection, 175–177MGCP, 177–178verification, 178

ISDN connections, call flow, 35–36isdn incoming-voice voice command, 177ISDN PRI, signaling format, 147ISDN PRI trunks, PSTN (public switched

telephone network), 162–164MGCP, 164–165verification, 165–168

isdn protocol-emulate command, 135isdn protocol-emulate network command, 196isdn spid1 command, 176isdn spid2 command, 177isdn switch-type command, 135, 162, 176isdn switch-type primary-qsig command, 205ITSP (Internet Telephony Service Provider),

10, 80

J–Llimit-dn command, 385line signaling, R2 signaling, 131–132linecode command, 159link fragmentation, IP WAN QoS, 239–240

frame relay FRF.12, 242–244frame relay MLPoFR, 244–245MLPoATM, 245–246multilink PPP, 240–242

load balancing, gatekeeper redundancy, 520–521local CAC, call route selection, 336

DS0 limitations, 336LVBO (local voice busyout), 337maximum connections, 336

local calls, PSTN numbering plans, 278local voice busyout (LVBO), 337location servers, SIP (Session Initiation Protocol),

81long distance calls, PSTN numbering plans, 278loop-start signaling, 118lrq forward-queries command, 501lrq reject-resource-low command, 479LVBO (local voice busyout), 337

MMalicious Call ID (MCID), 443match ip rtp 16383 2000 command, 224max-conferences command, 391max-conn command, 336max-dn command, 384max-ephones command, 384max-forwards command, SIP UA commands, 98MCID (Malicious Call ID), Tcl application, 443MCUs (multipoint control units), 12

H.323 protocol, 12network components, H.323 protocol, 56

measurement-based CAC, 337AVBO (advanced voice busyout), 346configuring IP SLA, 340–342IP Service Assurance Agent and Response Time

Reporter, 342–344IP SLA network monitoring, 338probable voice quality estimates, 338–340PSTN fallback, 344–346

measurement-based CAC

610

media, IPIPGW (Multiservice IP-to-IP gateway), 554–555

media flow-through|flow-around command, 556Media Gateway Control Protocol. See MGCPmedia termination point (MTP), 16messages

MGCP (Media Gateway Control Protocol), 29–31

SIP (Session Initiation Protocol), 82–88metacharacters, 309MGCP (Media Gateway Control Protocol), 6, 11,

25–26basics, 26–27

cons, 28pros, 27

call flow, 33analog phones, 33–35fallback, 37ISDN connections with backhaul, 35–36

CallManager configuration, 40–42, 45–46case study, 49–51configuring fallback, 39dial plan, 37DTMF relay configuration, 46E1 R2 trunks, 171fallback, 403–404fax services passthrough, 256–257FXO and FXS ports, 186FXO POTS trunk connection, 151gateway configuration, 38ISDN BRI trunks, 177–178ISDN PRI trunk connection, 164–165ISDN PRI trunks, 201–202music on hold enabling, 40operation, 28–29

CallManager registration, 31–32messages, 29–31

PRI and BRI backhaul configuration, 39PSTN (public switched telephone network),

FXO POTS trunk connection, 150–151securing gateways, 47source IP address assignment, 39SRST (Survivable Remote Site Telephony),

388–389T1 CAS trunks, 174troubleshooting, 47–49

mgcp call-agent command, 45, 201

mgcp command, 38mgcp modem passthrough command, 256, 259mgcp modem passthrough mode command, 256mgcp modem passthrough voip redundancy

command, 256mgcp package-capability command, 26mgcp tse payload command, 256MIME (Multipurpose Internet Mail Extension),

80min-se command, SIP voice service

configuration, 101MLPoATM, link fragmentation and interleave,

245–246mobiles, PSTN numbering plans, 278modems, IP WANs, 249–250, 258

modem relay, 258–259passthrough, 259–260

ModifyConnection (MDCX) command, 30modules, switch gateways, 8moh filename command, 396moh-live command, 396MPLS (Multiprotocol Label Switching),

18, 232–239MTP (media termination point), 16MTP resources, transcoding, 422–423multicast discovery, 465multilink PPP, link fragmentation and interleave,

