Implementation & Management of Cisco Unified Border · CUBE (Enterprise) Product Portfolio 2900...

Implementation & Management of Cisco Unified Border

Element (CUBE) Enterprise Hussain Ali, CCIE# 38068 (Voice)

Technical Marketing Engineer

BRKUCC-2934

John Vickroy

Product Manager

• SIP Trunking and CUBE Overview

• CUBE Architecture (Physical & Virtual)

• Transitioning to SIP Trunking using CUBE

• Advanced features on CUBE

• CUBE Management & Troubleshooting

• Futures & Key Takeaways

Agenda

CUBE Overview

CUBE (Enterprise) Product Portfolio

2900 Series ISR-G2 (2901, 2911, 2921, 2951)

ASR 1004/6 RP2

Active Concurrent Voice Calls Capacity

CP

S

<5

8-12

50-150

14-16K <50 500-600 900-1000

3900 Series ISR-G2 (3925, 3945)

17

3900E Series ISR-G2 (3925E, 3945E)

2000-2500

20-35

4

800 ISR

7000-10,000

50-100

12K-14K

ASR 1002-X

4500-6000

ISR 4451-X

ASR 1001-X

4000

ISR 4431

ISR-4K (4321, 4331)

ISR 4351

Note: New platforms in this

release : Introducing ISR-4K

Series (IOS-XE 3.13.1+

recommended)

Introducing CUBE on

CSR

vCUBE [Performance

dependent on vCPU and

memory]

CUBE Session Capacity Summary

Platform CUBE Sessions

NanoCUBE (8XX and SPIAD Platforms) 15 - 120

2901 – 4321 100

2911 – 2921 200 – 400

4331 500

2951 600

3925 – 3945 800 – 950

4351 1000

3925E – 3945E 2100 – 2500

4431 3000

4451 6000

ASR1001-X 12000

ASR1002-X 14000

ASR1004/1006 RP2 16000

Introduced in Oct 2013

Introduced in July 2013

Introduced in May 2014

For Your

Reference

ISR G2 CUBE Ent ASR Parity

with ISR

ASR / ISR-4K*/vCUBE (CSR)*

CUBE Vers.

2900/ 3900 FCS CUBE Vers.

IOS XE Release FCS

9.0.1 15.3.1T Oct 2012 >95% 9.0.1 3.8 15.3(1)S Oct 2012

9.0.2 15.3(2)T Mar 2013 >95% 9.0.2 3.9 15.3(2)S Mar 2013

9.5.1 15.3(3)M1 Oct 2013 >95% 9.5.1 3.10.1 15.3(3)S1 Oct 2013

10.0.0 15.4(1)T Nov 2013 >95% 10.0.0 3.11 15.4(1)S Nov 2013

10.0.1 15.4(2)T Mar 2014 >95% 10.0.1 3.12 15.4(2)S Mar 2014

10.0.2 15.4(3)M July 2014 >95% 10.0.2 3.13 15.4(3)S July 2014

10.5.0 15.5(1)T Nov 2014 >95% 10.5.0 3.14 15.5(1)S Nov 2014

11.0.0 15.5(2)T Mar 2015 >95% 11.0.0 3.15 15.5(2)S Mar 2015

11.1.0 15.5(3)M July 2015 >95% 11.1.0 3.16 15.5(3)S July 2015

11.5.0 15.6(1)T Nov 2015 >95% 11.5.0 3.17 15.6(1)S Nov 2015

12.0.0 15.6(2)T Mar 2016 >95% 12.0.0 3.18 15.6(2)S Mar 2016

12.1.0 15.6(3)M July 2016 >95% 12.1.0 3.19 15.6(3)S July 2016

* IOS-XE3.13.1 or later recommended for all ISR-4K series and XE3.15 for vCUBE

CUBE Software Release Mapping

CUBE INTEROPERABILITY

Proven Interoperability and Interworking with

Service Providers Worldwide Cisco Interoperability Portal:

www.cisco.com/go/interoperability

• Validated with Service

Providers World-Wide

• Independently Tested

with 3-Party PBXs in

tekVizion Labs

• Standards based Verified by

CUBE ISR(G2/4K), ASR and CSR Licensing

Platform Single-Use Licenses Redundancy Licenses

( 1 SKU for Active/Standby Pair)

Cisco 881, 886, 887, 888, 892F, SPIAD FL-NANOCUBE N/A

ISR G2 (2901, 2911, 2921, 2951, 3925, 3945, 3925E, 3945E)

