CISCO IOS XE 17.5 Unified Border Element Configuration Guide Through User Guide

IOS XE 17.5 Unified Border Element Configuration Guide Through

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5
Last Modified: 2022-08-15
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CONTENTS

CHAPTER 1 CHAPTER 2 CHAPTER 3 PART I CHAPTER 4
CHAPTER 5

Read Me First 1 Short Description 2
New and Changed Information 3 New and Changed Information 3
Supported Platforms 5 Feature Comparison on Supported Platforms 7
CUBE Fundamentals and Basic Setup 11
Overview of Cisco Unified Border Element 13 Information About Cisco Unified Border Element 13 SIP/H.323 Trunking 16 Typical Deployment Scenarios for CUBE 17 How to Configure Basic CUBE Features 18 Enabling the CUBE Application on a Device 19 Verifying the CUBE Application on the Device 21 Configuring a Trusted IP Address List for Toll-Fraud Prevention 22
Virtual CUBE 25 Feature Information for Virtual CUBE 25 Prerequisites for Virtual CUBE 26 Hardware 26 Software 26 Features Supported with Virtual CUBE 27

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CHAPTER 6 CHAPTER 7
CHAPTER 8

Restrictions 27 Information about Virtual CUBE 27
Media 27 Virtual CUBE Licensing Requirements 28
Virtual CUBE with CSR1000V 28 Virtual CUBE with Catalyst 8000V 28 Install Virtual CUBE on ESXi 28 How to Enable Virtual CUBE 29 Troubleshooting Virtual CUBE 29
Dial-Peer Matching 31 Dial Peers in CUBE 31 Configuring Inbound and Outbound Dial-Peer Matching for CUBE 33 Preference for Dial-Peer Matching 34
DTMF Relay 37 Feature Information for DTMF Relay 37 Information About DTMF Relay 38 DTMF Tones 38 DTMF Relay 38 Configuring DTMF Relays 41 Interoperability and Priority with Multiple DTMF Relay Methods 42 DTMF Interoperability Table 42 Verifying DTMF Relay 46
Introduction to Codecs 51 Why CUBE Needs Codecs 51 Restrictions for Voice- Class Codec Transparent 52 Voice Media Transmission 52 Voice Activity Detection 53 VoIP Bandwidth Requirements 54 Supported Audio and Video Codecs 56 How to Configure Codecs 57 Configuring Audio and Video Codecs at the Dial Peer Level 57

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Contents

CHAPTER 9 CHAPTER 10

Configuring Audio Codecs Using a Codec Voice Class and Preference Lists 59 Configuring Video Codecs Using Codec Voice Class 61 Verifying an Audio Call 62 Configuration Examples for Codecs 62
Call Admission Control 65 Configuring CAC Based on Total Calls, CPU or Memory 65 Example: Internal Error Code (IEC) for Default Call Rejection Based on CPU Utilization and Memory 67 Configuring CAC Based on Call Spike Detection 67 Configuring CAC Based on Maximum Calls per Destination 68 Bandwidth-Based Call Admission Control 69 Restrictions for Bandwidth-Based Call Admission Control 70 Information About Bandwidth-Based Call Admission Control 70 Maximum Bandwidth Calculation 70 Bandwidth Tables 70 How to Configure Bandwidth-Based Call Admission Control 72 Configuring Bandwidth-Based Call Admission Control at the Interface Level 72 Configuring Bandwidth-Based Call Admission Control at the Dial Peer Level 74 Configuring the Bandwidth-Based Call Admission Control SIP Error Response Code Mapping 75 Verifying Bandwidth-Based Call Admission Control 77 Troubleshooting Tips 78 Configuration Examples for Bandwidth-Based Call Admission Control 79 Example: Configuring Bandwidth-Based Call Admission Control at the Interface Level 79 Example: Configuring Bandwidth-Based Call Admission Control at the Dial Peer Level 79 Example: Configuring the Bandwidth-Based Call Admission Control SIP Error Response Code Mapping at the Global Level 80 Example: Configuring the Bandwidth-Based Call Admission Control SIP Error Response Code Mapping at the Dial Peer Level 80 Feature Information for Bandwidth-Based Call Admission Control 80
Basic SIP Configuration 83 Prerequisites for Basic SIP Configuration 83 Restrictions for Basic SIP Configuration 83

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CHAPTER 11 CHAPTER 12

Information About Basic SIP Configuration 84 SIP Register Support 84 SIP Redirect Processing Enhancement 84 Sending SIP 300 Multiple Choice Messages 85
How to Perform Basic SIP Configuration 85 Configuring SIP VoIP Services on a Cisco Gateway 86 Shut Down or Enable VoIP Service on Cisco Gateways 86 Shut Down or Enable VoIP Submodes on Cisco Gateways 86 Configuring SIP Register Support 87 Configuring SIP Redirect Processing Enhancement 89 Configure Call- Redirect Processing Enhancement 89 Configuring SIP 300 Multiple Choice Messages 92 Configuring Sending of SIP 300 Multiple Choice Messages 92 Configuring SIP Implementation Enhancements 93 Interaction with Forking Proxies 93 SIP Intra-Gateway Hairpinning 94 Verifying SIP Gateway Status 95 General Troubleshooting Tips 99
Configuration Examples for Basic SIP Configuration 101 SIP Register Support Example 101 SIP Redirect Processing Enhancement Examples 103 SIP 300 Multiple Choice Messages Example 107
Toll Fraud Prevention 108
SIP Binding 111 Feature Information for SIP Binding 111 Information About SIP Binding 112 Benefits of SIP Binding 112 Source Address 113 Voice Media Stream Processing 116 Configuring SIP Binding 118 Verifying SIP Binding 120
Media Path 127

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CHAPTER 13

Feature Information for Media Path 127 Media Flow-Through 128
Restrictions for Media Flow-Through 128 Configure Media Flow-Through 129 Media Flow-Around 130 Configure Media Flow-Around 130 Media Anti-Trombone 131 Prerequisites 132 Restrictions for Media Anti-Tromboning 132 Configuring Media Anti-Tromboning 132
SIP Profiles 135 Feature Information for SIP Profiles 135 Information About SIP Profiles 136 Important Characteristics of SIP Profiles 137 Restrictions for SIP Profiles 139 How to Configure SIP Profiles 139 Configuring a SIP Profile to Manipulate SIP Request or Response Headers 140 Configuring SIP Profiles for Copying Unsupported SDP Headers 141 Example: Configuring SIP Profile Rules (Attribute Passing) 143 Example: Configuring SIP Profile Rules (Parameter Passing) 143 Example: Configuration to Remove an Attribute 143 Configuring SIP Profile Using Rule Tag 143 Configuring a SIP Profile for Non- standard SIP Header 145 Upgrading or Downgrading SIP Profile Configurations 147 Configuring a SIP Profile as an Outbound Profile 148 Configuring a SIP Profile as an Inbound Profile 149 Verifying SIP Profiles 150 Troubleshooting SIP Profiles 151 Examples: Adding, Modifying, Removing SIP Profiles 152 Example: Adding a SIP, SDP, or Peer Header 152 Example: Modifying a SIP, SDP, or Peer Header 153 Example: Remove a SIP, SDP, or Peer Header 156 Example: Inserting SIP Profile Rules 157

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CHAPTER 14 CHAPTER 15
CHAPTER 16

Example: Upgrading and Downgrading SIP Profiles automatically 157 Example: Modifying Diversion Headers 158 Example: Sample SIP Profile Application on SIP Invite Message 159 Example: Sample SIP Profile for Non-Standard SIP Headers 160 Example: Copy a User-to-User from REFER Message 160
SIP Out-of-Dialog OPTIONS Ping Group 163 Information About SIP Out-Of-dialog OPTIONS Ping Group 163 SIP Out-of-Dialog OPTIONS Ping Group Overview 163 How to Configure SIP Out-Of-dialog OPTIONS Ping Group 164 Configuring SIP Out-of- Dialog OPTIONS Ping Group 164 Configuration Examples For SIP Out-of-Dialog OPTIONS Ping Group 166 Additional References 168 Feature Information for SIP Out-of-dialog OPTIONS Ping Group 169
Configure TCL IVR Applications 171 Tcl IVR Overview 171 Tcl IVR Enhancements 172 RTSP Client Implementation 172 TCL IVR Prompts Played on IP Call Legs 173 TCL Verbs 174 TCL IVR Prerequisite Tasks 177 TCL IVR Configuration Tasks List 177 Configuring the Call Application for the Dial Peer 178 Configuring TCL IVR on the Inbound POTS Dial Peer 180 Configuring TCL IVR on the Inbound VoIP Dial Peer 182 Verifying TCL IVR Configuration 184 TCL IVR Configuration Examples 185 TCL IVR for Gateway1 (GW1) Configuration Example 185 TCL IVR for GW2 Configuration Example 188
VoIP for IPv6 191 Prerequisites for VoIP for IPv6 191 Restrictions for Implementing VoIP for IPv6 191

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Information About VoIP for IPv6 193 SIP Features Supported on IPv6 193 SIP Voice Gateways in VoIPv6 194 VoIPv6 Support on Cisco UBE 195
How to Configure VoIP for IPv6 199 Configuring VoIP for IPv6 199 Shutting Down or Enabling VoIPv6 Service on Cisco Gateways 200 Shutting Down or Enabling VoIPv6 Submodes on Cisco Gateways 201 Configuring the Protocol Mode of the SIP Stack 201 Verifying SIP Gateway Status 203 RTCP Pass-Through 205 Configuring IPv6 Support for Cisco UBE 205 Verifying RTP Pass-Through 206 Configuring the Source IPv6 Address of Signaling and Media Packets 207 Configuring the SIP Server 208 Configuring the Session Target 209 Configuring SIP Register Support 210 Configuring Outbound Proxy Server Globally on a SIP Gateway 212 Configuring UDP Checksum 213 Configuring IP Toll Fraud 214 Configuring the RTP Port Range for an Interface 215 Configuring Message Waiting Indicator Server Address 216 Configuring Voice Ports 217 Configuring Cisco UBE Mid-call Re- INVITE Consumption 218 Configuring Passthrough of Mid-call Signalling 218 Configuring Passthrough SIP Messages at Dial Peer Level 219 Configuring H.323 IPv4-to-SIPv6 Connections in a Cisco UBE 220
Configuration Examples for VoIP over IPv6 222 Example: Configuring the SIP Trunk 222
Troubleshooting Tips for VoIP for IPv6 223 Verifying and Troubleshooting Tips 223
Verifying Cisco UBE ANAT Call Flows 223 Verifying and Troubleshooting Cisco UBE ANAT Flow-Through Call 225 Verifying Cisco UBE ANAT Flow-Around Calls 230
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CHAPTER 17 CHAPTER 18
PART II

Verifying VMWI SIP 235 Verifying SDP Passthrough Configuration 236 Feature Information for VoIP for IPv6 241
Monitoring of Phantom Packets 247 Restrictions of Monitoring of Phantom Packets 247 Information About Monitoring of Phantom Packets 248 Monitoring of Phantom Packets 248 How to Configure Monitoring of Phantom Packets 248 Configuring Monitoring of Phantom Packets 248 Configuration Examples For Monitoring of Phantom Packets 250 Additional References for Configurable Pass- Through of SIP INVITE Parameters 250 Feature Information for Monitoring of Phantom Packets 251
Configurable SIP Parameters via DHCP 253 Finding Feature Information 253 Prerequisites for Configurable SIP Parameters via DHCP 253 Restrictions for Configurable SIP Parameters via DHCP 254 Information About Configurable SIP Parameters via DHCP 254 How to Configure SIP Parameters via DHCP 258 Configuring the DHCP Client 258 Configuring the DHCP Client Example 259 Enabling the SIP Configuration 260 Enabling the SIP Configuration Example 261 Troubleshooting Tips 261 Configuring a SIP Outbound Proxy Server 262 Configuring a SIP Outbound Proxy Server in Voice Service VoIP Configuration Mode 262 Configuring a SIP Outbound Proxy Server in Voice Service VoIP Configuration Mode Example 263 Configuring a SIP Outbound Proxy Server and Session Target in Dial Peer Configuration Mode 263 Configuring a SIP Outbound Proxy Server in Dial Peer Configuration Mode Example 264 Feature Information for Configurable SIP Parameters via DHCP 265
Dial Peer Enhancements 267

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CHAPTER 19 CHAPTER 20
CHAPTER 21

Matching Inbound Dial Peers by URI 269 Configuring an Inbound Dial Peer to Match on URI 269 Examples for Configuring an Inbound Dial Peer to Match on a URI 271
URI-Based Dialing Enhancements 273 Feature Information for URI-Based Dialing Enhancements 273 Information About URI-Based Dialing Enhancements 274 Call Flows for URI-Based Dialing Enhancements 274 How to Configure URI-Based Dialing Enhancements 277 Configuring Pass Through of SIP URI Headers 277 Configuring Pass Though of Request URI and To Header URI (Global Level) 277 Configuring Pass Though of Request URI and To Header URI (Dial Peer Level) 278 Configuring Pass Through of 302 Contact Header 279 Configuring Pass Through of 302 Contact Header (Global Level) 279 Configuring Pass Through of 302 Contact Header (Dial Peer Level) 280 Deriving of Session Target from URI 282 Configuration Examples for URI-Based Dialing Enhancements 284 Example: Configuring Pass Though of Request URI and To Header URI 284 Example: Configuring Pass Though of Request URI and To Header URI (Global Level) 284 Example: Configuring Pass Though of Request URI and To Header URI (Dial Peer Level) 284 Example: Configuring Pass Through of 302 Contact Header 284 Example: Configuring Pass Through of 302 Contact Header (Global Level) 284 Example: Configuring Pass Through of 302 Contact Header (Dial Peer Level) 284 Example: Deriving Session Target from URI 285 Additional References for URI- Based Dialing Enhancements 285
Multiple Pattern Support on a Voice Dial Peer 287 Feature Information for Multiple Pattern Support on a Voice Dial Peer 287 Restrictions for Multiple Pattern Support on a Voice Dial Peer 288 Information About Multiple Pattern Support on a Voice Dial Peer 288 Configuring Multiple Pattern Support on a Voice Dial Peer 288 Verifying Multiple Pattern Support on a Voice Dial Peer 290 Configuration Examples for Multiple Pattern Support on a Voice Dial Peer 292

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CHAPTER 22 CHAPTER 23 CHAPTER 24 CHAPTER 25