240–242multipoint control units. See MCUsMultiprotocol Label Switching (MPLS),

18, 232–239Multipurpose Internet Mail Extension (MIME),

80Multiservice IP-to-IP gateway. See IPIPGWmusic on hold

enabling, MGCP (Media Gateway Control Protocol), 40

SRST (Survivable Remote Site Telephony), 396mutual authentication, 562

Nnamespaces, Tcl gateway configuration, 446–448NANP (North American Numbering Plan), 279NAT, IP WAN security, 265nat command, SIP UA commands, 99

media, IPIPGW (Multiservice IP-to-IP gateway)

611

National (Significant) Number (N(S)N), 140National Destination Code (NDC), 140NDC (National Destination Code), 140network-clock-participate command, 192network-clock-participate wic 2 command, 416network-clock-select command, 192networks

components, H.323, 54–56VoIP. See VoIP networks

NFAS (Non-Facility Associated Signaling), 140no auto grant command, 401no call service stop command, 65no mgcp command, 38Non-Facility Associated Signaling (NFAS), 140North American Numbering Plan (NANP), 279NPI (number plan identification), ISDN address

identifiers, 139–140number plan identification (NPI), ISDN address

identifiers, 139–140numbering plans, 276

overlapping, 281–282private, 276–277PSTN, 278–279

NANP (North American Numbering Plan), 279

UK NNP (UK National Numbering Plan), 279–281

numbers, expansion, 306–307num-exp command, 307

Ooptional call management, 9optional routing, call control signaling, 9optional security, 9Ousterhout, Dr. John, Tcl script, 439outbound dial peers, 284–285, 288–289overlapping numbering plans, 281–282

Pparamspace command, 447passthrough method, 249

fax services configurationH.323, 257–258

IP WAN, 256–257SIP, 257–258

modem services, 259–260path selection

CAC (Call Admission Control), 334–336local, 336–337measurement-based, 337–346resource-based, 346–348RSVP, 348–359

case study, 360–362hunt groups, 328

digit manipulation, 330–332huntstop command, 329–330preference command, 328–329

POTS-to-POTS considerations, 359–360TEHO (Tail-End Hop-Off), 333–334trunk groups, 332–333

pattern command, 395PBX (private branch exchange), 5

case study, 213–217digital trunks

E1/T1 physical layer connections, 190–196

ISDN PRI trunks, 196–205tie trunks, 206

POTS (plain old telephone service), 184configuring FXO and FXS port

connections, 184–186E&M trunk connection, 187–190

T-CCS (transparent common channel signaling), 207–213

toll bypass, 15physical connections, digital trunks, 190–191

clocking, 191–195verification, 195–196

plain old telephone service. See POTSpolicy maps, IP WAN QoS, 227–232post dial delays, 283POTS (plain old telephone service), 184

call route selection, 359–360dial peers, 293–294PBX connection, 184

configuring FXO and FXS port connections, 184–186

E&M trunk connection, 187–190

POTS (plain old telephone service)

612

POTS trunks, PSTN (public switched telephone network), 147

CAMA connections, 154–157DID connection configuration, 152–154FXO connection configuration, 147–151

preference command, hunt groups, 328–329prefix command, 306prefix digits, manipulation, 306premium calls, PSTN numbering plans, 278presence servers, SIP (Session Initiation

Protocol), 13, 81PRI interface, MGCP backhaul configuration, 39PRI trunks, digital trunks, 196–198

MGCP, 201–202QSIG protocol, 204–205troubleshooting, 203–204verification, 198–201

pri-group command, 135, 163pri-group timeslots command, 39priority command, 229private branch exchange. See PBXprivate numbering plans, 276–277protocols

H.323, 12, 53, 61call flow, 56–58CallManager, 67–69case study, 73–75cons, 59dial plan, 60–61DTMF Relay, 70–71gateway security, 72network components, 54–56pros, 58–59redundancy, 69–70specifications, 53–54toll bypass, 65–67troubleshooting, 72–73voice class configuration, 62–64voice service commands, 64–65when to use, 59–60

IPIPGW (Multiservice IP-to-IP gateway), 555–556

MGCP (Media Gateway Control Protocol), 11, 25–26

basics, 26–28call flow, 33–37CallManager configuration, 40–42, 45–46

case study, 49–51configuring fallback, 39dial plan, 37DTMF Relay configuration, 46gateway configuration, 38music on hold enabling, 40operation, 28–32PRI and BRI backhaul configuration, 39securing gateways, 47source IP address assignment, 39troubleshooting, 47–49