FL-CUBEE-5

FL-CUBEE-25

FL-CUBEE-100

FL-CUBEE-5-RED

FL-CUBEE-25-RED

FL-CUBEE-100-RED

ISR-4K (4321, 4331, 4351, 4431, 4451)

FL-CUBEE-5

FL-CUBEE-25

FL-CUBEE-100

FL-CUBEE-5-RED

FL-CUBEE-25-RED

FL-CUBEE-100-RED

Cisco ASR1001-X, 1002-X, 1004 RP2, 1006 RP2

FLASR1-CUBEE-100P

FLASR1-CUBEE-4KP

FLASR1-CUBEE-16KP

FLASR1-CUBEE-100R

FLASR1-CE-100R

FLASR1-CE-500R

FLASR1-CE-1KR

FLASR1-CUBEE-4K-R

FLASR1-CE-4KR

FLASR1-CUBEE-16KR

FLASR1-CE-16KR

vCUBE (CUBE on CSR 1000v)

APPX (No TLS/SRTP) or AX (All features) CSR licensing package

Same SKUs as ASR1K series Same SKUs as ASR1K series

http://www.cisco.com/c/en/us/products/collateral/unified-communications/unified-border-element/order_guide_c07_462222.html

For Your

Reference















Agenda

CUBE Call Flow

CUBE Call Processing

Actively involved in the call treatment, signaling and media streams

SIP B2B User Agent

Signaling is terminated, interpreted and re-originated

Provides full inspection of signaling, and protection against malformed and malicious packets

Media is handled in two different modes: Media Flow-Through

Media Flow-Around

Digital Signal Processors (DSPs) are only required for transcoding (calls with dissimilar codecs)

IP

CUBE

CUBE

IP

Media Flow-Around

Signaling and media terminated by the CUBE

Media bypasses the CUBE

Media Flow-Through

Signaling and media terminated by the CUBE

Transcoding and complete IP address hiding require this model

Cisco Unified Border Element Basic Call Flow

1. Incoming VoIP setup message from originating endpoint

2. This matches inbound VoIP dial peer 1 for characteristics such as codec, VAD, DTMF, protocol, etc.

3. Match the called number to outbound VoIP dial peer 2

4. Outgoing VoIP setup message

Incoming VoIP Call Outgoing VoIP Call

dial-peer voice 1 voip destination-pattern 1000 session protocol sipv2 session target ipv4:1.1.1.10 codec g711ulaw

dial-peer voice 2 voip destination-pattern 2000 session protocol sipv2 session target ipv4:2.2.2.10 codec g711ulaw

Originating Endpoint –

1000

Terminating Endpoint –

2000

CUBE

voice service voip

mode border-element

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

CUBE

1000 1.1.1.1

2000 2.2.2.2

Incoming VoIP Call Leg Matches an Incoming Dial-peer

Outgoing VoIP Call Leg Matches an Outbound Dial-peer

INVITE /w SDP INVITE /w SDP

100 TRYING 100 TRYING

10.10.10.10 20.20.20.20

c= 1.1.1.1 m=audio abc RTP/AVP 0

c= 20.20.20.20 m=audio xxx RTP/AVP 0

180 RINGING 180 RINGING

200 OK 200 OK c= 2.2.2.2

m=audio uvw RTP/AVP 0 c= 10.10.10.10 m=audio xyz RTP/AVP 0

RTP (Audio)

ACK ACK

1.1.1.1 10.10.10.10 20.20.20.20 2.2.2.2

Understanding the Call flow

BYE BYE

200 OK 200 OK

Basic Show commands for Active Calls

CUBE# show call active voice brief 121A : 17 13:02:24.215 IST Mon Jun 27 2011.1 +2040 pid:2 Answer 2000 active dur 00:00:14 tx:0/0 rx:0/0 IP 2.2.2.2:6001 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a 121A : 18 13:02:24.225 IST Mon Jun 27 2011.1 +2020 pid:1 Originate 1000 active dur 00:00:14 tx:0/0 rx:0/0 IP 1.1.1.1:6000 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 TextRelay: off media inactive detected:n media contrl rcvd:n/a timestamp:n/a long duration call detected:n long duration call duration:n/a timestamp:n/a Telephony call-legs: 0 SIP call-legs: 2 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 0 Multicast call-legs: 0 Total call-legs: 2 CUBE# show voip rtp connections VoIP RTP active connections : No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP 1 17 18 17474 6000 10.10.10.10 1.1.1.1 2 18 17 17476 6001 20.20.20.20 2.2.2.2 Found 2 active RTP connections

CUBE Architecture ISR G2 vs ASR1K vs ISR 4K vs vCUBE (CUBE on CSR1000v)