Outbound Dial-Peer Group as an Inbound Dial-Peer Destination 293 Feature Information for Outbound Dial-Peer Group as an Inbound Dial-Peer Destination 293 Restrictions 294 Information About Outbound Dial-Peer Group as an Inbound Dial-Peer Destination 294 Configuring Outbound Dial-Peer Group as an Inbound Dial-Peer Destination 295 Verifying Outbound Dial-Peer Groups as an Inbound Dial-Peer Destination 297 Troubleshooting Tips 298 Configuration Examples for Outbound Dial Peer Group as an Inbound Dial-Peer Destination 299
Inbound Leg Headers for Outbound Dial-Peer Matching 303 Feature Information for Inbound Leg Headers for Outbound Dial-Peer Matching 303 Prerequisites for Inbound Leg Headers for Outbound Dial-Peer Matching 304 Restrictions for Inbound Leg Headers for Outbound Dial-Peer Matching 304 Information About Inbound Leg Headers for Outbound Dial-Peer Matching 305 Configuring Inbound Leg Headers for Outbound Dial-Peer Matching 305 Verifying Inbound Leg Headers for Outbound Dial-Peer Matching 308 Configuration Example: Inbound Leg Headers for Outbound Dial-Peer Matching 310
Server Groups in Outbound Dial Peers 313 Feature Information for Configuring Server Groups in Outbound Dial Peers 313 Information About Server Groups in Outbound Dial Peers 314 How to Configure Server Groups in Outbound Dial Peers 315 Configuring Server Groups in Outbound Dial Peers 315 Verifying Server Groups in Outbound Dial Peers 318 Configuration Examples for Server Groups in Outbound Dial Peers 319
Domain-Based Routing Support on the Cisco UBE 323 Feature Information for Domain-Based Routing Support on the Cisco UBE 323 Restrictions for Domain- Based Routing Support on the Cisco UBE 324 Information About Domain-Based Routing Support on the Cisco UBE 324 How to Configure Domain-Based Routing Support on the Cisco UBE 325 Configuring Domain-Based Routing at Global Level 325 Configuring Domain-Based Routing at Dial Peer Level 326

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CHAPTER 26
PART III CHAPTER 27

Verifying and Troubleshooting Domain-Based Routing Support on the Cisco UBE 327 Configuration Examples for Domain-Based Routing Support on the Cisco UBE 330
Example Configuring Domain-Based Routing Support on the Cisco UBE 330
ENUM Enhancement per Kaplan Draft RFC 331 Feature Information for ENUM Enhancement per Kaplan Draft RFC 331 Restrictions for ENUM Enhancement per Kaplan Draft RFC 332 Information About ENUM Enhancement per Kaplan Draft RFC 333 How to Configure ENUM Enhancement per Kaplan Draft RFC 333 Enabling Source-Based Routing 333 Testing the ENUM Request 334 Verifying the ENUM Request 334 Troubleshooting Tips 336 Configuration Examples for ENUM Enhancement per Kaplan Draft RFC 336
Multi-Tenancy 339
Support for Multi-VRF 341 Feature Information for VRF 341 Information About Voice-VRF 343 Information About Multi-VRF 343 VRF Preference Order 344 Restrictions 344 Recommendations 345 Configuring VRF 345 Create a VRF 346 Assign Interface to VRF 347 Create Dial-peers 348 Bind Dial-peers 349 Configure VRF-Specific RTP Port Ranges 351 Example: VRF with overlapping and non-overlapping RTP Port Range 353 Directory Number (DN) Overlap across Multiple-VRFs 354 Example: Associating Dial-peer Groups to Overcome DN Overlap 355 IP Overlap with VRF 356

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CHAPTER 28 PART IV CHAPTER 29
CHAPTER 30

Using Server Groups with VRF 358 Inbound Dial-Peer Matching Based on Multi-VRF 359
Example: Inbound Dial-Peer Matching based on Multi-VRF 359 VRF Aware DNS for SIP Calls 361 High Availability with VRF 362 Configuration Examples 362
Example: Configuring Multi-VRF in Standalone Mode 362 Example: Configuring RG Infra High Availability with VRF 366 Example: Configuring HSRP High Availability with VRF 373 Example: Configuring Multi VRF where Media Flows Around the CUBE 380 Example: Configuring Multi VRF where Media Flows Through the CUBE 388 Troubleshooting Tips 393
Configuring Multi-Tenants on SIP Trunks 395 Feature Information for Configuring Multi-Tenants on SIP Trunks 395 Information About Configuring Multi-tenants on SIP Trunks 395 How to Configure Multi-Tenants on SIP Trunks 399 Configuring Multi-Tenants on SIP Trunks 399 Example: SIP Trunk Registration in Multi-Tenant Configuration 401
Codecs 403
Codec Support and Restrictions 405 Feature Information for Codec Support on CUBE 405 OPUS Codec Support on CUBE 406 Design Recommendations for Opus Codec 406 Restrictions for Opus Codec Support on CUBE 407 ISAC Codec Support on CUBE 408 Restrictions for ISAC Codec Support on CUBE 408 AAC-LD MP4A-LATM Codec Support on Cisco UBE 408 Restrictions for AAC-LD MP4A-LATM Codec Support on Cisco UBE 409
Codec Preference Lists 411 Feature Information for Negotiation of an Audio Codec from a List of Codecs 411

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PART V CHAPTER 31
CHAPTER 32 CHAPTER 33

Codecs Configured Using Preference Lists 412 Prerequisites for Codec Preference Lists 412 Restrictions for Codecs Preference Lists 413 How to Configure Codec Preference Lists 413
Configuring Audio Codecs Using a Codec Voice Class and Preference Lists 413 Disabling Codec Filtering 415 Troubleshooting Negotiation of an Audio Codec from a List of Codecs 416 Verifying Negotiation of an Audio Codec from a List of Codecs 417
DSP Services 421
Transcoding 423 Configure LTI-Based Transcoding 424 Configuration Examples for LTI Based Transcoding 426 Configuring SCCP-based Transcoding (ISR-G2 devices only) 428 TLS for SCCP Connection for DSP Services 431 Configuring Secure Transcoding 431 Configuring the Certificate Authority 431 Configuring a Trustpoint for the Secure Universal Transcoder 432 Configuring DSPFARM Services 434 Associating SCCP to the Secure DSPFARM Profile 434 Registering the Secure Universal Transcoder to the CUBE 437 Configuration Examples for SCCP Based Transcoding 439
Transrating 441 Configuring Transrating for a Codec 441
Call Progress Analysis Over IP-to-IP Media Session 443 Feature Information for Call Progress Analysis Over IP-IP Media Session 443 Restrictions for Call Progress Analysis Over IP-to-IP Media Session 444 Information About Call Progress Analysis Over IP-IP Media Session 445 Call Progress Analysis 445 CPA Events 445 How to Configure Call Progress Analysis Over IP-to-IP Media Session 446

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CHAPTER 34 CHAPTER 35
PART VI CHAPTER 36

Enabling CPA and Setting the CPA Parameters 446 Verifying the Call Progress Analysis Over IP-to-IP Media Session 448 Troubleshooting Tips 449 Configuration Examples for the Call Progress Analysis Over IP-to-IP Media Session 449 Example: Enabling CPA and Setting the CPA Parameters 449
Voice Packetization 451 Configuring Transrating for a Codec 451
Fax Detection for SIP Call and Transfer 453 Restrictions for Fax Detection for SIP Call and Transfer On Cisco IOS XE 453 Information About Fax Detection for SIP Call and Transfer 453 Local Redirect Mode 454 Refer Redirect Mode 455 Fax Detection with Cisco IOS XE High Availability 456 How to Configure Fax Detection for SIP Calls 456 Configure DSP Resource to Detect Fax Tone 456 Dial-peer Configuration to Redirect Fax Call 457 Verifying Fax Detection for SIP Calls 459 Troubleshooting Fax Detection for SIP Calls 460 Configuration Examples for Fax Detection for SIP Calls 460 Example: Configuring Local Redirect 460 Example: Configuring Refer Redirect 461 Feature Information for Fax Detection for SIP Call and Transfer 461
Video 463
Video Suppression 465 Feature Information for Video Suppression 465 Restrictions 465 Information About Video Suppression 466 Feature Behavior 466 Configuring Video Suppression 466 Troubleshooting Tips 467

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PART VII CHAPTER 37 PART VIII CHAPTER 38
CHAPTER 39

Media Services 469
Configuring RTCP Report Generation 471 Prerequisites 471 Restrictions 471 Configuring RTCP Report Generation on Cisco UBE 472 Troubleshooting Tips 473 Feature Information for Configuring RTCP Report Generation 474
Media Recording 477
Network-Based Recording 479 Feature Information for Network-Based Recording 479 Restrictions for Network-Based Recording 480 Information About Network- Based Recording Using CUBE 481 Deployment Scenarios for CUBE-based Recording 481 Open Recording Architecture 482 Network Layer 483 Capture and Media Processing Layer 483 Application Layer 483 Media Forking Topologies 484 Media Forking with Cisco UCM 484 Media Forking without Cisco UCM 484 SIP Recorder Interface 484 Metadata 484 How to Configure Network-Based Recording 485 Configuring Network-Based Recording (with Media Profile Recorder) 485 Configuring Network-Based Recording (without Media Profile Recorder) 488 Verifying the Network-Based Recording Using CUBE 490 Additional References for Network-Based Recording 505
SIPREC (SIP Recording) 507 Feature Information for SIPREC-based Recording 507

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CHAPTER 40

Prerequisites for SIPREC Recording 508 Restrictions for SIPREC Recording 508 Information About SIPREC Recording Using CUBE 509
Deployment 509 SIPREC High Availability Support 510 How to Configure SIPREC- Based Recording 510 Configuring SIPREC-Based Recording (with Media Profile Recorder) 510 Configuring SIPREC-Based Recording (without Media Profile Recorder) 513 Configuration Examples for SIPREC-based Recording 515 Example: Configuring SIPREC-based Recording with Media Profile Recorder 515 Example: Configuring SIPREC-based Recording without Media Profile Recorder 516 Validate SIPREC Functionality 516 Troubleshoot 517 Configuration Example for Metadata Variations with Different Mid-call Flows 521 Example: Complete SIP Recording Metadata Information Sent in INVITE or Re-INVITE 521 Example: Hold with Send- only / Recv-only Attribute in SDP 524 Example: Hold with Inactive Attribute in SDP 527 Example: Escalation 529 Example: De-escalation 531 Configuration Example for Metadata Variations with Different Transfer Flows 534 Example: Transfer of Re-INVITE/REFER Consume Scenario 534 Configuaration Examples for Metadata Variations with Caller-ID UPDATE Flow 535 Example: Caller-ID UPDATE Request and Response Scenario 535 Configuration Example for Metadata Variations with Call Disconnect 536 Example: Disconnect while Sending Metadata with BYE 536
Video Recording – Additional Configurations 537 Feature Information for Video Recording – Additional Configurations 537 Information About Additional Configurations for Video Recording 538 Full Intra-Frame Request 538 How to Configure Additional Configurations for Video Recording 538 Enabling FIR for Video Calls (Using RTCP of SIP INFO) 538 Configuring H.264 Packetization Mode 539 Monitoring Reference files or Intra Frames 540

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CHAPTER 41 CHAPTER 42

Verifying Additional Configurations for Video Recording 541
Third-Party GUID Capture for Correlation Between Calls and SIP-based Recording 543 Feature Information for Third-Party GUID Capture for Correlation Between Calls and SIP-based Recording 543 Restrictions for Third-Party GUID Capture for Correlation Between Calls and SIP-based Recording 544 Information About Third-Party GUID Capture for Correlation Between Calls and SIP-based recording 544 How to Capture Third-Party GUID for Correlation Between Calls and SIP- based Recording 544 Verifying Third-Party GUID Capture for Correlation Between Calls and SIP-based Recording 547 Configuration Examples for Third-Party GUID Capture for Correlation Between Calls and SIP-based Recording 548
Cisco Unified Communications Gateway Services–Extended Media Forking 551 Feature Information for Cisco Unified Communications Gateway Services–Extended Media Forking 551 Restrictions for Extended Media Forking 552 Information About Cisco Unified Communications Gateway Services 552 Extended Media Forking (XMF) Provider and XMF Connection 552 XMF Call-Based Media Forking 553 XMF Connection-Based Media Forking 554 Extended Media Forking API with Survivability TCL 554 Media Forking for SRTP Calls 555 Crypto Tag 555 Example of SDP Data sent in an SRTP Call 556 Multiple XMF Applications and Recording Tone 556 Forking Preservation 558 How to Configure UC Gateway Services 558 Configuring Cisco Unified Communication IOS Services on the Device 558 Configuring the XMF Provider 561 Verifying the UC Gateway Services 562 Troubleshooting Tips 565 Configuration Examples for UC Gateway Services 565 Example: Configuring Cisco Unified Communication IOS Services 565

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PART IX CHAPTER 43

Example: Configuring the XMF Provider 566 Example: Configuring UC Gateway Services 566
CUBE Media Proxy 567
CUBE Media Proxy 569 Feature Information for CUBE Media Proxy 569 Supported Platforms 570 Restrictions for CUBE Media Proxy 570 CUBE Media Proxy Using Unified CM Network-Based Recording 571 SIPREC-Based CUBE Media Proxy 571 About Multiple Media Forking Using CUBE Media Proxy 571 Secure Forking of Secure and Nonsecure Calls 572 Deployment Scenarios for CUBE Media Proxy 572 CUBE Media Proxy Using Unified CM Network-Based Recording 572 SIPREC-Based CUBE Media Proxy 574 Recording Metadata 575 Session Identifier 577 Session-ID Handling 577 Recording State Notification 579 SIP Info Messages from CUBE Media Proxy to Unified CM 579 SIP Info Message Sent During the Initial Call 580 SIP Info Message Sent During the Initial Call (All the Recorders as Optional) 580 SIP Info Message Sent During the Initial Call (One Recorder as Mandatory and Remaining as Optional) 581 How to Configure CUBE Media Proxy 582 How to Configure CUBE Media Proxy for Network-Based Recording Solutions 582 Configure Outbound Dial-Peers to the Recorders 582 Configure CUBE Media Proxy 584 Configure Inbound Dial-Peer from Unified CM 586 How to Configure CUBE Media Proxy for SIPREC Solutions 587 Verification of CUBE Media Proxy Configuration 587 Supported Features 598 Mid-Call Message Handling 598

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PART X CHAPTER 44 CHAPTER 45
CHAPTER 46
PART XI CHAPTER 47

Secure Recording of Secure Calls and Nonsecure Calls 598 Support for High Availability 599 Media Latch 599
SIP Header Manipulation 601
Passing Headers Unsupported by CUBE 603 Feature Information for Copying with SIP Profiles 603 Example: Passing a Header Not Supported by CUBE 603
Copying SIP Headers 605 Feature Information for Copying with SIP Profiles 605 How to Copy SIP Header Fields to Another 606 Copying From an Incoming Header and Modifying an Outgoing Header 606 Copying From One Outgoing Header to Another 608 Example: Copying the To Header into the SIP-Req-URI 609
Manipulate SIP Status-Line Header of SIP Responses 611 Feature Information for Manipulating SIP Responses 611 Copying Incoming SIP Response Status Line to Outgoing SIP Response 612 Modifying Status-Line Header of Outgoing SIP Response with User Defined Values 615
Payload Type Interoperability 617
Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls 619 Feature Information for Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls 619 Restrictions for Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls 620 Symmetric and Asymmetric Calls 620 High Availability Checkpointing Support for Asymmetric Payload 621 How to Configure Dynamic Payload Type Passthrough for DTMF and Codec Packets for SIP-to-SIP Calls 622 Configuring Dynamic Payload Type Passthrough at the Global Level 622

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PART XII CHAPTER 48
CHAPTER 49 CHAPTER 50