RTP (Real-Time Transport Protocol), 14SCCP (skinny client control protocol), 13SIP (Session Initiation Protocol), 12–13, 79

allowing H.323 connections, 110basics, 79–88call flow, 88–95CallManager registration, 102–107case study, 111–113cons, 96dial peer configuration, 98dial plan, 96–97DTMF Relay, 107–109pros, 95security, 109–110SIP UA configuration, 98–100toll bypass, 102troubleshooting, 110–111voice service configuration, 101–102when to use, 96

proxy command, 398proxy servers

network components, H.323 protocol, 56SIP (Session Initiation Protocol), 13

call flow, 90–92functional components, 81

PSTN (public switched telephone network), 5, 145

case study, 179–180digital trunks, 158

E1 R2 trunks, 168–172E1/T1 connection, 158–161ISDN BRI trunks, 175–178ISDN PRI trunks, 162–168T1 CAS trunks, 172–175

fallback, measurement-based CAC, 344–346numbering plans, 278–281

POTS trunks, PSTN (public switched telephone network)

613

POTS trunks, 147CAMA connections, 154–157FXO connection configuration, 147–154

trunk circuits, 145–147public switched telephone network. See PSTNPVDM versus PVDM2, 414PVDM2 versus PVDM, 414

QQ Signaling. See QSIGQoS (quality of service), 221

IP WAN design, 221IP WANs, 221–222

AutoQoS, 248–249classify traffic using class maps, 222–227compression, 246–248link fragmentation and interleave,

239–246MPLS class mapping, 232–239policy maps, 227–232

IPIPGW (Multiservice IP-to-IP gateway), 565QSIG (Q Signaling), 141

ISDN digital circuits, 141PRI trunks, 204–205

quality of service. See QoS

RR2 signaling, 131

inter-register signaling, 132–133line signaling, 131–132

R2-Compelled inter-register signaling, 133R2-Noncompelled inter-register signaling, 133R2-Semi-compelled inter-register signaling, 133RAI (Resource Availability Indication), 348, 478

gatekeeper deployment models, 478–479gatekeeper redundancy, 522–527resource-based CAC, 348

random-detect command, 230random-detect dscp-based command, 230RAS signaling, gatekeeper signaling, 461–464

discovery, 464–465H.323 call flow, 466–468

Real-Time Transport Control Protocol (RTCP), 14

Real-Time Transport Protocol (RTP), 6, 14redirect contact order command, SIP voice

service configuration, 101redirect ip2ip command, 98, 101redirect servers, SIP (session initiation protocol),

13, 81redundancy

CallManager, MGCP (Media Gateway Control Protocol), 45–46

gatekeeper configurationclustering, 516–520HSRP (Hot Standby Routing Protocol),

513–516load balancing, 520–521RAI (Resource Availability Indication),

522–527troubleshooting gatekeeper cluster,

521–522gatekeeper deployment models, 476–478H.323 protocol, 69–70

register servers, SIP (Session Initiation Protocol), 81

registrar command, SIP UA commands, 99registrar server command, 101, 397registrar servers, SIP (session initiation protocol),

13registration, troubleshooting gatekeepers,

503–505regular expressions, voice translation profiles,

309changing call type or numbering plan, 313–314deleting specific digits, 312rerouting call over PSTN, 310–312sets and replacement digits, 312–313

relay method, 249remote sites

call preservation, 398–399MGCP gateway fallback, 403–404SRST (Survivable Remote Site Telephony).

See SRSTresidential gateways, 26Resource Availability Indication. See RAIresource-based CAC, call route selection, 346

gatekeeper zone bandwidth, 347

resource-based CAC, call route selection

614

local system resources, 346–347RAI (Resource Availability Indication), 348

Response Time Reporter (RTR), 342–344restoral time, SRTS (Survivable Remote Site

Telephony), 383retry command, SIP UA commands, 99ring number command, 148routers, gateway type, 7RSVP

call route selection, 348–349agent configuration, 354–358CallManager controlled, 352–353configuring gateway controlled RSVP,

351–352gateway controlled RSVP, 349–350IntServ/DiffServ model, 358voice network incorporation, 358–359

IPIPGW (Multiservice IP-to-IP gateway), 565RTCP (Real-Time Transport Control Protocol),

14RTP (Real-Time Transport Protocol), 6, 14RTP-NTE method, sending tones, 70RTR (Response Time Reporter), 342–344