ASR/ISR-4K & ISR-G2 Architecture Comparison

ISR: Pkt fwd’ing and signaling are handled by the same CPU

ASR: Pkt fwd’ing and signaling are handled by different CPUs

‒ ESP must be programmed or instructed by the control plane to do specific media functions

‒ Performed by Forwarding Plane Interface (FPI)

I/O ESP I/O

Kernel

IOS IOS

Msg I/f

Control Plane

Data (Forwarding) Plane

ASR/ISR-4K Architecture

RP IOS

CPU

I/O I/O

Control

Plane

Data Plane

ISR G2 Architecture

Sig

na

ling

Sig

na

ling

Media

Introducing vCUBE (CUBE on CSR 1000v) Architecture

• CSR (Cloud Services Router) 1000v runs on a Hypervisor – IOS XE without the router

Console Mgmt ENET Ethernet NICs Flash / Disk Memory Virtual CPU

RP (control plane)

Chassis Mgr.

Forwarding Mgr. IOS-XE

Kernel (incl. utilities)

ESP (data plane)

Chassis Mgr.

Forwarding Mgr.

QFP Client / Driver

FFP code

Hypervisor

Hardware

vSwitch NIC

GE GE … X86 Multi-Core CPU Memory Banks

ESXi Container

CUBE signaling CUBE media processing

CSR 1000v (virtual IOS-XE)

Introducing vCUBE (CUBE on CSR 1000v) • CSR1000v is a virtual machine, running on x86 server (no specialized hardware) with

physical resources are managed by hypervisor and shared among VMs

• Can be installed either using an OVA file or deployed with an ISO image

• Requires APPX (No TLS/SRTP) or AX (All vCUBE features) CSR licensing package to access voice CLI and increase throughput from 100 kbps default. CUBE Licensing follows ASR1K SKUs and still trust based

• No DSP based features (transcoding/inband-RFC2833 DTMF/ASP/NR) available

• vMotion for vCUBE not supported today

• vCUBE Tested Reference Configurations [UCS base-M2-C460, C220-M3S, ESXi 5.1.0 & 5.5.0]

ASR, CSR & ISR-G2/4K Feature Comparison General SBC Features ASR1K ISR-G2 4300/4400 (XE3.13.1+) vCUBE (XE3.15+)

High Availability Implementation Redundancy-Group

Infrastructure HSRP Based

Redundancy-Group

Infrastructure

Redundancy-Group

Infrastructure

TDM Trunk Failover/Co-

existence Not Available Exists Exists Not Available

Media Forking XE3.8 15.2.1T XE3.10 Exists

Software MTP registered to

CUCM (Including HA Support) XE3.6 Exists Exists Exists

DSP Card SPA-DSP PVDM2/PVDM3 PVDM4 Not Available

Transcoder registered to CUCM Not Available Exists via SCCP Exists via SCCP (XE3.11) Not Available

Transcoder Implementation Local Transcoder Interface

(LTI)

SCCP or

LTI (starting IOS 15.2.3T) SCCP and LTI Not Available

Embedded Packet Capture Exists Exists Exists Exists

Web-based UC API XE3.8 15.2.2T Exists Exists

Noise Reduction & ASP Exists 15.2.3T Exists Not Available

Call Progress Analysis XE3.9 15.3.2T Exists Not Available

CME/SRST and CUBE co-

existence Not Available Exists XE3.11 Roadmap

SRTP-RTP Call flows Exists (NO DSPs needed) Exists (DSPs required) Exists (NO DSPs needed) Exists (No DSPs needed)

VXML GW Not Available Exists Not Available Not Available

vCUBE Installation using OVA

vCUBE – CSR1000v Installation with OVA

• Download CSR1000v OVA from cisco.com

vCUBE – Download XE3.15 image

vCUBE – Deploy OVA

vCUBE – Installation Cont’d

vCUBE – Choose Form factor

vCUBE Installation Cont’d

vCUBE – Assign LAN, WAN, and VM Network

vCUBE Installation Cont’d

vCUBE Installation – Edit Settings to add Serial Port

Serial Port – Connect via Network

Serial Port – Define URL

Serial Port – Verify Settings

vCUBE Installation – Power On VM

Install process takes some time

vCUBE – Initial Configuration

• Assign IP to VM Network Interface, Gig3 below, and enable console access with “platform console serial” CLI, and set enable password

vCUBE – Initial Configuration – Telnet into Router

Initial Configuration – Copy License File to Flash:

Initial Configuration – Install License File

Initial Configuration – Verify New Throughput Level and boot CSR to the correct package

vCUBE Initial Setup – Voice CLI is now accessible







Agenda

High-density Dedicated

Gateways

Transitioning to Centralized SIP Trunking... Re-purpose your existing Cisco voice gateway’s as Session Border Controllers

SIP/H323/MGCP

Media

TDM PBX

SRST CME

A Enterprise Campus

Enterprise

Branch Offices

MPLS

BEFORE Media

SIP Trunks

SRST

Enterprise

Campus

IP PSTN A

TDM PBX

CME

MPLS

CUBE with High

Availability

Active

Standby

CUBE

CUBE

PSTN is now

used only for

emergency

calls over FXO

lines

AFTER

46

Enterprise

Branch Offices

• Step 1 – Configure IP PBX to route all calls (HQ and branch offices) to the edge SBC

• Step 2 – Get SIP Trunk details from the provider

• Step 3 – Enable CUBE application on Cisco routers

• Step 4 – Configure call routing on CUBE (Incoming & Outgoing dial-peers)

• Step 5 – Normalize SIP messages to meet SIP Trunk provider’s requirements

• Step 6 – Execute the test plan

Steps to transitioning...

Media

SRST

Enterprise

Campus

IP PSTN A

TDM PBX

CME

MPLS

Enterprise Branch

Offices

CUBE with High

Availability

Active

Standby

CUBE

CUBE

PSTN is now

used only for

emergency

calls over FXO

lines

SIP Trunk

SIP Trunk Pointing to CUBE

Step 1: Configure CUCM to route calls to the edge SBC

Standby

IP PSTN A

TDM PBX

SRST

CME

MPLS

Enterprise Branch Offices

Enterprise

Campus CUBE with High

Availability

Active

CUBE

CUBE

PSTN is now used only for emergency calls over FXO lines

• Configure CUCM to route all PSTN calls (central and branch) to CUBE via a SIP trunk

• Make sure all different patterns of calls – local, long distance, international, emergency, informational etc.. are pointing to CUBE

Also see BRKUCC-2006

Step 2: Get details from SIP Trunk provider

Item SIP Trunk service provider requirement Sample Response

1 SIP Trunk IP Address (Destination IP Address for INVITES) 20.1.1.2 or DNS

2 SIP Trunk Port number (Destination port number for INVITES) 5060

3 SIP Trunk Transport Layer (UDP or TCP) UDP

4 Codecs supported G711, G729

5 Fax protocol support T.38

6 DTMF signaling mechanism RFC2833

7 Does the provider require SDP information in initial INVITE (Early offer

required)

Yes

8 SBC’s external IP address that is required for the SP to accept/authenticate

calls (Source IP Address for INVITES)

20.1.1.1

9 Does SP require SIP Trunk registration for each DID? If yes, what is the

username & password

No

10 Does SP require Digest Authentication? If yes, what is the username &

password

No

Step 3: Enable CUBE Application on Cisco routers

voice service voip

mode border-element license capacity 200

allow-connections sip to sip

2. Configure any other global settings to meet SP’s requirements

voice service voip

sip

early-offer forced

header-passing

error-passthru

3. Create a trusted list of IP addresses to prevent toll-fraud

voice service voip ip address trusted list Applications initiating signaling towards CUBE, e.g. CUCM, ipv4 20.20.20.20 CVP, Service Provider’s SBC ipv4 10.1.1.50 sip silent discard-untrusted Default configuration starting XE 3.10.1 /15.3(3)M1 to mitigate TDoS Attack

1. Enable CUBE Application

Step 4: Configure Call routing on CUBE

IP PSTN A

TDM PBX

SRST

CME

MPLS

Enterprise Branch Offices

Enterprise Campus

CUBE with High

Availability

Active

Standby

CUBE

CUBE

PSTN is now used only for emergency calls over FXO lines

WAN Dial-Peers LAN Dial-Peers

• Dial-Peer – “static routing” table mapping phone numbers to interfaces or IP addresses

• LAN Dial-Peers – Dial-peers that are facing towards the IP PBX for sending and

receiving

calls to & from the PBX

• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for sending

& receiving calls to & from the provider

WAN Dial-Peer Configuration

dial-peer voice 100 voip

description *** Inbound WAN side dial-peer ***

incoming called-number 408527….$

session protocol sipv2

voice-class sip bind control source gig0/1

voice-class sip bind media source gig0/1

codec g711ulaw

dtmf-relay rtp-nte

Outbound Dial-Peer for call legs from CUBE to SP

Specific to your DID range

assigned by the SP

Dial-peer for making long distance

calls to SP, based on NANP (North

American Numbering Plan)

Note: Separate outgoing DP to be created for Local, International,

Emergency, Informational calls etc.