Configuring Dynamic Payload Type Passthrough for a Dial Peer 623 Verifying Dynamic Payload Interworking for DTMF and Codec Packets Support 624 Troubleshooting Tips 624 Configuration Examples for Assymetric Payload Interworking 625 Example: Asymmetric Payload Interworking–Passthrough Configuration 625 Example: Asymmetric Payload Interworking–Interworking Configuration 626
Protocol Interworking 627
Delayed-Offer to Early-Offer 629 Feature Information for Delayed-Offer to Early-Offer 629 Prerequisites for Delayed-Offer to Early-Offer 630 Restrictions for Delayed-Offer to Early-Offer Media Flow-Around 630 Delayed- Offer to Early-Offer in Media Flow-Around Calls 630 Configuring Delayed Offer to Early Offer 631 Configuring Delayed Offer to Early Offer for Video Calls 632 Configuring Delayed Offer to Early Offer Medial Flow-Around 633 MidCall Renegotiation Support for Delayed-Offer to Early-Offer Calls 634 Restrictions for MidCall Renegotiation Support for DO-EO Calls 635 Configuring Mid Call Renegotiation Support for Delayed-Offer to Early-Offer Calls 635 High-Density Transcoding Calls in Delayed-Offer to Early-Offer 636 Restrictions for High- Density Transcoding DO-EO Calls 637 Configuring High-Density Transcoding 637
H.323-to-SIP Interworking on CUBE 639 Prerequisites 639 Restrictions 639 H.323 -to-SIP Basic Call Interworking 640 H.323-to-SIP Supplementary Features Interworking 642 H.323-to-SIP Codec Progress Indicator Interworking for Media Cut-Through 643 Configuring H.323-to-SIP Interworking 643
H.323-to-H.323 Interworking on CUBE 645 Feature Information for H.323-to-H.323 Interworking 645

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Contents

CHAPTER 51
PART XIII CHAPTER 52

Prerequisites 646 Restrictions 646 Slow Start to Fast-Start Interworking 646
Restrictions for Slow-Start and Fast-Start Interworking 647 Enabling Interworking between Slow Start and Fast Start 647 Call Failure Recovery (Rotary) 648 Enabling Call Failure Recovery (Rotary) without Identical Codec Configuration 648 Managing H.323 IP Group Call Capacities 649 Configuration Examples for Managing H.323 IP Group Call Capacities 651 Overlap Signaling 654 Configuring Overlap Signaling 654 Verifying H.323-to-H.323 Interworking 655 Troubleshooting H.323-to-H.323 Interworking 657
SIP RFC 2782 Compliance with DNS SRV Queries 659 Prerequisites SIP RFC 2782 Compliance with DNS SRV Queries 659 Information SIP RFC 2782 Compliance with DNS SRV Queries 659 How to Configure SIP-RFC 2782 Compliance with DNS SRV Queries 660 Configuring DNS Server Query Format RFC 2782 Compliance with DNS SRV Queries 660 Configuring DNS Server Lookups 661 Verifying 663 Feature Information for SIP RFC 2782 Compliance with DNS SRV Queries 663
Support for SRTP 665
SRTP-SRTP Interworking 667 Feature Information for SRTP-SRTP Interworking 667 Prerequisites for SRTP-SRTP Interworking 668 Restrictions for SRTP-SRTP Interworking 668 Information About SRTP-SRTP Interworking 668 Supplementary Services Support 669 How to Configure SRTP-SRTP Interworking 670 Configuring SRTP 670 Configuring Cipher Suite Preference (optional) 672

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Contents

CHAPTER 53 CHAPTER 54

Applying Crypto Suite Selection Preference (optional) 673 Enabling SRTP Fallback 675 Configuration Examples 678 Example: Configuring SRTP-SRTP Interworking 678 Example: Changing the Cipher-Suite Preference 680
SRTP-RTP Interworking 683 Feature Information for SRTP-RTP Interworking 683 Prerequisites for SRTP-RTP Interworking 684 Restrictions for SRTP-RTP Interworking 684 Information About SRTP-RTP Interworking 684 Support for SRTP- RTP Interworking 684 Using SRTP-RTP Chain for Interworking Between AES_CM_128_HMAC_SHA1_32 and AES_CM_128_HMAC_SHA1_80 Crypto Suites 686 Supplementary Services Support 687 How to Configure Support for SRTP-RTP Interworking 688 Configuring SRTP-RTP Interworking Support 688 Configuring Crypto Authentication 690 Enabling SRTP Fallback 692 Troubleshooting Tips 694 Verifying SRTP-RTP Supplementary Services Support 694 Configuration Examples for SRTP-RTP Interworking 695 Example: SRTP-RTP Interworking 695 Example: Configuring Crypto Authentication 696 Example: Configuring Crypto Authentication (Dial Peer Level) 696 Example: Configuring Crypto Authentication (Global Level) 696
SRTP-SRTP Pass-Through 697 Feature Information for Support of SRTP-SRTP Pass- Through Calls 697 Information About SRTP-SRTP Pass-Through 698 Pass-Through of Unsupported Crypto Suites 698 Configure Pass-Through of Unsupported Crypto Suites for a Specific Dial Peer 699 Configure Pass-Through of Unsupported Crypto Suites Globally 701 Configuration Examples for SRTP-SRTP Pass-Through 702

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Contents

PART XIV CHAPTER 55
CHAPTER 56

High Availability 705
High Availability on Cisco 4000 Series ISR and Cisco Catalyst 8000 Series Edge Platforms 707 About CUBE High Availability on Cisco 4000 Series ISR and Cisco Catalyst 8000 Series Edge Platforms 707 Box-to-Box Redundancy 707 Redundancy Group (RG) Infrastructure 708 Network Topology 708 Considerations and Restrictions 710 Considerations 710 Restrictions 711 How to Configure CUBE High Availability on Cisco 4000 Series ISR and Cisco Catalyst 8000 Series Edge Platforms 712 Before You Begin 712 Configure High Availability 713 Configuration Examples 718 Example: Control Interface Protocol Configuration 718 Example: Redundancy Group Protocol Configuration 718 Example: Redundant Traffic Interface Configuration 718 Verify Your Configuration 718 Troubleshoot High Availability Issues 726
High Availability on Cisco ASR 1000 Series Aggregation Services Routers 729 About CUBE High Availability on Cisco ASR 1000 Series Routers 729 Inbox Redundancy 730 Box-to-Box Redundancy 731 Redundancy Group (RG) Infrastructure 731 PROTECTED Mode 732 Network Topology 732 Considerations and Restrictions 734 Considerations 734 Restrictions 735 How to Configure CUBE High Availability on Cisco ASR 1000 Series Router 736

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Contents

CHAPTER 57 CHAPTER 58

Before You Begin 736 Configure Inbox High Availability 737 Configure Box-to- Box High Availability 737 Configuration Examples 743 Verify Your Configuration 749 Verify Redundancy State on Active and Standby Routers 749 Verify Call State After Switchover 751 Verify SIP IP Address Bindings 754 Verify Current CPU Use 755 Force a Manual Failover for Testing 755 Troubleshoot High Availability Issues 756
High Availability on Cisco CSR 1000V or C8000V Cloud Services Routers 759 About vCUBE High Availability on CSR 1000V or C8000V Cloud Services Routers 759 Box-to-Box Redundancy 760 Redundancy Group (RG) Infrastructure 760 Network Topology 761 Considerations and Restrictions 763 Considerations 764 Restrictions 765 How to Configure vCUBE High Availability on Cisco CSR 1000v or C8000V 766 Before You Begin 766 Configure High Availability 766 Configuration Example 768 Troubleshoot High Availability Issues 769
High Availability on Cisco Integrated Services Routers (ISR-G2) 771 About CUBE High Availability on Cisco ISR-G2 771 Box-to-Box Redundancy 771 Hot Standby Router Protocol (HSRP) 772 Network Topology 772 Configure CUBE High Availability Using HSRP 773 Verify Redundancy State 784 Verify Call State After a Switchover 787

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Contents

CHAPTER 59 CHAPTER 60

Considerations and Restrictions 790 Considerations 790 Restrictions 791
How to Configure CUBE High Availability on Cisco ISR-G2 791 Before You Begin 791 Configure High Availability 791 Configuration Examples 800 Example Configuration for Dual-Attached CUBE HSRP Redundancy 800 Example Configuration for Single-Attached CUBE HSRP Redundancy 803
Verify Your Configurations 805 Verify SIP IP Address Bindings 805 Verify Current CPU Use 805 Verify the Call Processing During a Switchover 805 Force a Manual Failover for Testing 806
Troubleshoot High Availability Issues 808
DSP High Availability Support 811 Feature Information for DSP High Availability Support on CUBE 811 Prerequisites for DSP High Availability 811 Features Supported with DSP High Availability 812 Restrictions for DSP High Availability 812 Troubleshooting DSP HA Support on CUBE 812 Configuration Examples for DSP HA 813
Stateful Switchover Between Redundancy Paired Intra- or Inter-box Devices 815 Feature Information for Stateful Switchover Between Redundancy Paired Intra- or Inter-box Devices 815 Prerequisites for Stateful Switchover Between Redundancy Paired Intra- or Inter-box Devices 816 Restrictions for Stateful Switchover Between Redundancy Paired Intra- or Inter-box Devices 817 Information About Stateful Switchover Between Redundancy Paired Intra- or Inter-box Devices 817 Call Escalation with Stateful Switchover 818 Call De- escalation with Stateful Switchover 818 Media Forking with High Availability 819 High Availability Protected Mode and Box-to-Box Redundancy for ASR 819 Support for Box-to-Box High Availability with Virtual IP Addresses 820

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Contents

CHAPTER 61
PART XV CHAPTER 62

Monitoring Call Escalation and De-escalation with Stateful Switchover 820 Monitoring Media Forking with High Availability 822 Verifying the High Availability Protected Mode 824 Support for REFER and BYE/Also after Stateful Switch-Over 825 Troubleshooting Tips 825 Example: Configuring the Interfaces for ISR-G2 Devices 827 Example: Configuring the Interfaces for ASR Devices 827 Example: Configuring SIP Binding 827
CVP Survivability TCL support with High Availability 829 Feature Information for CVP Survivability TCL support with High Availability 829 Prerequisites 830 Restrictions 830 Recommendations 830 CVP Survivability TCL support with High Availability 830 Configuring CVP Survivability TCL support with High Availability 830
ICE-Lite Support on CUBE 831
ICE-Lite Support on CUBE 833 Feature Information for ICE-Lite Support on CUBE 833 Restrictions for ICE-lite Support on CUBE 834 Information About ICE-Lite Support on CUBE 834 Characteristics 834 ICE Candidate 835 ICE Lite 835 High Availability Support with ICE 835 How to Configure ICE-Lite Support on CUBE 836 Configuring ICE on the CUBE 836 Verifying ICE-Lite on the CUBE (Success Flow Calls) 837 ICE-Lite on CUBE (Error Flow Calls) 840 Troubleshooting ICE- Lite Support on CUBE 845 Additional References 845

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Contents

PART XVI CHAPTER 63
CHAPTER 64 CHAPTER 65

SIP Protocol Handling 847
Mid-call Signaling Consumption 849 Feature Information for Mid-call Signaling 849 Prerequisites 850 Mid-call Signaling Passthrough – Media Change 850 Restrictions for Mid-Call Signaling Passthrough – Media Change 851 Behavior of Mid-call Re-INVITE Consumption 851 Configuring Passthrough of Mid-call Signalling 853 Example Configuring Passthrough SIP Messages at Dial Peer Level 854 Example Configuring Passthrough SIP Messages at the Global Level 854 Mid- call Signaling Block 854 Restrictions for Mid-Call Signaling Block 854 Blocking Mid-Call Signaling 855 Example Blocking SIP Messages at Dial Peer Level 856 Example: Blocking SIP Messages at the Global Level 856 Mid Call Codec Preservation 857 Configuring Mid Call Codec Preservation 857 Example: Configuring Mid Call Codec Preservation at the Dial Peer Level 858 Example: Configuring Mid Call Codec Preservation at the Global Level 858
Early Dialog UPDATE Block 859 Feature Information for Early Dialog UPDATE Block 859 Prerequisites 860 Restrictions 860 Information about Early Dialog UPDATE Block 860 Important Characteristics of Early Dialog UPDATE Block 860 Configuring Early Dialog UPDATE Block 861 Configuring Early Dialog UPDATE Block Renegotiate 862 Troubleshooting Tips 863
Consumption of Forked 18x Responses with SDP During Early Dialog 865

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Contents

CHAPTER 66 CHAPTER 67
PART XVII

Feature Information for Consumption of Multiple Forked 18x Responses with SDP During Early Dialog 865
Prerequisites 866 Restrictions 866 Information About Consumption of Forked 18x Responses with SDP During Early Dialog 866
Characteristics of Forked 18x Responses with SDP during Early Dialog 866 Configuring Consumption of Forked 18x Responses with SDP During Early Dialog 867 Configuring Consumption of Forked 18x Responses with SDP During Early Dialog Renegotiate 868 Troubleshooting Tips 870
Support for Pass-Through of Unsupported Content Types in SIP INFO Messages 871 Feature Information 871 Configure SIP INFO Message with Unsupported Content Type 871 Information About Pass-Through of Unsupported Content Types in SIP INFO Messages 872
Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element 873 Feature Information for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element 883 Prerequisites for Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element 884 Restrictions for Support for PAID PPID Privacy PCPID and PAURI Headers on the Cisco Unified Border Element 885 Configuring P-Header and Random-Contact Support on the Cisco Unified Border Element 885 Configuring P-Header Translation on a Cisco Unified Border Element 885 Configuring P-Header Translation on an Individual Dial Peer 886 Configuring P-Called-Party-Id Support on a Cisco Unified Border Element 887 Configuring P-Called-Party-Id Support on an Individual Dial Peer 888 Configuring Privacy Support on a Cisco Unified Border Element 889 Configuring Privacy Support on an Individual Dial Peer 890 Configuring Random-Contact Support on a Cisco Unified Border Element 891 Configuring Random-Contact Support for an Individual Dial Peer 893
SIP Supplementary Services 895

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Contents

CHAPTER 68
CHAPTER 69
PART XVIII CHAPTER 70 CHAPTER 71

Dynamic Refer Handling 897 Feature Information for Dynamic REFER Handling 897 Prerequisites 898 Restrictions 898 Configuring REFER Passthrough with Unmodified Refer-to 898 Configuring REFER Consumption 900 Troubleshooting Tips 902
Cause Code Mapping 903 Feature Information for Cause Code Mapping 903 Cause Code Mapping 904 Configuring Cause Code Mapping 905 Verifying Cause Code Mapping 906
Hosted and Cloud Services 909
Hosted and Cloud Services Delivery with CUBE 911
CUBE SIP Registration Proxy 913 Registration Pass-Through Modes 913 End-to-End Mode 913 Peer-to-Peer Mode 914 Registration in Different Registrar Modes 915 Registration Overload Protection 916 Registration Overload Protection–Call Flow 916 Registration Rate-limiting 916 Registration Rate-limiting Success–Call Flow 917 Prerequisites for SIP Registration Proxy on Cisco UBE 917 Restrictions 917 Configuring CUBE SIP Registration Proxy 917 Enabling Local SIP Registrar 917 Configuring SIP Registration Proxy at the Global Level 919 Configuring SIP Registration Proxy at the Tenant Level 920