SSAA (Service Assurance Agent), 342–344SBC (Session Border Controller), 551

IPIPGW (Multiservice IP-to-IP gateway)basics, 551–556case study, 570–574configuration, 556–560supported features, 560–570

scalable dial plans, 283SCCP (Skinny Client Control Protocol), 6, 13scripts

Tcl application, 446Tcl download, 444–445Tcl implementation, 448–450

secondary-dialtone digit command, 390secure RTP, IP WAN security, 261–263Secure versions of RTP (SRTP), 14security

gatekeepers, 482–483configuration, 528–530troubleshooting, 530–536

gatewaysH.323 protocol, 72MGCP (Media Gateway Control

Protocol), 47IP WANs, 260–261

design, 221firewalls, 265–266NAT, 265V3PN, 264–265voice media, 261–264

IPIPGW (Multiservice IP-to-IP gateway)H.323 network deployment, 562HTTP digest authentication, 562–563TLS (Transport Layer Security), 562

SIP (Session Initiation Protocol), 109–110SRST (Survivable Remote Site Telephony),

399–403security password level command, 72security token required-for registration

command, 72serialization delay, 239Service Assurance Agent (SAA), 342–344service mgcp command, 39service-policy command, 222service voip configuration mode, 556Session Border Controller. See SBCSession Initiation Protocol. See SIPSession protocol command, 557session protocol sipv2 command, 557session target command, 557session target ras command, 490session transport command, 98, 101SGCP (Simple Gateway Control Protocol), 25sharing DSPs, 421–422show atm vc command, 246show ccm-manager backhaul command, 39show ccm-manager command, 47–48, 203–204show class-map command, 226, 232show commands, 519, 569show controller command, 195–196show controller e1 command, 161show dial-peer voice command, 384show dial-peer voice summary command, 72show dialplan number command, 73, 307show ephone command, 405show frame-relay fragment command, 244show gatekeeper calls command, 493

resource-based CAC, call route selection

615

show gatekeeper endpoints command, 488, 503show gatekeeper zone prefix command, 490show gatekeeper zone status command, 497show h323 gateway command, 73show interface multilink 1 command, 241show ip sla statistics command, 341show isdn status command, 39, 135, 178, 198–199,

203show mgcp command, 49, 253show mgcp endpoint command, 49show network-clocks command, 191show policy interface command,

230–232, 243–244show policy-map command, 232show policy-map interface command, 246show queuing command, 232show run command, 71show sip-ua command, 110show sip-ua statistics command, 111show sip-ua status command, 100–102show voice call summary command, 171, 175, 213show voice dsp voice command, 415–417show voice port command,

49, 149, 154–156, 188–189show voice port summary command, 149, 154,

171, 174, 185, 213show voice translation-profile command, 316show voice translation-rule command, 316signal cama command, 155signal did command, 153signal encryption, 562signal type subcommand, 185signaling

circuits, 115–116gatekeepers, 461

GKTMP (Gatekeeper Transaction Message Protocol), 469

GUP (Gatekeeper Update Protocol), 468–469

RAS, 461–468sides, 187

signals, MGCP (Media Gateway Control Protocol), 28

Simple Gateway Control Protocol (SGCP), 25SIP (Session Initiation Protocol), 6, 12–13, 79

allowing H.323 connections, 110basics, 79–81

functional components, 81–82messages, 82–88

call flow, 88Cisco CallManager 5, 93–95multiple servers, 92–93proxy server, 90–92two SIP gateways, 88–90

CallManager registration, 102–107case study, 111–113cons, 96dial peer configuration, 98dial plan, 96–97DTMF Relay, 107–109fax services passthrough, 257–258pros, 95proxy server, call control agents, 15security, 109–110SIP UA configuration, 98–100toll bypass, 102troubleshooting, 110–111voice service configuration, 101–102when to use, 96

SIP ACK message, 88sip command, 101SIP INVITE message, 85–86SIP networks, IPIPGW (Multiservice IP-to-IP

gateway), 555SIP OK responses, 87–88SIP Ringing responses, 87SIP SDP message, 86sip-server command, SIP UA commands, 99SIP SRST gateways, 397

SIP registrar server configuration, 397voice register pool, 397–398

SIP Trying responses, 87sip-ua command, 98SIP UA configuration, SIP (Session Initiation