Inbound Dial-Peer for call legs from SP to CUBE

Apply bind to all dial-peers when

CUBE has multiple interfaces.

Gig0/1 faces SP.

Translation rule/profile to strip the

access code (9) before delivering

the call to the SP


description *** Outbound WAN side dial-peer ***

translation-profile outgoing Digitstrip

destination-pattern 91[2-9]..[2-9]......$




session target ipv4:<SIP_Trunk_IP_Address>

codec g711ulaw

dtmf-relay rtp-nte

LAN Dial-Peer Configuration


description *** Inbound LAN side dial-peer ***

incoming called-number 9T




codec g711ulaw

dtmf-relay rtp-nte

Outbound Dial-Peer for call legs from CUBE to CUCM


description *** Outbound LAN side dial-peer ***

destination-pattern 408527….$




session target ipv4:<CUCM_IP_Address>

codec g711ulaw

dtmf-relay rtp-nte

CUCM sending 9 (access code) + All

digits dialed

SP will be sending 10 digits (NANP)

based on your DID that is being

delivered to CUCM

Inbound Dial-Peer for call legs from CUCM to CUBE

Apply bind to all dial-peers when

CUBE has multiple interfaces. Gig0/0

faces CUCM.

Default codec is G729 if none is

specified

Note: If more than 1 CUCM cluster exists, you will have to create multiple such LAN dial-peers with “preference CLI” for

CUCM redundancy/load balancing as the traditional way to accommodate multiple trunks

Step 5: SIP Normalization

SIP incompatibilities arise due to:

• A device rejecting an unknown header (value or parameter) instead of ignoring it

• A device expecting an optional header value/parameter or can be implemented in multiple ways

• A device sending a value/parameter that must be changed or suppressed (“normalized”) before it leaves/enters the enterprise to comply with policies

• Variations in the SIP standards of how to achieve certain functions

• With CUBE 10.0.1 SIP Profiles can be applied to inbound SIP messages as well

More information at www.cisco.com/go/cube > Configure > Configuration Examples and TechNotes

Incoming Outgoing

INVITE

sip:[email protected]:5060

user=phone SIP/2.0

INVITE


SIP/2.0

voice class sip-profiles 100

request INVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"

request REINVITE sip-header SIP-Req-URI modify "; SIP/2.0" ";user=phone SIP/2.0"

Add user=phone for INVITEs

Modify a “sip:” URI to a “tel:” URI in INVITEs

Incoming

Outgoing

INVITE

tel:2222000020

SIP/2.0

INVITE


SIP/2.0


request INVITE sip-header SIP-Req-URI modify "sip:(.*)@[^ ]+" "tel:\1"

request INVITE sip-header From modify "<sip:(.*)@.*>" "<tel:\1>"

request INVITE sip-header To modify "<sip:(.*)@.*>" "<tel:\1>"

CUBE

CUBE

SIP profiles is a mechanism to normalize or customize SIP at the network border to provide interop between incompatible devices

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

http://www.cisco.com/web/tsweb/tools/sip-profile/index.html




Normalize Inbound SIP Message (Example 1)


request INVITE sip-header Diversion modify “sip:” sip:1234@


description Incoming/outgoing SP

voice-class sip profiles 400 inbound

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

………

User-Agent: SP-SBC

………

Diversion: <sip:9.44.44.4>;privacy=off;

reason=unconditional;screen=yes

……...

m=audio 6001 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

……...

Configure Inbound

SIP Profile to add a

dummy user part

Apply to Dial-peer

or Globally

Sent:

INVITE sip:[email protected]:5060 SIP/2.0

……….

User-Agent: SP-SBC

……….

Diversion: <sip:[email protected]>;

privacy=off;reason=unconditional;screen=yes

……….

m=audio 32278 RTP/AVP 18 8 101

a=rtpmap:0 PCMU/8000

………..

voice service voip

sip

sip profiles 400 inbound

CUBE

Requirement SIP Diversion header must include a user portion

SIP INVITE received by CUBE SIP INVITE CUBE expects

Enable Inbound SIP

Profile feature voice service voip

sip

sip-profiles inbound

For Your

Reference

SIP Profile Rule Tagging

SIP Profile – Existing Feature Overview

1. Insertion

• New rules are always inserted at the end, there was no way to insert a rule at the beginning or in between existing rules

• Only way to achieve this is by removing the complete profile and configuring it again in the desired order

2. Deletion

• While deleting a rule User has to give complete no form of that rule

• If there are duplicate rules, always 1st one is deleted

3. Modification

• There is no direct way to modify an existing rule. User has to delete and reconfigure the profile