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Contents

CHAPTER 72

Configuring SIP Registration Proxy at the Dial Peer Level 922 Configuring Registration Overload Protection Functionality 923 Configuring Cisco UBE to Route a Call to the Registrar Endpoint 924 Verifying the SIP Registration on Cisco UBE 925 Configuration Example–CUBE SIP Registration Proxy 926 Feature Information for CUBE SIP Registration Proxy 927
Survivability for Hosted and Cloud Services 929 Information About Survivability for Hosted and Cloud Services 929 Advantages of Using CUBE Survivability Feature 929 Local Fallback 929 Registration Synchronization 930 Registration Through Alias Mapping 930 CUBE when WAN is UP 931 CUBE Survivability When WAN Is Down 932 How to Configure Survivability for Hosted and Cloud Services 934 Configuring Local Fallback or Registration Synchronization Globally 934 Configuring Local Fallback or Registration Synchronization at the Tenant Level 935 Configuring Local Fallback or Registration Synchronization on a Dial Peer 936 Configuring Survivability for Phones Sending Single Register Request 937 Configuring OPTIONS Ping 938 Configuring Registration Timer 939 Configuring the REGISTER Message Throttling in CUBE 940 Configuring the Class of Restrictions (COR) List 941 Verifying Survivability for Hosted and Cloud Services 943 Configuration Examples–Survivability for Hosted and Cloud Services 945 Example: Configuring Local Fallback Globally 945 Example: Configuring Local Fallback at the Tenant Level 946 Example: Configuring Local Fallback on a Dial Peer 946 Example: Configuring Survivability for Phones Sending Single Register Request 946 Example: Configuring OPTIONS Ping 946 Example: Configuring the Registration Timer 946 Example: Configuring REGISTER Message Throttling 947 Example: Configuring the COR List 947

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Contents

CHAPTER 73
PART XIX CHAPTER 74

Feature Information for Survivability for Hosted and Cloud Services 947
SUBSCRIBE-NOTIFY Passthrough 949 Restrictions for SUBSCRIBE-NOTIFY Passthrough 949 Information About SUBSCRIBE-NOTIFY Passthrough 950 SUBSCRIBE-NOTIFY Passthrough Request Routing 950 SUBSCRIBE-NOTIFY Passthrough Survivability Mode 951 Configure SUBSCRIBE-NOTIFY Passthrough 951 Configuring an Event List 951 Configuring SUBSCRIBE-NOTIFY Event Passthrough Globally 952 Configuring SUBSCRIBE-NOTIFY Event Passthrough at the Dial-Peer Level 953 Verifying SUBSCRIBE-NOTIFY Passthrough 954 Troubleshooting Tips 956 Configuration Examples for SUBSCRIBE-NOTIFY Passthrough 956 Example: Configuring an Event List 956 Example: Configuring SUBSCRIBE-NOTIFY Event Passthrough Globally 956 Example: Configuring SUBSCRIBE-NOTIFY Event Passthrough under a Dial Peer 957 Feature Information for SUBSCRIBE-NOTIFY Passthrough 957
Cisco Unified Communications Manager Line-Side Support 959
Cisco Unified Communications Manager Line-Side Support 961 Feature Information for Cisco Unified Communications Manager Line-Side Support 961 Restrictions for Cisco Unified Communications Manager Line-Side Support 962 Information About Cisco Unified Communications Manager Line-Side Support 963 Cisco UBE Line-Side Deployment 963 Line-Side Deployment Scenarios 963 Line-Side Support for CUCM on CUBE 964 Configuring a PKI Trustpoint 965 Importing the CUCM and CAPF Key 966 Creating a CTL File 967 Configuring a Phone Proxy 968 Attaching a Phone Proxy to a Dial Peer 969 Verifying CUCM Lineside Support 971

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Contents

PART XX CHAPTER 75
PART XXI CHAPTER 76
CHAPTER 77

Example: Configuring a PKI Trustpoint 973 Example: Importing the CUCM and CAPF Key 974 Example: Creating a CTL File 974 Example: Configuring a Phone Proxy 974 Example: Attaching a Phone Proxy to a Dial Peer 974 Example: Configuring CUCM Secure Line-Side 975 Example: Configuring CUCM Non-Secure Line-Side 977
Security 981
SIP TLS Support on CUBE 983 Feature Information for SIP TLS Support on CUBE 983 Restrictions 984 Information About SIP TLS Support on CUBE 985 Deployment 985 TLS Cipher Suite Category 985 How to Configure SIP TLS Support on CUBE 986 Configuring SIP TLS on CUBE 986 Verifying SIP TLS Configuration 994 SIP TLS Configuration Examples 995 Example: SIP TLS Configuration 995
Voice Quality in CUBE 1001
CUBE Call Quality Statistics Enhancement 1003 Feature Information for Call Quality Statistics Enhancement 1003 Restrictions for Call Quality Statistics Enhancement 1004 Information About Call Quality Statistics Enhancement 1004 How to Configure Call Quality Parameters 1005 Configuring Call Quality Criteria Parameters 1005 Troubleshooting Call Quality Statistics 1006 Configuration Example for Call Quality Statistics 1007
Voice Quality Monitoring 1009

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PART XXII CHAPTER 78
PART XXIII CHAPTER 79

Feature Information for Voice Quality Monitoring 1009 Prerequisites for Voice Quality Monitoring 1010 Restrictions for Voice Quality Monitoring and Voice Quality Statistics 1011 Information About Voice Quality Monitoring 1011
VQM Metrics 1012 How to Configure Voice Quality Monitoring 1012
Enabling Media Statistics Globally 1012 Verifying Voice Quality Monitoring 1013 Troubleshooting Tips 1015 Configuration Examples for Voice Quality Monitoring 1016 Example: Configuring Media Statistics Globally 1016 Example: CDR Enabled MOS Output 1016
Smart Licensing 1017
CUBE Smart Licensing 1019 Smart License Operation 1019 Smart Software Licensing Task Flow for CUBE 1021 Obtain the Registration ID Token 1021 Configure Smart Licensing Transport Settings 1021 Associate the Host Platform with CSSM 1022 Configure CUBE Licensed Features 1022 Verify Smart Licensing Operation for CUBE 1023 CUBE High Availability Configurations 1027 Smart Licensing with CUBE Box-to-Box High Availability 1027 Verify Smart Licensing Operation for Box-to-Box High Availability 1028 Smart Licensing with CUBE Inbox High Availability 1030 Verify Smart Licensing Operation for Inbox High Availability 1031 Syslog Messages 1032
Serviceability 1033
VoIP Trace for CUBE 1035 VoIP Trace for CUBE 1035

Contents

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Contents

CHAPTER 80
PART XXIV CHAPTER 81

Prerequisites for Voip Trace 1036 Benefits of VoIP Trace 1036 Guide to using VoIP Trace Framework 1037 RTP Port Clear 1038 Feature Information for VoIP Trace 1039
Support for Session Identifier 1041 Feature Information for Session Identifier Support 1041 Restrictions 1042 Information About Session Identifier 1042 Feature Behavior 1043 Configuring Support for Session Identifier 1043 Troubleshooting Tips 1043
Security Compliance 1051
Common Criteria (CC) and The Federal Information Processing Standards (FIPS) Compliance 1053 Feature Information for Common Criteria (CC) and the Federal Information Standards (FIPS) Compliance 1054 Supported Hardware and Software for Virtual CUBE 1054 Common Criteria Configuration on Cisco CSR 1000v 1054 Enable Common Criteria Mode 1054 SIP TLS Configuration 1055 SIP TLS Configuration Task Flow 1055 Generate RSA Public Key 1055 Configure Certificate Authority Server 1056 Configure CSR Trustpoint 1057 Configure Peer Trustpoint 1058 Add Client Verification Trustpoint 1059 Enforce Strict SRTP 1060 HTTPS TLS Configuration 1061 HTTPS TLS Configuration Task Flow 1061 Prepare Cisco CSR 1000v Router’s HTTP Server to Run in CC Mode 1061 Create Certificate Map for HTTPS Peer Trustpoint 1062

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PART XXV CHAPTER 82
CHAPTER 83

Configure HTTPS TLS Version 1063 Configure Supported Cipher Suites 1064 Apply Certificate Map to HTTPS Peer Trustpoint 1064 NTP Configuration Restrictions in Common Criteria Mode 1065 FIPS Configuration on Cisco CSR 1000v 1066 Configuration Requirements for FIPS Compliance 1066
Appendixes 1067
Additional References 1069 Related References 1069 Standards 1070 MIBs 1070 RFCs 1070 Technical Assistance 1072
Glossary 1073 Glossary 1073

Contents

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Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5

Read Me First
Important Information

1 C H A P T E R

Note For CUBE feature support information in Cisco IOS XE Bengaluru 17.6.1a and later releases, see Cisco Unified Border Element IOS-XE Configuration Guide.

Note The documentation set for this product strives to use bias-free language. For purposes of this documentation set, bias-free is defined as language that does not imply discrimination based on age, disability, gender, racial identity, ethnic identity, sexual orientation, socioeconomic status, and intersectionality. Exceptions may be present in the documentation due to language that is hardcoded in the user interfaces of the product software, language used based on standards documentation, or language that is used by a referenced third-party product.
Feature Information Use Cisco Feature Navigator to find information about feature support, platform support, and Cisco software image support. An account on Cisco.com is not required.
Related References · Cisco IOS Command References, All Releases
Obtaining Documentation and Submitting a Service Request · To receive timely, relevant information from Cisco, sign up at Cisco Profile Manager. · To get the business impact you’re looking for with the technologies that matter, visit Cisco Services. · To submit a service request, visit Cisco Support. · To discover and browse secure, validated enterprise-class apps, products, solutions and services, visit Cisco Marketplace. · To obtain general networking, training, and certification titles, visit Cisco Press. · To find warranty information for a specific product or product family, access Cisco Warranty Finder.

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 1

Short Description

Read Me First

· Short Description, on page 2
Short Description
The documentation set for this product strives to use bias-free language. For purposes of this documentation set, bias-free is defined as language that does not imply discrimination based on age, disability, gender, racial identity, ethnic identity, sexual orientation, socioeconomic status, and intersectionality. Exceptions may be present in the documentation due to language that is hardcoded in the user interfaces of the product software, language used based on standards documentation, or language that is used by a referenced third-party product.
Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. To view a list of Cisco trademarks, go to this URL: https://www.cisco.com/c/en/us/about/ legal/trademarks.html. Third-party trademarks mentioned are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (1721R)

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 2

2 C H A P T E R
New and Changed Information
· New and Changed Information, on page 3
New and Changed Information

Note

· For detailed information on CUBE features supported on Cisco IOS Releases, Cisco IOS XE 3S Releases,

and Cisco IOS XE Denali 16.3.1 and later Releases, refer to CUBE Cisco IOS Feature Roadmap, CUBE

Cisco IOS XE 3S Feature Roadmap, and CUBE Cisco IOS XE Releases Feature Roadmap respectively.

· For CUBE feature support information for Cisco IOS XE Bengaluru 17.6.1a and later releases, see Cisco Unified Border Element IOS-XE Configuration Guide.

· H.323 protocol is no longer supported from Cisco IOS XE Bengaluru 17.6.1a onwards. Consider using SIP for multimedia applications.

· The documentation set for this product strives to use bias-free language. For purposes of this documentation set, bias-free is defined as language that does not imply discrimination based on age, disability, gender, racial identity, ethnic identity, sexual orientation, socioeconomic status, and intersectionality. Exceptions may be present in the documentation due to language that is hardcoded in the user interfaces of the product software, language used based on RFP documentation, or language that is used by a referenced third-party product.

Description
Secure forking of nonsecure calls through Media Proxy
Support for Cisco 8200L Catalyst Edge Series Platforms
Support for VoIP Trace Serviceability Framework

Documented at CUBE Media Proxy, on page 569 Supported Platforms, on page 5 VoIP Trace for CUBE, on page 1035

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 3

New and Changed Information

New and Changed Information

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 4

Supported Platforms

3 C H A P T E R

Note Cisco Cloud Services Router 1000V Series (CSR 1000V) is no longer supported from Cisco IOS XE Bengaluru 17.4.1a onwards. If you are using CSR 1000V, you have to upgrade to Cisco Catalyst 8000V Edge Software (Catalyst 8000V). For End-of-Life information on CSR 1000V, see End-of-Sale and End-of- Life Announcement for the Select Cisco CSR 1000v Licenses.
Cisco Unified Border Element is supported on various platforms running on Cisco IOS Software Releases and Cisco IOS XE Software Releases.

Note For information on migrating from existing Cisco IOS XE 3S releases to the Cisco IOS XE Denali 16.3 release, see Cisco IOS XE Denali 16.3 Migration Guide for Access and Edge Routers

The following table provides information on Cisco router platform support for Cisco Unified Border Element:

Cisco Router Platforms

Cisco Router Models

Cisco IOS Software Releases

Cisco Integrated Cisco 2900 Series Integrated Services Services Generation Routers 2 Routers (ISR G2) Cisco 3900 Series Integrated Services
Routers

Cisco IOS 12 M and T Cisco IOS 15 M and T 1

Cisco 4000 Series Integrated Services Routers (ISR G3)

Cisco 4321 Integrated Services Routers Cisco 4331 Integrated Services Routers Cisco 4351 Integrated Services Routers

Cisco 4431 Integrated Services Routers

Cisco 4451 Integrated Services Routers

Cisco IOS XE 3S Cisco IOS XE Denali 16.3.1 onwards 2

Cisco 4461 Integrated Services Routers Cisco IOS XE Amsterdam 17.2.1r onwards

Cisco 1000 Series All router models belonging to Cisco 1100 Cisco IOS XE Gibraltar 16.12.1a onwards Integrated Services Integrated Services Routers Routers (ISR)

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 5

Supported Platforms

Cisco Router Platforms

Cisco Router Models

Cisco IOS Software Releases

Cisco Aggregated Services Routers (ASR)

Cisco ASR1001-X Aggregated Services Routers
Cisco ASR1002-X Aggregated Services Routers
Cisco ASR1004 Aggregated Services Routers with RP2
Cisco ASR1006 Aggregated Services Routers with RP2 and ESP40

Cisco IOS XE 3S Cisco IOS XE Denali 16.3.1 onwards

Cisco ASR1006-X Aggregated Services Cisco IOS XE Everest 16.6.1 onwards Routers with RP2 and ESP40

Cisco ASR1006-X Aggregated Services Cisco IOS XE Everest 16.6.1 onwards Routers with RP3 and ESP40/ESP100

Cisco ASR1006-X Aggregated Services Cisco IOS XE Amsterdam 17.3.2 onwards Routers with RP3 and ESP100X

Cisco Cloud Services Routers (CSR)

Cisco Cloud Services Router 1000V series Cisco IOS XE 3.15 onwards Cisco IOS XE Denali 16.3.1 onwards

Cisco Catalyst 8000V Edge Software (Catalyst 8000V)

Cisco Catalyst 8000V Edge Software (Catalyst 8000V)

Cisco IOS XE Bengaluru 17.4.1a onwards

Cisco 8300 Catalyst C8300-1N1S-6T

Edge Series Platforms

C8300-1N1S-4T2X

C8300-2N2S-6T

C8300-2N2S-4T2X

Cisco IOS XE Amsterdam 17.3.2

Cisco 8200 Catalyst C8200-1N-4T Edge Series Platform

Cisco IOS XE Bengaluru 17.4.1a

Cisco 8200L

C8200L-1N-4T

Catalyst Edge Series

Platform

Cisco IOS XE Bengaluru 17.5.1a

1 Support for CUBE on Cisco 2900 Series Integrated Services Routers and Cisco 3900 Series Integrated Services Routers are only up to release 15.7 M.
2 All CUBE features from release 11.5.0 (Cisco IOS XE Release 3.17) and features introduced in CUBE 11.5.1 on Cisco Integrated Services Generation 2 Routers (ISR G2) are included in CUBE release 11.5.2 for the Cisco IOS XE based platforms from Cisco IOS XE Denali 16.3.1 onwards.