Protocol), 98–100sip-ua configuration mode, 563Skinny Client Control Protocol (SCCP), 6, 13SN (Subscriber Number), 140source-ipaddr command, 343source-port, 343specifications, H.323 protocol, 53–54SRST (Survivable Remote Site Telephony), 27,

381basics, 381–382

fallback, 382–383restoral time, 383

case study, 405–408

SRST (Survivable Remote Site Telephony)

616

configurationCallManager, 386gateway, 383–385

COR list assignment, 370–371dial plan, 387

configuring patterns, 389–390direct extension dialing, 389H.323 gateways, 388MGCP gateways, 388–389

featuresAA (Auto Attendant), 390–391call transfers, 391–392conferencing, 391forwarding calls, 392–393maximum line appearances, 391music on hold, 396voice-mail, 393–396

security, 399–403SIP SRST gateway, 397

SIP registrar server configuration, 397voice register pool, 397–398

troubleshooting, 404–405SRTP (Secure versions of RTP), 14standalone voice gateways, 7–8station-id command, 121station ID commands, caller ID manipulation,

318station-side connections, 184stripping, digit manipulation, 304–305Subscriber Number (SN), 140subscription maximum command, SIP voice

service configuration, 101supervisor disconnect command, 150supervisory disconnect, FXS/FXO, 121–122supervisory signaling

circuits, 115FXS/FXO, 117–118

Survivable Remote Site Telephony. See SRSTswitches, gateway modules, 8

TT.37 Store and Forward Fax, Tcl application, 443T.120 specification, H.323 protocol, 54T1 CAS

signaling format, 147

trunks, PSTN (public switched telephone network) connection, 172–175

T1 circuits, 127–129CAS (channel-associated signaling), 129E&M signaling, 129–130FGD (feature group D), 130–131

T1 connections, digital trunks, 190–191clocking, 191–195verification, 195–196

T1 digital circuits, digital trunks, 158–161T1/E1 PRI ports, 7T-CCS (transparent common channel signaling),

207–213Tail-End Hop-Off (TEHO), 333–334Tcl (Toolkit Command Language), 439

applicationsAA (auto attendant), 442Basic ACD, 443CVP (Cisco Voice Portal), 444EEM (Embedded Event Manager), 444Fax Detect script, 443MCID (Malicious Call ID), 443T.37 Store and Forward Fax, 443

audio files, 450basics, 439–442case study, 451–454configuring gateway, 445

initializing and specifying parameters, 445

packages and parameter namespaces, 446–448

script application, 446downloading scripts from Cisco, 444–445restrictions, 451script implementation, 448–450

Tcl scripts (Toolkit Command Language scripts), 568

technology prefixesconfiguration, 489E.164 number resolution, 471

TEHO (Tail-End Hop-Off), 333–334telephony-service command, H.323 voice class

option, 62terminals, H.323 protocol, 12, 55–56test voice translation-rule command, 316

SRST (Survivable Remote Site Telephony)

617

timers command, SIP UA commands, 99TLS

IP WAN security, 263–264IPIPGW (Multiservice IP-to-IP gateway)

security, 562toll bypass

configuration, SIP (Session Initiation Protocol), 102

H.323 protocol, 65–67PBX, 15

toll calls, PSTN numbering plans, 278toll-free calls, PSTN numbering plans, 278TON (type of number), ISDN address identifiers,

139–140Toolkit Command Language. See TclToolkit Command Language scripts (Tcl scripts),

568traffic, classifying using class maps, 222

Layer 2, 226–227Layer 3, 225–226Layer 4, 222–225

transcoding, 411DSPs

CallManager Express, 434–435configuration, 425–427enhanced, 429–432

IPIPGW (Multiservice IP-to-IP gateway), 568transfer-pattern command, 391–392transfer-system command, 392translation-profile command, 315translation profiles, voice, 307–308

application, 314–315blocking calls, 315–316creating, 314creating voice rules, 308–309regular expressions, 309–314testing rules, 316

translation rules, 307transparent common channel signaling (T-CCS),

207–213transport switch udp tcp command, SIP voice

service configuration, 101troubleshooting

digit manipulation, 320–321gatekeeper clustering, 521–522gatekeeper security, 530–536

gatekeepers, 503call routing issues, 505–508registration issues, 503–505

gateways, H.323 protocol, 72–73MGCP (Media Gateway Control Protocol),

47–49PRI trunks, 203–204SIP (Session Initiation Protocol), 110–111SRST (Survivable Remote Site Telephony),