4. Duplication

• If the same profile/rules applied more than once, then the rules are be duplicated

SIP Profile Tagging Enhancement

New rule tagging mechanism has been introduced

1. Insertion

• New rules can be inserted at any position i.e at the beginning, at the end or in between existing rules by specifying rule tag number

2. Deletion

• Rules can be deleted by giving no form of the rule with just the tag number

3. Modification

• Any of the existing rules can be modified by specifying the rule tag number

4. Duplication

• When a rule with an existing tag number is applied again, the rule will be over-written, without creating any duplicate rules

A mechanism to automatically upgrade the legacy SIP Profile configurations to the new rule format has been provided. The following exec CLI is being provided to upgrade existing implementation

voice sip sip-profiles upgrade

A mechanism to automatically downgrade the SIP Profile configurations with the rule tags to non-rule format has been provided. The following exe CLI has been provided for this purpose

voice sip sip-profiles downgrade

Note: When SIP Profiles are configured in “rule <tag>” format and the IOS version is migrated to a version which does not have this capability, then all the SIP Profile configurations will be lost. Hence, it is advisable to execute voice sip sip-profiles downgrade before IOS version migration.

SIP Profile Tagging Enhancement – Cont’d

• For tagging the rules, an additional option of “rule <tag>” has been provided

SIP Profile Tagging – Configuration

CUBE(config)#voice class sip-profiles 1

CUBE(config-class)#?

VOICECLASS configuration commands:

exit Exit from voice class configuration mode

help Description of the interactive help system

no Negate a command or set its defaults

request sip request

response sip response

rule Specify the rule

CUBE(config-class)#rule ?

<1-1073741823> Specify the rule tag before The rule to be inserted before

CUBE(config-class)#rule 1 ?

request sip request


The new

keyword “rule”

“tag” to be

provided with

rule keyword

• For inserting a rule between two rules, “before” option has been provided

SIP Profile Tagging – Configuration Cont’d


CUBE(config-class)#rule before ?

<1-1073741823> Specify the rule tag

CUBE(config-class)#rule before 3 ?

request sip request


• If rule <tag> option is used to configure a SIP Profile rule, then this rule can be deleted by specifying just the tag number instead of specifying the entire rule configuration.


CUBE(config-class)#no rule before <tag>

For inserting a rule

between two rules, the new before keyword

is being introduced

Configuration Example

• For tagging the rules:


rule 1 request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”

rule 2 request INVITE sip-header Supported Add “Supported: ”

• For inserting a rule between two rules using “before” option:

rule before 2 request INVITE sip-header To Modify “(.*)” “\1;temp=abc”



rule 2 request INVITE sip-header To Modify “(.*)” “\1;temp=abc”


before

option

The new rule has been

inserted between #1

and #3

• Auto-Upgrade : Exec command - “voice sip sip-profiles upgrade”

• Suppose we have the following rules configured:

request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”

request INVITE sip-header Supported Add “Supported: ”

request REGISTER sip-header Contact Modify “(.*)” “\1;temp=abc”

• After auto upgrade, the rules will be automatically upgraded as follows:



rule 3 request REGISTER sip-header Contact Modify “(.*)” “\1;temp=abc”

Configuration Example continued….

• Auto-Downgrade : Exec command - “voice sip sip-profiles downgrade”

• Suppose we have the following rules configured:



rule 3 request REGISTER sip-header Contact Modify “(.*)” “\1;temp=abc”

• After auto downgrade, the rules will be automatically downgraded as follows:

request INVITE sip-header Contact Modify “(.*)” “\1;temp=xyz”

request INVITE sip-header Supported Add “Supported: ”

request REGISTER sip-header Contact Modify “(.*)” “\1;temp=abc”

Configuration Example continued….

SIP Profile Support for Non-Standard Headers

SIP Profile support for Non-Standard Headers Introducing support for adding/copying/removing/modifying non-

standard SIP headers using SIP profiles

A new 'WORD' option has been added to the SIP Profiles CLI chain to allow the user to configure any non-standard SIP Header


CUBE(config-class)#request INVITE sip-header ?

Accept-Contact SIP header Accept-Contact

…….

Via SIP header Via

WORD Any other SIP header name WWW-Authenticate SIP header WWW-Authenticate

CUBE(config-class)#request INVITE sip-header

ADD addition of the header

COPY Copy a header

MODIFY Modification of a header

REMOVE Removal of a header

The new “WORD”

option for specifying

unsupported headers

CUBE-Best-Session-in-SanDiego ?