· Feature Comparison on Supported Platforms , on page 7

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 6

Supported Platforms

Feature Comparison on Supported Platforms

Feature Comparison on Supported Platforms
The following table provides high level details of CUBE features supported on different platforms.

Note Collaboration feature support on Cisco ISR 4000 Series Routers is available from Cisco IOS XE Release 3.13.1S onwards. Cisco Cloud Services Routers 1000V Series support is available from Cisco IOS XE Release 3.15S onwards.

Table 1: Feature Comparisons for Supported Platforms

Features

Cisco ASR 1000 Series Routers

Cisco ISR G2 Series Routers

Cisco ISR 4000 Series Cisco ISR 1000

Routers

Series Routers

High Availability Implementation

Redundancy Group Hot Standby

Redundancy Group No

Infrastructure

Protocol (HSRP) Infrastructure

Based

Media Forking

Yes (Cisco IOS XE Yes (Cisco IOS Yes (Cisco IOS XE No

Release 3.8S

Relase 15.2 (1) T Release 3.10S

onwards)

onwards

onwards)

DSP Card Type SPA-DSP

PVDM2/PVDM3 PVDM4

No

SM-X-PVDM

Transcoder

No

registered to CUCM

Yes (Exists via SCCP)

Yes (Exists via SCCP No – Cisco IOS XE Release 3.11S onwards)

Transcoder–LTI Yes

Yes

Yes

No

Cisco UC Gateway Yes (Cisco IOS XE Yes (Cisco IOS Yes

Yes

Services API

Release 3.8S

Release 15.2(2)T

onwards)

onwards

Noise Reduction Yes

Yes (Cisco IOS Yes

No

and ASP

Release 15.2(3)T

onwards)

Call Progress Analysis

Yes

Yes

Yes

No

(Cisco IOS XE

Cisco IOS Release Recommended –

Release 3.9S onwards 15.3(2)T onwards; Cisco IOS XE

; Recommended – Recommended Release 3.15S

Cisco IOS XE

-Cisco IOS

Release 3.15S)

Release 15.5(2)T

onwards)

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 7

Feature Comparison on Supported Platforms

Supported Platforms

Features
SRTP-RTP Interworking
CUBE for SP Managed and Hosted Services Unified SRST colocation with CUBE
IPv6

Cisco ASR 1000 Series Routers
Yes – No DSP resources required (Cisco IOS XE Release 3.7S onwards)
Yes

Cisco ISR G2 Series Routers

Cisco ISR 4000 Series Cisco ISR 1000

Routers

Series Routers

Yes – DSP

Yes – No DSP

resources required resources required

(Cisco IOS Release 12.4(22)YB onwards)

Cisco IOS XE Release 3.12S onwards

Yes – No DSP resources required

Yes

Yes

Yes

Not supported Yes

SCCP SRST is supported
SIP SRST is not supported

Yes (Cisco IOS XE Fuji 16.7.1 Release onwards)

Yes. From Cisco IOS XE Bengaluru 17.5.1a

Yes

Yes

Yes

Table 2: Feature Comparisons for Supported Platforms (Contd…)

Features

Cisco CSR 1000V Cisco 8000V Cisco 8300

Cisco 8200

Cisco 8200L

Series Routers Catalyst Series Catalyst Edge Catalyst Edge Catalyst Edge

Edge Platforms Series Platforms Series Platforms Series Platforms

HA

RG

RG

RG

RG

RG

Implementation Infrastructure Infrastructure Infrastructure Infrastructure Infrastructure

Media Forking Yes

Yes

Yes

Yes

Yes

DSP Card Type No

No

NIM-PVDM NIM-PVDM NIM-PVDM

SM-X-PVDM SM-X-PVDM SM-X-PVDM

Transcoder

No

No

Yes (via SCCP) Yes (via SCCP) Yes (via SCCP)

registered to

CUCM

Transcoder–LTI No

No

Yes

Yes

Yes

Cisco UC

Yes

Yes

Yes

Yes

Yes

Gateway

Services API

Noise Reduction No

No

Yes

Yes

Yes

& ASP

Call Progress No

No

Yes

Yes

Yes

Analysis

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 8

Supported Platforms

Feature Comparison on Supported Platforms

Features

Cisco CSR 1000V Cisco 8000V Cisco 8300

Cisco 8200

Cisco 8200L

Series Routers Catalyst Series Catalyst Edge Catalyst Edge Catalyst Edge

Edge Platforms Series Platforms Series Platforms Series Platforms

SRTP-RTP Interworking

Yes – No DSP resources required
(Cisco IOS XE Release 3.15S onwards)

Yes – No DSP resources required

Yes – No DSP resources required

Yes – No DSP resources required

Yes – No DSP resources required

CUBE for SP Yes

Yes

Yes

Yes

Yes

Managed and

Hosted Services

Unified SRST Not supported No colocation with CUBE

Yes

Yes

Yes

IPv6

Yes

Yes

Yes

Yes

Yes

Note For more information on Unified SRST and Unified Border Element Co- location, see Unified SRST and Unified Border Element Co-location.
Co-location of Cisco Unified Border Element – High Availability (HA) with Unified SRST is not supported.

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 9

Feature Comparison on Supported Platforms

Supported Platforms

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 10

I P A R T
CUBE Fundamentals and Basic Setup
· Overview of Cisco Unified Border Element, on page 13 · Virtual CUBE, on page 25 · Dial-Peer Matching, on page 31 · DTMF Relay , on page 37 · Introduction to Codecs, on page 51 · Call Admission Control, on page 65 · Basic SIP Configuration, on page 83 · SIP Binding , on page 111 · Media Path, on page 127 · SIP Profiles, on page 135 · SIP Out-of-Dialog OPTIONS Ping Group, on page 163 · Configure TCL IVR Applications, on page 171 · VoIP for IPv6, on page 191 · Monitoring of Phantom Packets, on page 247 · Configurable SIP Parameters via DHCP, on page 253

4 C H A P T E R
Overview of Cisco Unified Border Element
Cisco Unified Border Element (CUBE) bridges voice and video connectivity between two separate VoIP networks. It is similar to a traditional voice gateway, except for the replacement of physical voice trunks with an IP connection. Traditional gateways connect VoIP networks to telephone companies using a circuit-switched connection, such as PRI. The CUBE connects VoIP networks to other VoIP networks and is often used to connect enterprise networks to Internet telephony service providers (ITSPs).
· Information About Cisco Unified Border Element, on page 13 · How to Configure Basic CUBE Features, on page 18
Information About Cisco Unified Border Element
Cisco Unified Border Element (CUBE) can terminate and originate signaling (H.323 and Session Initiation Protocol [SIP]) and media streams (Real-Time Transport Protocol [RTP] and RTP Control Protocol [RTCP]). CUBE extends the functionality provided by conventional session border controllers (SBCs) in terms of protocol interworking, especially on the enterprise side. As shown in the chart below, the CUBE provides the following additional features:
Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 13

Information About Cisco Unified Border Element Figure 1: Cisco Unified Border Element–More Than an SBC

CUBE Fundamentals and Basic Setup

The CUBE provides a network-to-network interface point for: · Signaling interworking–H.323 and SIP. · Media interworking–dual-tone multifrequency (DTMF), fax, modem, and codec transcoding. · Address and port translations–privacy and topology hiding. · Billing and call detail record (CDR) normalization. · Quality-of-service (QoS) and bandwidth management–QoS marking using differentiated services code point (DSCP) or type of service (ToS), bandwidth enforcement using Resource Reservation Protocol (RSVP), and codec filtering.
Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 14

CUBE Fundamentals and Basic Setup

Information About Cisco Unified Border Element

CUBE functionality is implemented on devices using a special IOS feature set, which allows CUBE to route a call from one VoIP dial peer to another.
Protocol interworking is possible for the following combinations:
· H.323-to-SIP interworking
· H.323-to-H.323 interworking
· SIP-to-SIP interworking
The CUBE provides a network-to-network demarcation interface for signaling interworking, media interworking, address and port translations, billing, security, quality of service, call admission control, and bandwidth management.
The CUBE is used by enterprise and small and medium-sized organizations to interconnect SIP PSTN access with SIP and H.323 enterprise unified communications networks.
A CUBE interoperates with several different network elements including voice gateways, IP phones, and call-control servers in many different application environments, from advanced enterprise voice and/or video services with Cisco Unified Communications Manager or Cisco Unified Communications Manager Express, as well as simpler toll bypass and voice over IP (VoIP) transport applications. The CUBE provides organizations with all the border controller functions integrated into the network layer to interconnect unified communications voice and video enterprise-to-service-provider architectures.
Figure 2: Why Does an Enterprise Need the CUBE?

If an enterprise subscribes to VoIP services offered by an ITSP, connecting the enterprise CUCM through a CUBE provides network demarcation capabilities, such as security, topology hiding, transcoding, call admission control, protocol normalization and SIP registration, none of which is possible if CUCM connects directly to the ITSP. Another use case involves mergers or acquisitions in an enterprise and the need to integrate voice
Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 15

SIP/H.323 Trunking

CUBE Fundamentals and Basic Setup

equipment, such as CUCMs, IP PBXs, VM servers, and so on. If the networks in the two organizations have overlapping IP addresses, CUBE can be used to connect the two distinct networks until the acquired organization can be migrated into the enterprise addressing plan.
SIP/H.323 Trunking
Note H.323 protocol is no longer supported from Cisco IOS XE Bengaluru 17.6.1a onwards. Consider using SIP for multimedia applications.
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over IP networks. SIP (or H.323) trunking is the use of VoIP to facilitate the connection of PBX to other VoIP endpoints across the Internet. To use SIP trunking, an enterprise must have a PBX (internal VoIP system) that connects to all internal end users, an Internet telephony service provider (ITSP), and a gateway that serves as the interface between the PBX and the ITSP. One of the most significant advantages of SIP and H.323 trunking is the ability to combine data, voice, and video in a single line, eliminating the need for separate physical media for each mode.
Figure 3: SIP/H.323 Trunking

SIP trunking overcomes TDM barriers, in that it: · Improves efficiency of interconnection between networks · Simplifies PSTN interconnection with IP end-to-end · Enables rich media services to employees, customers, and partners · Carries converged voice, video, and data traffic
Figure 4: SIP Trunking Overcomes TDM Barriers

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 16

CUBE Fundamentals and Basic Setup

Typical Deployment Scenarios for CUBE

Note For Cisco IOS XE Gibraltar 16.11.1a and later releases, the SIP processes are initiated only when either of the following CLIs is configured: · Voice dial-peer with session protocol as SIP. · voice register global · sip-ua In the releases before Cisco IOS XE Gibraltar 16.11.1a, the following commands initiated the SIP processes: · dial-peer voice (any) · ephone-dn · max-dn under call-manager-fallback · ds0-group 0 timeslots 1 type e&m-wink-start
Typical Deployment Scenarios for CUBE
CUBE in an enterprise environment serves two main purposes: · External Connections–CUBE is the demarcation point within a unified communications network and provides interconnectivity with external networks. This includes H.323 and SIP voice and video connections. · Internal Connections–When used within a VoIP network, CUBE increases flexibility and interoperability between devices.
Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 17

How to Configure Basic CUBE Features Figure 5: Typical Deployment Scenarios

CUBE Fundamentals and Basic Setup

How to Configure Basic CUBE Features
Consider a scenario where XYZ corporation uses a VoIP network to provide phone services and uses a PRI connection for telecommunications services, and the PRI trunk is controlled by MGCP. Migration from MGCP PRI to SIP trunk is provided by ITSP telecommunications. CUCM sends the telephone number, as 10 digits, to CUBE. CUCM may send only the extension (4 digits) to the CUBE. When the call is diverted (using call-forward), the requirement of the ITSP is that they need the full 10-digit number in the SIP Diversion field.
Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 18

CUBE Fundamentals and Basic Setup Figure 6: CUBE Configuration Workflow

Enabling the CUBE Application on a Device

The following sections describe the basic setup of CUBE through the steps involved in migrating the XYZ corporation to CUBE using a SIP trunk.

Enabling the CUBE Application on a Device

SUMMARY STEPS

1. enable 2. configure terminal 3. voice service voip 4. mode border-element license [capacity sessions | periodicity {mins value | hours value | days value}] 5. allow-connections from-type to to-type 6. end

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 19

Enabling the CUBE Application on a Device

CUBE Fundamentals and Basic Setup

DETAILED STEPS

Step 1

Command or Action enable Example:

Purpose
Enables privileged EXEC mode. Enter your password if prompted.

Step 2

Device> enable
configure terminal Example:

Enters global configuration mode.

Step 3

Device# configure terminal
voice service voip Example:

Enters global VoIP configuration mode.

Step 4

Device(config)# voice service voip

mode border-element license [capacity sessions | periodicity {mins value | hours value | days value}]

Enables CUBE configuration and configures the number of licenses (capacity).

Example:
Device(conf-voi-serv)# mode border-element license capacity 200
Device(conf-voi-serv)# mode border-element license periodicity days 15

· Effective from Cisco IOS XE Amsterdam 17.2.1r, the capacity keyword and sessions argument are deprecated. However, the keyword and argument are available in the Command Line Interface (CLI). If you try to configure license capacity using CLI, the following error message is displayed:

Error: CUBE SIP trunk licensing is now based on dynamic session counting. Static
license capacity configuration has been deprecated.
· Effective from Cisco IOS XE Amsterdam 17.2.1r, the periodicity keyword and [mins | hours| days] argument are introduced. The periodicity keyword configures periodicity interval for license entitlement requests for CUBE. If you do not configure license periodicity, the default license period of 7 days is enabled.

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 20

CUBE Fundamentals and Basic Setup

Verifying the CUBE Application on the Device

Command or Action

Purpose Note

We recommend you to configure interval in days. Configuring interval in minutes or hours increases the frequency of entitlement requests and thereby increases the processing load on Cisco Smart Software Manager (CSSM). License periodicity configuration of minutes or hours is recommended to be used only with Cisco Smart Software Manager On-Prem (formerly known as Cisco Smart Software Manager satellite) mode.

Step 5 Step 6

allow-connections from-type to to-type Example:
Device(conf-voi-serv)# allow-connections sip to sip

Allows connections between specific types of endpoints in a VoIP network.
· The two protocols (endpoints) refer to the VoIP protocols (SIP or H.323) on the two call legs.

end Example:

Returns to privileged EXEC mode.