404–405trunk circuits, PSTN (public switched telephone

network), 145–146analog connections, 146digital connections, 147

trunk groupscall route selection, 332–333outbound dial peer targets, 291–292

trunk sides, 187trunking gateways, 26trunks, PSTN (public switched telephone

network)digital, 158–178POTS, 147–157

Type I signaling, E&M (Ear and Mouth), 124Type II signaling, E&M (Ear and Mouth), 124Type III signaling, E&M (Ear and Mouth), 124Type IV signaling, E&M (Ear and Mouth), 125Type V signaling, E&M (Ear and Mouth), 125type jitter command, 343type of number (TON), ISDN address identifiers,

139–140

UUACs (user agent clients), 81UAs, SIP (Session Initiation Protocol), 81UASs (user agent servers), 81UDP (User Datagram Protocol), 14udp-jitter command, 340UK National Numbering Plan (UK NNP),

279–281UK NNP (UK National Numbering Plan),

279–281unicast discovery, 464user agent clients (UACs), 81user agent servers (UASs), 81

user agent servers (UASs)

618

user agents, SIP (session initiation protocol), 13User Datagram Protocol (UDP), 14

VV3PN, IP WAN security, 264–265VIA zones (Voice Infrastructure and Applications

zones), 10VIC (voice interface cards), 125VIC-2CAMA interface module, 155VIC-2DID model, 153VIC-4FXS/DID model, 153VIC2-2FXO interface module, 155VIC2-4FXO interface module, 155video, IPIPGW (Multiservice IP-to-IP gateway),

560–561vm-integration command, 394voice class codec command, 62voice class h323 command, 62voice class h323 10 command, 70voice-class sip transport switch udp tcp

command, 98voice classes, H.323 protocol, 62–64voice gateways. See gatewaysVoice Infrastructure and Applications zones (VIA

zones), 10voice interface cards (VIC), 125voice-mail, SRST (Survivable Remote Site

Telephony)centralized voice-mail system, 393–396CUE integration, 393

voice media, IP WAN security, 261IPsec, 263secure RTP, 261–263TLS, 263–264

Voice over IP networks. See VoIP networksvoice-port command, 148, 185voice register dn command, 372voice register pool configuration command, 372voice register pools, SIP SRST gateways, 397–398voice rules, creating, 308–309voice service command, 64voice service commands, H.323 protocol, 64–65voice service voip command, 65, 101, 397voice service voip configuration mode, 556

voice services, configuration, 101–102voice termination, DSPs, 411, 414–418

codecs, 418–419configuring codec complexity, 419–420flex complexity, 420–421

voice translation-profile command, 314voice translation-rule command, 308voices, translation profiles, 307–308

application, 314–315blocking calls, 315–316creating, 314creating voice rules, 308–309regular expressions, 309–314testing rules, 316

VoiceXML, 439–442VoIP, dial peers, 293–294voip-incoming translation-profile command, 315VoIP networks (Voice over IP networks), 5

gatekeepers role, 8–9gateways

role, 5–7types, 7–8

IP-to-IP gateways role, 10–11protocols

H.323, 12MGCP (Media Gateway Control

Protocol), 11RTP (Real-Time Transport Protocol), 14SCCP (skinny client control protocol), 13SIP (session initiation protocol), 12–13

Vonage, 80VSML, IPIPGW (Multiservice IP-to-IP

gateway), 568

WWANs

business application, 219–220design, 220–221fax services, 249–251, 254–255

Cisco fax relay, 251–252passthrough configuration, 256–258T.38 for MGCP gateways, 252

modems, 249–250, 258modem relay, 258–259passthrough, 259–260

user agents, SIP (session initiation protocol)

619

QoS (quality of service), 221–222AutoQoS, 248–249classifying traffic using class maps,

222–227compression, 246–248link fragmentation and interleave,

239–246MPLS class mapping, 232–239policy maps, 227–232

security, 260–261firewalls, 265–266NAT, 265V3PN, 264–265voice media, 261–264

wink-start operation, address signaling, 126

Zzone cluster command, 517zone local command, 486zone management, 9zone prefixes

gatekeeper configuration, 490–493gatekeeper E.164 number resolution, 470–471

zonesgatekeeper configuration, 486–487IPIPGW (Multiservice IP-to-IP gateway),

558–560

zones