Device(conf-voi-serv)# end

Verifying the CUBE Application on the Device

SUMMARY STEPS

1. enable 2. show cube status

DETAILED STEPS

Step 1

enable Enables privileged EXEC mode. Example: Device> enable

Step 2

show cube status
Displays the CUBE status, the software version, the license capacity, the image version, and the platform name of the device. In releases before Cisco IOS XE Amsterdam 17.2.1r, CUBE status display is enabled only if mode border- element command is configured with call license capacity. Effective from Cisco IOS XE Amsterdam 17.2.1r, this dependency is removed and Licensed-Capacity information is excluded from output.
Example:

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 21

Configuring a Trusted IP Address List for Toll-Fraud Prevention

CUBE Fundamentals and Basic Setup

Before Cisco IOS XE Amsterdam 17.2.1r:
Device# show cube status
CUBE-Version : 12.5.0 SW-Version : 16.11.1, Platform CSR1000V HA-Type : none Licensed-Capacity : 10 Calls blocked (Smart Licensing Not Configured) : 0 Calls blocked (Smart Licensing Eval Expired) : 0
Effective from Cisco IOS XE Amsterdam 17.2.1r:
Device# show cube status
CUBE-Version : 12.8.0 SW-Version : 17.2.1, Platform CSR1000V HA-Type : none

Configuring a Trusted IP Address List for Toll-Fraud Prevention

SUMMARY STEPS

1. enable 2. configure terminal 3. voice service voip 4. ip address trusted list 5. ipv4 ipv4-address [network-mask] 6. ipv6 ipv6-address 7. end

DETAILED STEPS

Step 1

Command or Action enable Example:
Device> enable

Step 2

configure terminal Example:
Device# configure terminal

Step 3

voice service voip Example:
Device(config)# voice service voip

Step 4

ip address trusted list Example:
Device(conf-voi-serv)# ip address trusted list

Purpose Enables privileged EXEC mode.
· Enter your password if prompted. Enters global configuration mode.
Enters global VoIP configuration mode.
Enters IP address trusted list mode and enables the addition of valid IP addresses.

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 22

CUBE Fundamentals and Basic Setup

Configuring a Trusted IP Address List for Toll-Fraud Prevention

Step 5 Step 6 Step 7

Command or Action ipv4 ipv4-address [network-mask] Example:
Device(cfg-iptrust-list)# ipv4 192.0.2.1 255.255.255.0
ipv6 ipv6-address Example:
Device(cfg-iptrust-list)# ipv6 2001:DB8:0:ABCD::1/48
end Example:
Device(cfg-iptrust-list)# end

Purpose Allows you to add up to 100 IPv4 addresses in the IP address trusted list. Duplicate IP addresses are not allowed.
· The network-mask argument allows you to define a subnet IP address.
Allows you to add IPv6 addresses to the trusted IP address list.
Returns to privileged EXEC mode.

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 23

Configuring a Trusted IP Address List for Toll-Fraud Prevention

CUBE Fundamentals and Basic Setup

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 24

5 C H A P T E R
Virtual CUBE
The Cisco Unified Border Element (CUBE) feature set has traditionally been delivered with hardware router platforms, such as the Cisco Integrated Services Router (ISR) series. A subset of CUBE features (vCUBE) may be used in virtualized environments with the Cisco CSR 1000v Series Cloud Services Router or Cisco Catalyst 8000V Edge Software (Catalyst 8000V).

Note When upgrading to Catalyst 8000V software from a CSR1000V release, an existing throughput configuration will be reset to a maximum of 250 Mbps. Install an HSEC authorization code, which you can obtain from your Smart License account, before reconfiguring your required throughput level.
· Feature Information for Virtual CUBE, on page 25 · Prerequisites for Virtual CUBE, on page 26 · Features Supported with Virtual CUBE , on page 27 · Restrictions, on page 27 · Information about Virtual CUBE, on page 27 · Install Virtual CUBE on ESXi, on page 28 · How to Enable Virtual CUBE , on page 29 · Troubleshooting Virtual CUBE, on page 29

Feature Information for Virtual CUBE

The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Table 3: Feature Information for Virtual CUBE Support

Feature Name

Releases

Feature Information

Virtual CUBE in Cisco Catalyst Cisco IOS XE Bengaluru Virtual CUBE introduced for Cisco Catalyst

8000V Edge Software (Catalyst 17.4.1a

8000V Edge Software (Catalyst 8000V) in

8000V)

VMware ESXi and AWS environments.

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 25

Prerequisites for Virtual CUBE

CUBE Fundamentals and Basic Setup

Feature Name
vCUBE in Amazon Web Services (AWS)
Virtual CUBE

Releases

Feature Information

Cisco IOS XE Gibraltar vCUBE offer introduced in AWS for Cisco CSR

16.12.4a

1000v Series Cloud Services Router.

Cisco IOS XE 3.15S

Virtual CUBE introduced for Cisco CSR 1000v Series Cloud Services Router in VMware ESXi environments.

Prerequisites for Virtual CUBE

Hardware

· The vCUBE feature set is bundled as part of the Cisco virtual router software and is used when deployed in VMware ESXi virtualized environments. For more information on how to deploy Cisco virtualized routers in VMware ESXi environments, see Installing the Cisco CSR 1000V in VMware ESXi Environments and Installing in VMware ESXi Environment.
· For information on the best practices for setting ESXi host BIOS parameters for performance, see BIOS Settings.
· Virtual CUBE is supported on the CSR 1000V and C8000V platforms.
· Virtual CUBE is also supported in AWS. You must use the AWS Marketplace product listing for virtual CUBE.
· For more information about the Cisco CSR 1000V in AWS, see Cisco CSR 1000V Series Cloud Services Router Deployment Guide for Amazon Web Services.

Note

· The CSR1000V and Catalyst 8000V product may be used in several different public and private cloud

environments. However, vCUBE is only supported when deployed on VMware ESXi and AWS platforms

currently.

· When you use a consolidated (.bin) image to upgrade a CSR 1000V medium configuration (2 vCPU, 4 GB RAM) to Catalyst 8000V, you must change the virtual machine vRAM allocation to at least 5 GB to ensure advertised performance. Alternatively and when deploying in AWS environments, boot the router using individual packages rather than a consolidated image without the need for additional memory. Refer to Installing Subpackages from a Consolidated Package for details.

Software

· Obtain the relevant license for the router platform. See Virtual CUBE Licensing Requirements , on page 28 for more information.
· In AWS, only Bring Your Own License (BYOL) is supported for vCUBE. Pay as You Go (Subscription) versions of the CSR 1000V and C8000V are not supported. Make sure you choose the vCUBE AWS Marketplace product listing. Refer to Cisco Virtual CUBE-BYOL.

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 26

CUBE Fundamentals and Basic Setup

Features Supported with Virtual CUBE

· For more information about Cisco virtual routers, see CSR 1000V Data Sheet and Catalyst 8000V Data Sheet.
Features Supported with Virtual CUBE
vCUBE supports most of the CUBE features available in IOS XE releases. vCUBE does not support the following:
· DSP-based features · Codec Transcoding, Transrating · Raw Inband to RTP-NTE DTMF Interworking · Call progress Analysis (CPA) · Noise Reduction (NR), Acoustic Shock Protection (ASP), and Audio Gain
· H.323 Interworking · IOS-based Hardware Media Termination Point (MTP)

Note CUBE high availability is not currently supported on vCUBE when deployed in AWS.
Restrictions
· Software MTP is not supported. · CSR1000V used as MTP/TRP for CUCM is not supported.

Note All caveats, restrictions, and limitations of Cisco ASR IOS-XE 3.15 and later releases are applicable to virtual CUBE.

Information about Virtual CUBE

Media

vCUBE media performance depends on the underlying host platform consistently providing packet switching latency of less than 5 milliseconds. The recommended hardware and virtual machine configurations ensure this performance when followed closely.
For more information on how to monitor media performance, see Voice Quality Monitoring.

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 27

Virtual CUBE Licensing Requirements

CUBE Fundamentals and Basic Setup

Virtual CUBE Licensing Requirements
For information about licensing of virtual CUBE with CSR1000V and C8000V, refer to CUBE Smart Licensing.

Virtual CUBE with CSR1000V

vCUBE is enabled for the CSR1000V with the APPX and AX platform licenses. vCUBE processes and CLI commands are enabled when either of these licenses are enabled. Secure call features require the AX license. In common with all CUBE instances, L-CUBE Smart License options are required for each active session.
The following table details the license requirements for Virtual CUBE on the CSR1000V.

Virtual CUBE Session License

Platform License

Features

Throughput License

L-CUBE Smart License APPX options
AX

No TLS / SRTP support Session count * (signaling

All vCUBE features

+ bidirectional media bandwidth)

For detailed information about licensing, see Cisco CSR 1000v Software Configuration Guide.

Virtual CUBE with Catalyst 8000V

vCUBE is enabled for the Catalyst 8000V with the DNA Network Essentials license.

Virtual CUBE Session License

DNA Subscription

Features

DNA Bandwidth License

L-CUBE Smart License Essentials or above options

All vCUBE features

Session count * (signaling + bidirectional media bandwidth)/2

For detailed information on licensing, see Licensing.

Install Virtual CUBE on ESXi

SUMMARY STEPS

1. Use the CSR1000V or the Catalyst 8000V OVA application file (available from software.cisco.com) to deploy a new virtual instance directly in VMware ESXi.

DETAILED STEPS

Step 1

Command or Action

Purpose

Use the CSR1000V or the Catalyst 8000V OVA application Note

Select the required instance size during the

file (available from software.cisco.com) to deploy a new

OVA deployment.

virtual instance directly in VMware ESXi.

For further details on how to perform the deployment, see

Cisco CSR 1000V Series Cloud Services Router Software

Configuration Guide or Cisco Catalyst 8000V Edge

Software Installation And Configuration Guide.

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 28

CUBE Fundamentals and Basic Setup

How to Enable Virtual CUBE

How to Enable Virtual CUBE

SUMMARY STEPS

1. Power on the virtual machine. 2. Enable platform and throughput licenses and register to a Cisco licensing server. 3. Enable virtual CUBE using the steps in Enabling the CUBE Application on a Device.

DETAILED STEPS

Step 1

Command or Action Power on the virtual machine.

Purpose Powers on the vCUBE.

Step 2

Enable platform and throughput licenses and register to a Enables platform and throughput licenses and registers that

Cisco licensing server.

virtual CUBE to a licensing server.

Step 3

Enable virtual CUBE using the steps in Enabling the CUBE Enables vCUBE on a device. Application on a Device.

Troubleshooting Virtual CUBE
To troubleshoot vCUBE, follow the same procedure for Cisco ASR routers. This procedure includes crash file decoding, decoding traceback, and so on. For more details, see Troubleshoot Cisco ASR 1000 Series Aggregation Services Routers Crashes.
To troubleshoot virtual machine issues, see Cisco CSR 1000V Series Cloud Services Router Software Configuration Guide and Cisco Catalyst 8000V Edge Software Configuration Guide.

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 29

Troubleshooting Virtual CUBE

CUBE Fundamentals and Basic Setup

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 30

6 C H A P T E R
Dial-Peer Matching
CUBE allows VoIP-to-VoIP connection by routing calls from one VoIP dial peer to another. As VoIP dial peers can be handled by either SIP or H.323, CUBE can be used to interconnect VoIP networks of different signaling protocols. VoIP interworking is achieved by connecting an inbound dial peer with an outbound dial peer.
Note All CUBE Enterprise deployments must have signaling and media bind statements specified at the dial-peer or voice class tenant level. For voice call tenants, you must apply tenants to dial-peers used for CUBE call flows if these dial-peers do not have bind statements specified.
· Dial Peers in CUBE, on page 31 · Configuring Inbound and Outbound Dial-Peer Matching for CUBE, on page 33 · Preference for Dial-Peer Matching, on page 34
Dial Peers in CUBE
A dial peer is a static routing table, mapping phone numbers to interfaces or IP addresses. A call leg is a logical connection between two routers or between a router and a VoIP endpoint. A dial peer is associated or matched to each call leg according to attributes that define a packet-switched network, such as the destination address. Voice-network dial peers are matched to call legs based on configured parameters, after which an outbound dial peer is provisioned to an external component using the component’s IP address. For more information, refer to the Dial Peer Configuration Guide. Dial-peer matching can also be done based on the VRF ID associated with a particular interface. For more information, see Inbound Dial-Peer Matching Based on Multi-VRF, on page 359. In CUBE, dial peers can also be classified as LAN dial peers and WAN dial peers based on the connecting entity from which CUBE sends or receives calls.
Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 31

Dial Peers in CUBE Figure 7: LAN and WAN Dial Peers

CUBE Fundamentals and Basic Setup

A LAN dial peer is used to send or receive calls between CUBE and the Private Branch Exchange (PBX)–a system of telephone extensions within an enterprise. Given below are examples of inbound and outbound LAN dial peers.
Figure 8: LAN Dial Peers

A WAN dial peer is used to send or receive calls between CUBE and the SIP trunk provider. Given below are examples of inbound and outbound WAN dial peers.
Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 32

CUBE Fundamentals and Basic Setup Figure 9: WAN Dial Peers

Configuring Inbound and Outbound Dial-Peer Matching for CUBE

Configuring Inbound and Outbound Dial-Peer Matching for CUBE

The following commands can be used for inbound and outbound dial peer matching in the CUBE:
Table 4: Incoming Dial-Peer Matching

Command in Dial-Peer Configuration
incoming called-number DNIS-string

Description

Call Setup Element

This command uses the destination number that was called DNIS number to match the incoming call leg to an inbound dial peer. This number is called the dialed number identification service (DNIS) number.

answer-address ANI-string

This command uses the calling number to match the

ANI string

incoming call leg to an inbound dial peer. This number is

called the originating calling number or automatic number

identification (ANI) string.

destination-pattern ANI-string

This command uses the inbound call leg to the inbound ANI string for

dial peer.

inbound

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 33

Preference for Dial-Peer Matching

CUBE Fundamentals and Basic Setup

Command in Dial-Peer Configuration

Description

Call Setup Element

{incoming called | incoming This command uses a group of incoming called (DNIS) or E.164 Patterns

calling} e164-pattern-map incoming calling (ANI) number patterns to match the

pattern-map-group-id

inbound call leg to an inbound dial peer.

The command calls a globally defined voice class identifier where the E.164 pattern groups are configured.

voice class uri

This command uses the directory URI (Uniform Resource Directory URI

URI-class-identifier with Identifier) number of an incoming INVITE from a SIP

incoming uri {from | request entity to match an inbound dial peer. This directory URI

| to | via} URI-class-identifier is part of the SIP address of a device.

The command calls a globally defined voice class identifier where the directory URI is configured. It requires the configuration of session protocol sipv2

incoming uri {called |

This command uses the directory URI (Uniform Resource Directory URI

callling} URI-class-identifier Identifier) number to match the outgoing H.323 call leg to

an outgoing dial peer.

The command calls a globally defined voice class identifier where the directory URI is configured.

Table 5: Outgoing Dial-Peer Matching

Dial-Peer Command destination-pattern DNIS-string
destination URI-class-identifier
destination e164-pattern-map pattern-map-group-id

Description

Call Setup Element

This command uses DNIS string to match the outbound DNIS string for

call leg to the outbound dial peer.

outbound

ANI string for inbound

This command uses the directory URI (Uniform Resource Directory URI Identifier) number to match the outgoing call leg to an outgoing dial peer. This directory URI is part of the SIP address of a device.
The command actually refers to a globally defined voice class identifier where the directory URI is configured.

This command uses a group of destination number

E.164 patterns

patterns to match the outbound call leg to an outbound

dial peer.

The command calls a globally defined voice class identifier where the E.164 pattern groups are configured.

Preference for Dial-Peer Matching
The following is the order in which inbound dial-peer is matched for SIP call- legs:

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CUBE Fundamentals and Basic Setup

Preference for Dial-Peer Matching

· voice class uri URI-class-identifier with incoming uri {via} URI-class- identifier · voice class uri URI-class-identifier with incoming uri {request} URI-class-identifier · voice class uri URI-class-identifier with incoming uri {to} URI-class-identifier · voice class uri URI-class-identifier with incoming uri {from} URI-class-identifier · incoming called-number DNIS-string · answer- address ANI-string
The following is the order in which inbound dial-peer is matched for H.323 call-legs: · incoming uri {called} URI-class-identifier · incoming uri {callling} URI-class-identifier · incoming called-number DNIS-string · answer- address ANI-string
The following is the order in which outbound dial-peer is matched for SIP call-legs: · destination route-string · destination URI-class-identifier with target carrier-id string · destination-pattern with target carrier-id string · destination URI-class-identifier · destination-pattern · target carrier-id string
Note If CUBE with Cisco Unified Communications Manager Express (CUCME) is configured with the same DNs, then the ANI is given the preference. The system dial-peer for the DN is selected over the other dial-peers created.

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Preference for Dial-Peer Matching

CUBE Fundamentals and Basic Setup

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 36

7 C H A P T E R

DTMF Relay

The DTMF Relay feature allows CUBE to send dual-tone multi-frequency (DTMF) digits over IP.
This chapter talks about DTMF tones, DTMF relay mechanisms, how to configure DTMF relays, and interoperability and priority with multiple relay methods.
· Feature Information for DTMF Relay , on page 37 · Information About DTMF Relay , on page 38 · Verifying DTMF Relay , on page 46

Feature Information for DTMF Relay

The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.
Table 6: Feature Information for DTMF Relay

Feature Name

Releases

Feature Information

DTMF Relay

Cisco IOS Release 12.1(2)T The DTMF relay feature allows CUBE to send

Cisco IOS XE 2.1

DTMF digits over IP.

The dtmf-relay command was added.

Support for sip-info to rtp-nte Cisco IOS XE Everest 16.6.1 This feature adds support for sip-info to

DTMF relay mechanism for

rtp-nte DTMF relay mechanism for SIP-SIP

SIP-SIP calls

calls.

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Information About DTMF Relay

CUBE Fundamentals and Basic Setup

Information About DTMF Relay
DTMF Tones
DTMF tones are used during a call to signal to a far-end device; these signals may be for navigating a menu system, entering data, or for other types of manipulation. They are processed differently from the DTMF tones that are sent during the call setup as part of the call control. TDM interfaces on Cisco devices support DTMF by default. Cisco VoIP dial-peers do not support the DTMF relay by default and to enable, requires DTMF relay capabilities.
Note DTMF tones that are sent by phones do not traverse the CUBE.
DTMF Relay
Dual-tone multifrequency (DTMF) relay is the mechanism for sending DTMF digits over IP. The VoIP dial peer can pass the DTMF digits either in the band or out of band. In-band DTMF-Relay passes the DTMF digits using the RTP media stream. It uses a special payload type identifier in the RTP header to distinguish DTMF digits from actual voice communication. This method is more likely to work on lossless codecs, such as G.711.
Note The main advantage of DTMF relay is that in-band DTMF relay sends low- bandwidth codecs such as the G.729 and G.723 with greater fidelity. Without the use of DTMF relay, calls established with low-bandwidth codecs has trouble accessing automated DTMF-based systems. For example, voicemail, menu-based Automatic Call Distributor (ACD) systems, and automated banking systems.
Out-of-band DTMF-Relay passes DTMF digits using a signaling protocol (SIP or H.323) instead of using the RTP media stream. The VoIP compressed code causes the loss of integrity of the DTMF digits. However, the DTMF relay prevents the loss of integrity of DTMF digits. The relayed DTMF regenerates transparently on the peer side.
Figure 10: DTMF Relay Mechanism

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DTMF Relay

The following lists the DTMF relay mechanisms that support the VoIP dial-peers based on the configured keywords. The DTMF relay mechanism can be either out- of-band (H.323 or SIP) or in-band (RTP).
· h245-alphanumeric and h245-signal–These two methods are available only on H.323 dial peers. It is an out-of-band DTMF relay mechanism that transports the DTMF signals using H.245, which is the media control protocol of the H.323 protocol suite.
The H245-signal method carries more information about the DTMF event (such as its actual duration) than the H245-Alphanumeric method. It addresses a potential problem with the alphanumeric method when interworking with other vendors’ systems.
· sip-notify–This method is available on the SIP dial peers only. It is a Cisco proprietary out-of-band DTMF relay mechanism that transports DTMF signals using SIP-Notify message. The SIP Call-Info header indicates the use of the SIP-Notify DTMF relay mechanism. Acknowledging the message with a 18x or 200 response message containing a similar SIP Call-Info header.
The Call-Info header for a NOTIFY-based out-of-band relay is as follows:
Call-Info: ; method=”NOTIFY;Event=telephone-event;Duration=msec”
DTMF relay digits are a 4 bytes in binary encoded format.
The mechanism is useful for communicating with SCCP IP phones that do not support in-band DTMF digits and analog phones that are attached to analog voice ports (FXS) on the router.
If multiple DTMF relay mechanisms enable and negotiate successfully on a SIP dial peer, NOTIFY-based out-of-band DTMF relay takes precedence.
· sip-kpml–This method is available only on SIP dial peers. The RFC 4730 defines the out-of-band DTMF relay mechanism to register the DTMF signals using the SIP-Subscribe messages. It transports the DTMF signals using the SIP-Notify messages containing an XML-encoded body. This method is calked Key Press Markup Language.
If you configure KPML on the dial peer, the gateway sends INVITE messages with KPML in the Allow-Events header.
A registered SIP endpoint to Cisco Unified Communications Manager or Cisco Unified Communications Manager Express uses this method. This method is useful for non-conferencing calls and for interoperability between SIP products and SIP phones.
If you configure rtp-nte, sip-notify, and sip-kmpl, the outgoing INVITE contains an SDP with a rtp-nte payload, a SIP Call-Info header, and an Allow- Events header with KPML.
The following SIP-Notify message displays after the subscription. The endpoints transmit digits using SIP-Notify messages with KPML events through XML. The following example transmits, the digit “1”:
NOTIFY sip:192.168.105.25:5060 SIP/2.0 Event: kpml <?xml version=”1.0″ encoding=”UTF-8″?> <kpml-response version=”1.0″ code=”200″ text=”OK” digits=”1″ tag=”dtmf”/>
· sip-info–The sip-info method is available only on SIP dial peers. It is an out-of-band DTMF relay mechanism that registers the DTMF signals using SIP- Info messages. The body of the SIP message consists of signaling information and uses the Content-Type application/dtmf-relay.
The method enables for SIP dial peers, and invokes on receiving a SIP INFO message with DTMF relay content.
The gateway receives the following sample SIP INFO message with specifics about the DTMF tone. The combination of the From, To, and Call-ID headers identifies the call leg. The signal and duration

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DTMF Relay

CUBE Fundamentals and Basic Setup

headers specify the digit, in this case 1, and duration, 160 milliseconds in the example, for DTMF tone play.
INFO sip:2143302100@172.17.2.33 SIP/2.0 Via: SIP/2.0/UDP 172.80.2.100:5060 From: <sip:9724401003@172.80.2.100>;tag=43 To: <sip:2143302100@172.17.2.33>;tag=9753.0207 Call-ID: 984072_15401962@172.80.2.100 CSeq: 25634 INFO Supported: 100rel Supported: timer Content-Length: 26 Content-Type: application/dtmf-relay Signal= 1 Duration= 160
· rtp-nte–Real-Time Transport Protocol (RTP) Named Telephone Events (NTE). The RFC2833 defines the in-band DTMF relay mechanism. RFC2833 defines the formats of NTE-RTP packets to transport DTMF digits, hookflash, and other telephony events between two peer endpoints. Using the RTP stream, sends the DTMF tones as packet data after establishing call media. It is differentiated from the audio by the RTP payload type field, preventing compression of DTMF-based RTP packets. For example, sending the audio of a call on a session with an RTP payload type identifies it as G.711 data. Similarly sending the DTMF packets with an RTP payload type identifies them as NTEs. The consumer of the stream utilizes the G.711 packets and the NTE packets separately.
The SIP NTE DTMF relay feature provides a reliable digit relay between Cisco VoIP gateways on using a low-bandwidth codec.
Note By default, Cisco device uses Payload type 96 and 97 for fax. A third- party device may use Payload type 96 and 97 for DTMF. In such scenarios, we recommend you to perform one of the following:
· Change the Payload type for fax in both incoming and outgoing dial-peers using rtp payload-type command
· Use assymetric payload dtmf command
For more information on configuring rtp payload-type and assymetric payload DTMF, see Dynamic Payload Type Interworking for DTMF and Codec Packets for SIP-to-SIP Calls.
Payload types and attributes of this method negotiate between the two ends at call setup. They use the Session Description Protocol (SDP) within the body section of the SIP message.
Note This method is not similar to the “Voice in-band audio/G711” transport. The latter is just the audible tones being passed as normal audio without any relay signaling method being “aware” or involved in the process. It is plain audio passing through end-to-end using the G711Ulaw/Alaw codec.
· cisco-rtp–It is an in-band DTMF relay mechanism that is Cisco proprietary, where the DTMF digits are encoded differently from the audio and are identified as Payload type 121. The DTMF digits are part

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Configuring DTMF Relays

of the RTP data stream and distinguished from the audio by the RTP payload type field. Cisco Unified Communications Manager does not support this method.

Note The cisco-rtp operates only between two Cisco 2600 series or Cisco 3600 series devices. Otherwise, the DTMF relay feature does not function, and the gateway sends DTMF tones in-band.
· G711 audio–It is an in-band DTMF relay mechanism that is enabled by default and requires no configuration. Digits are transmitted within the audio of the phone conversation, that is, it is audible to the conversation partners; therefore, only uncompressed codecs like g711 Alaw or mu-law can carry in-band DTMF reliably. Female voices sometimes trigger the recognition of a DTMF tone.
The DTMF digits pass like the rest of your voice as normal audio tones with no special coding or markers. It uses the same codec as your voice, generated by your phone.

Configuring DTMF Relays
You can configure the DTMF relay using the dtmf-relay method1 […[method6]] command in the VoIP dial peer. Perform DTMF negotiation based on the matching inbound dial-peer configuration. Use any of the following variablesmethod:
· h245-alphanumeric · h245-signal · sip-notify · sip-kpml · sip-info · rtp-nte [digit-drop] · ciso-rtp

Configure multiple DTMF methods on CUBE simultaneously in order to minimize MTP requirements. If you configure more than one out-of-band DTMF method, preference goes from highest to lowest in the order of configuration. If an endpoint does not support any of the configured DTMF relay mechanisms on CUBE, an MTP or transcoder is required.
The following table lists the supported DTMF relay types on a SIP and H.322 gateway.
Table 7: Supported H.323 and SIP DTMF Relay Methods

In-band Out-of-band

H.323 Gateway

SIP Gateway

cisco-rtp, rtp-nte

rtp-nte

h245-alphanumeric, h245-signal sip-notify, sip-kpml, sip-info

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CUBE Fundamentals and Basic Setup

Interoperability and Priority with Multiple DTMF Relay Methods
· CUBE negotiates both rtp-nte and sip-kmpl if both support and advertise in the incoming INVITE. However, If CUBE does not initiate sip-kmpl, CUBE relies on the rtp-nte DTMF method to receive digits and a SUBSCRIBE. CUBE still accepts SUBSCRIBEs for KPML. It prevents double-digit reporting problems at CUBE.
· CUBE negotiates to one of the following: · cisco-rtp · rtp-nte · rtp-nte and kpml · kpml · sip-notify
· If you configure rtp-nte, sip-notify, and sip-kpml, the outgoing INVITE contains a SIP Call-Info header, an Allow-Events header with KPML, and an SDP with rtp-nte payload.
· If you configure more than one out-of-band DTMF method, preference goes from highest to lowest in the order of configuration.
· CUBE selects DTMF relay mechanisms using the following priority: · sip- notify or sip-kpml (highest priority) · rtp-nte · None–Send DTMF in-band
H.323 gateways select DTMF relay mechanisms using the following priority: · cisco-rtp · h245-signal · h245-alphanumeric · rtp-nte · None–Send DTMF in-band
DTMF Interoperability Table
This table provides the DTMF interoperability information between various DTMF relay types in different call flow scenarios. For instance, refer table 3 if you must configure sip-kpml on an inbound dial peer and h245-signaling on an outbound dial peer in an RTP-RTP Flow through configuration. The table shows that the combination supports (as image information is present) the required image IOS 12.4(15)T or IOS XE or above. The following are the call scenarios provided:
· RTP-RTP Flow-Through · RTP-RTP with transcoder Flow-Through

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DTMF Interoperability Table

· RTP-RTP Flow Around · RTP-RTP with high-density transcoder Flow Through · SRTP-RTP Flow Through

Table 8: RTP-RTP Flow-Through

Outbound H.323

SIP

dial-peer

protocol

In-band

Inbound DTMF h245- h245dial-peer Relay Type alphanumeric signal protocol

Rtp-nte Rtp-nte Sip-kpml Sipnotify

Sip-info Voice in-band (G.711)

H.323

h245-alpha Supported numeric

Supported Supported Supported Supported

h245-signal

Supported Supported Supported Supported Supported

rtp-nte Supported Supported Supported Supported

Supported

Supported*

SIP

rtp-nte Supported Supported Supported Supported Supported Supported

Supported*

sip-kpml Supported Supported

Supported Supported

sip-notify Supported Supported Supported Supported

Supported

sip-info

Supported
3

In-band Voice in-band (G.711)

Supported Supported

Supported

3 Supported from Cisco IOS XE Everest 16.6.1 onwards for calls that do not involve DSP resources.

• media resource is required (Transcoder) for IOS versions.

Table 9: RTP-RTP with DSP Involved Flow-Through Calls

Outbound H.323

SIP

dial-peer

protocol

In-band

Inbound DTMF

h245- h245-

dial-peer Relay Type alphanumeric signal

protocol

Rtp-nte Rtp-nte Sip-kpml Sipnotify

Sip-info Voice in-band (G.711)

H.323

h245-alpha Supported numeric

Supported Supported Supported Supported

h245-signal

Supported Supported Supported Supported Supported

rtp-nte Supported Supported Supported Supported

Supported

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DTMF Interoperability Table

CUBE Fundamentals and Basic Setup

Outbound H.323

SIP

dial-peer

protocol

In-band

Inbound DTMF

h245- h245-

dial-peer Relay Type alphanumeric signal

protocol

Rtp-nte Rtp-nte Sip-kpml Sipnotify

Sip-info Voice in-band (G.711)

SIP

rtp-nte Supported Supported Supported Supported

Supported

sip-kpml Supported Supported

Supported

sip-notify Supported Supported Supported

Supported

sip-info
In-band Voice in-band (G.711)

Supported Supported

Table 10: RTP-RTP Flow Around

Outbound H.323

SIP

dial-peer

protocol

In-band

Inbound DTMF

h245- h245-

dial-peer Relay Type alphanumeric signal

protocol

Rtp-nte Rtp-nte Sip-kpml Sipnotify

Sip-info Voice in-band (G.711)

H.323

h245-alpha Supported numeric

h245-signal

Supported

rtp-nte

Supported

Supported*

SIP

rtp-nte

Supported

Supported*

sip-kpml

Supported

sip-notify

Supported

sip-info
In-band Voice in-band (G.711)

Supported Supported

Supported

• media resource is required (Transcoder) for IOS versions. CUBE falls back to flow-through mode if the media resource is unavailable.

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DTMF Interoperability Table

Table 11: RTP-RTP with High-Density Transcoder Flow Through

Outbound H.323

SIP

dial-peer

protocol

In-band

Inbound DTMF

h245- h245-

dial-peer Relay Type alphanumeric signal

protocol

Rtp-nte Rtp-nte Sip-kpml Sipnotify

Sip-info Voice in-band (G.711)

H.323

h245-alpha Supported numeric

h245-signal

Supported

Supported Supported Supported Supported

rtp-nte

Supported Supported

Supported

SIP

rtp-nte

Supported Supported Supported

Supported

sip-kpml Supported Supported

Supported

sip-notify Supported Supported

Supported

sip-info
In-band Voice in-band (G.711)

Supported Supported

Table 12: SRTP-RTP Flow Through

Outbound H.323 dial-peer protocol

Inbound DTMF

h245- h245-

dial-peer Relay Type alphanumeric signal

protocol

H.323 SIP

h245-alpha numeric h245-signal rtp-nte rtp-nte

sip-kpml

sip-notify

sip-info
In-band Voice in-band (G.711)

SIP

In-band

Rtp-nte Rtp-nte Sip-kpml Sipnotify

Sip-info Voice in-band (G.711)

Supported Supported Supported

Supported Supported

Supported

Supported

Supported

Supported Supported

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Verifying DTMF Relay

CUBE Fundamentals and Basic Setup

Note For calls sent from an in-band (RTP-NTE) to an out-of band method, configure the dtmf-relay rtp-nte digit-drop command on the inbound dial-peer and the desired out-of-band method on the outgoing dial-peer. Otherwise, send the same digit in OOB and in-band, and gets interpreted as duplicate digits by the receiving end. On configuring the digit-drop option on the inbound leg, CUBE suppresses NTE packets and configures only relay digits using the OOB method on the outbound leg.

Verifying DTMF Relay

SUMMARY STEPS

1. show sip-ua calls 2. show sip-ua calls dtmf-relay sip-info 3. show sip-ua history dtmf-relay kpml 4. show sip-ua history dtmf-relay sip-notify

DETAILED STEPS

Step 1

show sip-ua calls The following sample output shows that the DTMF method is SIP-KPML. Example:

Device# show sip-ua calls

SIP UAC CALL INFO

Call 1

SIP Call ID

: 57633F68-2BE011D6-8013D46B-B4F9B5F6@172.18.193.251

State of the call

: STATE_ACTIVE (7)

Substate of the call : SUBSTATE_NONE (0)

Calling Number

:

Called Number

: 8888

Bit Flags

: 0xD44018 0x100 0x0

CC Call ID

:6

Source IP Address (Sig ): 192.0.2.1

Destn SIP Req Addr:Port : 192.0.2.2:5060

Destn SIP Resp Addr:Port: 192.0.2.3:5060

Destination Name

: 192.0.2.4.250

Number of Media Streams : 1

Number of Active Streams: 1

RTP Fork Object

: 0x0

Media Mode

: flow-through

Media Stream 1

State of the stream

: STREAM_ACTIVE

Stream Call ID

:6

Stream Type

: voice-only (0)

Negotiated Codec

: g711ulaw (160 bytes)

Codec Payload Type

:0

Negotiated Dtmf-relay : sip-kpml

Dtmf-relay Payload Type : 0

Media Source IP Addr:Port: 192.0.2.5:17576

Media Dest IP Addr:Port : 192.0.2.6:17468

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Step 2

Orig Media Dest IP Addr:Port : 0.0.0.0:0 Number of SIP User Agent Client(UAC) calls: 1 SIP UAS CALL INFO Number of SIP User Agent Server(UAS) calls: 0
show sip-ua calls dtmf-relay sip-info
The following sample output displays active SIP calls with INFO DTMF Relay mode.
Example:

Device# show sip-ua calls dtmf-relay sip-info

Total SIP call legs:2, User Agent Client:1, User Agent Server:1

SIP UAC CALL INFO

Call 1

SIP Call ID

: 9598A547-5C1311E2-8008F709-2470C996@172.27.161.122

State of the call

: STATE_ACTIVE (7)

Calling Number

: sipp

Called Number

: 3269011111

CC Call ID

:2

No.

Timestamp

Digit

Duration

=======================================================

0 01/12/2013 17:23:25.615 2

250

1 01/12/2013 17:23:25.967 5

300

2 01/12/2013 17:23:26.367 6

300

Call 2

SIP Call ID

: 1-29452@172.25.208.177

State of the call

: STATE_ACTIVE (7)

Calling Number

: sipp

Called Number

: 3269011111

CC Call ID

:1

No.

Timestamp

Digit

Duration

=======================================================

0 01/12/2013 17:23:25.615 2

250

1 01/12/2013 17:23:25.967 5

300

2 01/12/2013 17:23:26.367 6

300

Number of SIP User Agent Client(UAC) calls: 2

SIP UAS CALL INFO

Call 1

SIP Call ID

: 1-29452@172.25.208.177

State of the call

: STATE_ACTIVE (7)

Calling Number

: sipp

Called Number

: 3269011111

CC Call ID

:1

No.

Timestamp

Digit

Duration

=======================================================

0 01/12/2013 17:23:25.615 2

250

1 01/12/2013 17:23:25.967 5

300

2 01/12/2013 17:23:26.367 6

300

Call 2

SIP Call ID

: 9598A547-5C1311E2-8008F709-2470C996@172.27.161.122

State of the call

: STATE_ACTIVE (7)

Calling Number

: sipp

Called Number

: 3269011111

CC Call ID

:2

No.

Timestamp

Digit

Duration

=======================================================

0 01/12/2013 17:23:25.615 2

250

1 01/12/2013 17:23:25.967 5

300

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Step 3 Step 4

2 01/12/2013 17:23:26.367 6

300

Number of SIP User Agent Server(UAS) calls: 2

show sip-ua history dtmf-relay kpml The following sample output displays SIP call history with KMPL DTMF Relay mode. Example:

Device# show sip-ua history dtmf-relay kpml

Total SIP call legs:2, User Agent Client:1, User Agent Server:1

SIP UAC CALL INFO

Call 1

SIP Call ID

: D0498774-F01311E3-82A0DE9F-78C438FF@10.86.176.119

State of the call

: STATE_ACTIVE (7)

Calling Number

: 2017

Called Number

: 1011

CC Call ID

: 257

No.

Timestamp

Digit

Duration

=======================================================

Call 2

SIP Call ID

: 22BC36A5-F01411E3-81808A6A-5FE95113@10.86.176.142

State of the call

: STATE_ACTIVE (7)

Calling Number

: 2017

Called Number

: 1011

CC Call ID

: 256

No.

Timestamp

Digit

Duration

=======================================================

Number of SIP User Agent Client(UAC) calls: 2

SIP UAS CALL INFO

Call 1

SIP Call ID

: 22BC36A5-F01411E3-81808A6A-5FE95113@10.86.176.142

State of the call

: STATE_ACTIVE (7)

Calling Number

: 2017

Called Number

: 1011

CC Call ID

: 256

No.

Timestamp

Digit

Duration

=======================================================

Call 2

SIP Call ID

: D0498774-F01311E3-82A0DE9F-78C438FF@10.86.176.119

State of the call

: STATE_ACTIVE (7)

Calling Number

: 2017

Called Number

: 1011

CC Call ID

: 257

No.

Timestamp

Digit

Duration

=======================================================

Number of SIP User Agent Server(UAS) calls: 2

show sip-ua history dtmf-relay sip-notify The following sample output displays SIP call history with SIP Notify DTMF Relay mode. Example:

Device# show sip-ua history dtmf-relay sip-notify

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Verifying DTMF Relay

Total SIP call legs:2, User Agent Client:1, User Agent Server:1

SIP UAC CALL INFO

Call 1

SIP Call ID

: 29BB98C-F01311E3-8297DE9F-78C438FF@10.86.176.119

State of the call

: STATE_ACTIVE (7)

Calling Number

: 2017

Called Number

: 1011

CC Call ID

: 252

No.

Timestamp

Digit

Duration

=======================================================

Call 2

SIP Call ID

: 550E973B-F01311E3-817A8A6A-5FE95113@10.86.176.142

State of the call

: STATE_ACTIVE (7)

Calling Number

: 2017

Called Number

: 1011

CC Call ID

: 251

No.

Timestamp

Digit

Duration

=======================================================

Number of SIP User Agent Client(UAC) calls: 2

SIP UAS CALL INFO

Call 1

SIP Call ID

: 550E973B-F01311E3-817A8A6A-5FE95113@10.86.176.142

State of the call

: STATE_ACTIVE (7)

Calling Number

: 2017

Called Number

: 1011

CC Call ID

: 251

No.

Timestamp

Digit

Duration

=======================================================

Call 2

SIP Call ID

: 29BB98C-F01311E3-8297DE9F-78C438FF@10.86.176.119

State of the call

: STATE_ACTIVE (7)

Calling Number

: 2017

Called Number

: 1011

CC Call ID

: 252

No.

Timestamp

Digit

Duration

=======================================================

Number of SIP User Agent Server(UAS) calls: 2

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 49

Verifying DTMF Relay

CUBE Fundamentals and Basic Setup

Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 50

8 C H A P T E R
Introduction to Codecs
A codec is a device or software capable of encoding or decoding a digital data stream or signal. Audio codecs can code or decode a digital data stream of audio. Video codecs enable compression or decompression of digital video. CUBE uses codecs to compress digital voice samples to reduce bandwidth usage per call. This chapter describes the basics of encoding digital voice samples using codecs and how to configure them.
· Why CUBE Needs Codecs, on page 51 · Voice Media Transmission, on page 52 · Voice Activity Detection, on page 53 · VoIP Bandwidth Requirements, on page 54 · Supported Audio and Video Codecs, on page 56 · How to Configure Codecs, on page 57 · Configuration Examples for Codecs, on page 62
Why CUBE Needs Codecs
CUBE uses codecs to compress digital voice samples to reduce bandwidth usage per call. Refer to Table 14: Codec and Bandwidth Information, on page 54 to see the relationship between codec and bandwidth utilization. Configuring codecs on a device (configured as CUBE) allows the device to act as a demarcation point on a VoIP network and allows a dial peer to be established only if the desired codec criteria are satisfied. Additionally, preferences can be used to determine which codecs are selected over others. If codec filtering is not required, CUBE also supports transparent codec negotiations. This enables negotiations between endpoints with CUBE leaving the codec information untouched. The illustrations below show how codec negotiation is performed on CUBE. Two VoIP clouds need to be interconnected. In this scenario, both VoIP 1 and VoIP 2 networks have G.711 a-law configured as the preferred codec.
Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 51

Restrictions for Voice-Class Codec Transparent Figure 11: Codec Negotiation on CUBE

CUBE Fundamentals and Basic Setup

In the first example, the CUBE router is configured to use the G.729a codec. This can be done by using the appropriate codec command on both VoIP dial peers. When a call is set up, CUBE will accept only G.729a calls, thus influencing the codec negotiation. In the second example, the CUBE dial peers are configured with a transparent codec and this leaves the codec information contained within the call signaling untouched. Because both VoIP 1 and VoIP 2 have G.711 a-law as their first choice, the resulting call will be a G.711 a-law call.
Restrictions for Voice-Class Codec Transparent
· While using the voice-class codec transparent, only the offer is passed transparently (without filtering). Codec filtering is done on the SDP present in answer and the first codec is passed to other side.
· CUBE does not support Early-Offer to Delayed-Offer (EO-DO) call flows.
Note You can use ‘pass-thru content sdp’, if you do not want to involve CUBE in the codec negotiation.
Voice Media Transmission
When a VoIP call is established, using the signaling protocols, the digitized voice samples need to be transmitted. These voice samples are often called the voice media. Voice media protocols found in a VoIP environment are the following:
· Real-Time Transport Protocol (RTP)–RTP is a Layer 4 protocol that is encapsulated inside UDP segments. RTP carries the actual digitized voice samples in a call.
Cisco Unified Border Element Configuration Guide Through Cisco IOS XE 17.5 52

CUBE Fundamentals and Basic Setup

Voice Activity Detection

· Real-Time Control Protocol (RTcP)–RTcP is a companion protocol to RTP. Both RTP and RTcP operate at Layer 4 and are encapsulated in UDP. RTP and RTCP typically use UDP ports 16384 to 32767, though these ranges may vary according to hardware platform. However, RTP uses the even port numbers in that range, whereas RTcP uses the odd port numbers. While RTP is responsible for carrying the voice stream, RTcP carries information about the RTP stream such as latency, jitter, packets, and octets sent and received.
· Compressed RTP (cRTP)–One of the challenges with RTP is its overhead. Specifically, the combined IP, UDP, and RTP headers are approximately 40 bytes in size, whereas a common voice payload size on a VoIP network is only 20 bytes, which includes 20 ms of voice by default. In that case, the header is twice the size of the payload. cRTP is used for RTP header compression and can reduce the 40-byte header to 2 or 4 bytes in size (depending on whether UDP checksums are in use), as shown in the figure below.
Figure 12: Compressed RTP

· Secure RTP (sRTP)–To help prevent an attacker from intercepting and decoding or possibly manipulating voice packets, sRTP supports encryption of RTP packets. In addition, sRTP provides message authentication, integrity checking, and protection against replay attacks.
VPN technology like IP Security (IPSec) may be used to protect traffic between sites. Encrypting sRTP traffic at the source of transmission results in encrypting already encrypted traffic, adding significant overhead and bandwidth needs. So it is recommended that sRTP is used for voice traffic, and that this traffic is excluded from IPSec encapsulation. sRTP uses lesser bandwidth, has the same level of security, and can be used by devices at any location because the payload is originated and terminated at the voice endpoint. Because endpoints can be mobile, the security follows the phone.
Voice Activity Detection
Voice Activity Detection (VAD) is a technology that works with the human nature of voice conversations, mainly that one person listens while the other talks. VAD classifies traffic as speech, unknown, and silence. Speech and unknown payloads are transported, but silence is dropped. This accounts for approximately 30 percent savings in bandwidth over time.
VAD can significantly reduce the amount of bandwidth required by a media stream. However, VAD has a few negative attributes that need to be considered. Because no packets are sent during silence, the listener can get the impression that the talker has been disconnected. Another characteristic is that it takes a moment for VAD to recognize the speech as having started again, and as a result, the first part of the sentence can be clipped. This can be annoying to the listening party. Music on Hold (MoH) and fax can also cause VAD to become ineffective because the media stream is constant.
VAD is enabled by default in CUBE dial peers as long as the codec selected support

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