Step 6: Execute the Test Plan

• Inbound and outbound Local, Long distance, International calls for G711 & G729 codecs (if supported by provider)

• Outbound calls to information and emergency services

• Caller ID and Calling Name Presentation

• Supplementary services like Call Hold, Resume, Call Forward & Transfer

• DTMF Tests

• Fax calls – T.38 and fallback to pass-through (if option available)







Agenda

CUBE Dial-Peers Call Routing

Understanding Dial-Peer matching Techniques: LAN & WAN Dial-Peers

• LAN Dial-Peers – Dial-peers that are facing towards the IP PBX for sending and receiving calls to & from the PBX

• WAN Dial-Peers – Dial-peers that are facing towards the SIP Trunk provider for sending & receiving calls to & from the provider

CUCM SIP Trunk SP SIP Trunk

CUBE

A

Outbound Calls Outbound WAN Dial-Peer Inbound LAN Dial-Peer

IP PSTN

Inbound WAN Dial-Peer Outbound LAN Dial-Peer Inbound Calls

Understanding Inbound Dial-Peer Matching Techniques


CUBE

A

Inbound LAN Dial-Peer

IP

PSTN

Inbound WAN Dial-Peer Inbound Calls

Outbound Calls

Match based on Called

Number

Match based on Calling

number

1 Match Based on URI of

an incoming INVITE

message

Default Dial-Peer = 0

Exact Pattern

match

Host Name/IP

Address

User portion of

URI

Phone-number of

tel-uri Received:

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-

tag="cid:[email protected]";;branch=z9hG4bK-23955-1-0

From: "555" <sip:[email protected]:5060>;tag=1

To: ABC <sip:[email protected]:5060>

Call-ID: [email protected]

CSeq: 1 INVITE

Contact: sip:[email protected]:5060

Supported: timer

Max-Forwards: 70

Subject: BRKUCC-2934 Session

Content-Type: application/sdp

Content-Length: 226

........

2

3

4

Priority



incoming called-number 654321


answer-address 555


destination-pattern 555

voice class uri 1001 sip

host ipv4:10.1.1.1


host ipv4:10.2.1.1


incoming uri via 1001


incoming uri request 2001


incoming uri to 2001


incoming uri from 1001

Received:







CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

Priority

A

B

C

D


CUBE

A


IP

PSTN


Outbound Calls





answer-address 555




host ipv4:10.1.1.1


host ipv4:10.2.1.1









Received:







CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

A

B

C

D

Priority CUCM SIP Trunk SP SIP Trunk

CUBE

A


IP

PSTN


Outbound Calls





answer-address 555




host ipv4:10.1.1.1


host ipv4:10.2.1.1









Received:







CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

A

B

C

D


CUBE

A


IP

PSTN


Outbound Calls

Understanding Outbound Dial-Peer Matching Techniques


Number & carrier-id

target

Match Based on URI of

incoming INVITE

message & carrier-id

target

Received:







CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

Match based on URI of

an incoming INVITE

message


number


CUBE

A

Outbound LAN Dial-Peer

IP

PSTN

Outbound WAN Dial-Peer

Inbound Calls

Outbound Calls

Exact Pattern

match

Host Name/IP

Address

User portion of

URI

Phone-number of

tel-uri

1

2

3

4

Exact Pattern

match

Host Name/IP

Address

User portion of

URI

Phone-number of

tel-uri

Priority

CSCua14749 – Carrier-id CLI not working on XE based

platforms




carrier-id target orange


host ipv4:10.2.1.1


destination uri 2001




host ipv4:10.2.1.1




80

Received:



tag="cid:[email protected]";branch=z9hG4bK-23955-1-0




CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

Priority


CUBE

A


IP

PSTN

Inbound Calls

Outbound Calls






host ipv4:10.2.1.1






host ipv4:10.2.1.1




81

Received:







CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

Priority


CUBE

A


IP

PSTN

Inbound Calls

Outbound Calls






host ipv4:10.2.1.1






host ipv4:10.2.1.1




82

Received:







CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

Priority


CUBE

A


IP

PSTN

Inbound Calls

Outbound Calls






host ipv4:10.2.1.1






host ipv4:10.2.1.1




Received:







CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

Priority


CUBE

A


IP

PSTN

Inbound Calls

Outbound Calls

CUBE Advanced Call Routing



Number & carrier-id target

Match Based on URI of

incoming INVITE message

& carrier-id target

Received:

INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060;x-route-





CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

Match based on URI of an

incoming INVITE message


number

SIP Trunk SP SIP Trunk

CUBE

A


IP PSTN


Inbound Calls

Outbound Calls

Exact Pattern

match

Host Name/IP

Address

User portion of URI

Phone-number of

tel-uri

1

2

3

4

Exact Pattern

match

Host Name/IP

Address

User portion of URI

Phone-number of

tel-uri

Priority

Additional Headers for Outbound Dial-Peer Matching

Match based on DIVERSION Header of incoming

INVITE

Match Based on URI of incoming INVITE message with

or without carrier-id target

Received:







CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........ Match based on CALLING Number

Match based on CALLED Number with or without

carrier-id target


CUBE

A


IP PSTN


Inbound Calls

Outbound Calls

Match Based on FROM Header of incoming INVITE

Match Based on TO Header of incoming INVITE

Match Based on VIA Header of incoming INVITE

Match based on REFERRED-BY Header of incoming

INVITE

Introducing Outbound Dial-peer Provision Policy • Flexibility to choose how outbound dial-peers are selected

• Dynamically set the priority based on Inbound dial-peers

• Additional Inbound Leg Headers for Outbound Dial-peer Matching

VIA FROM TO DIVERSION REFERRED-BY Calling Number

• User-defined outbound dial-peer provision policy on a per incoming call bases

1. A provision policy contains two rules to save the match attributes and its precedence

2. Up to two match attributes can be defined from each rule of a provision policy

3. A provision policy setup will be used to match outbound dial-peers once it is associated to an incoming VoIP call.

• Outbound dial-peer match attributes

destination uri-via destination uri-diversion destination e164-pattern-map

destination uri-to destination uri-referred-by destination uri

destination uri-from destination calling destination-pattern

Dial-peer Provision Policy Configuration

1. Define Voice Class Dial-peer Provision Policy

CUBE(config)#voice class dial-peer provision-policy <tag>

CUBE(config-class)# description “Match outbound dial-peer based on this Criteria”

CUBE(config-class)#preference ?

<1-2> Preference order

CUBE(config-class)#preference 1 first-attribute second-attribute

called Match called number calling Match calling number

carrier-id Match carrier id diversion Match diversion uri

from Match from uri to Match to uri

uri Match destination uri via Match via uri

referred-by Match referred-by uri

voice class dial-peer provision-policy <tag>

description ‘Match outbound dial-peer based on criteria defined here’

preference 1 first-attribute second-attribute

preference 2 first-attribute second-attribute

Dial-peer Provision Policy Configuration – Cont’d

2. Associate Voice Class Provision Policy to an Incoming Dial-peer


description Inbound Dial-peer

destination provision-policy <tag>

3. Define Outbound Dial-peer with match patterns based on attributes in a policy

CUBE(config)#dial-peer voice 2 voip

CUBE(config-dial-peer)#description Outbound Dial-peer

CUBE(config-dial-peer)#destination ?

calling Match destination calling number

e164-pattern-map Configure voice class to match destination e164-pattern-map

uri Configure voice class to match destination URI

uri-diversion voice class uri to match sip diversion header

uri-from voice class uri to match sip from header

uri-referred-by voice class uri to match sip referred-by header

uri-to voice class uri to match sip to header

uri-via voice class uri to match sip via header

Dial-peer Provision Policy Configuration – Cont’d

Configuring a match command for an outbound dial-peer according to the provision policy rule attribute configured

Provision Policy Rule Attribute Outbound Dial-peer Match command

Called destination-pattern pattern

destination e164-pattern-map pattern-map-class-id

Calling destination calling e164-pattern-map pattern-map-class-id

carrier-id carrier-id target

Uri destination uri uri-class-tag

Via destination uri-via uri-class-tag

To destination uri-to uri-class-tag

from destination uri-from uri-class-tag

diversion destination uri-diversion uri-class-tag

referred-by destination uri-referred-by uri-class-tag

For Your

Reference

Destination Server Group • Supports multiple destinations (session targets) be defined in a group and applied to

a single outbound dial-peer

• Once an outbound dial-peer is selected to route an outgoing call, multiple destinations within a server group will be sorted in either round robin or preference [default] order

• This reduces the need to configure multiple dial-peers with the same capabilities but different destinations. E.g. Multiple subscribers in a cluster

voice class server-group 1

hunt-scheme {preference | round-robin}

ipv4 1.1.1.1 preference 5

ipv4 2.2.2.2

ipv4 3.3.3.3 port 3333 preference 3

ipv6 2010:AB8:0:2::1 port 2323 preference 3

ipv6 2010:AB8:0:2::2 port 2222 * DNS target not supported in server group


description Outbound DP



codec g711ulaw

dtmf-relay rtp-nte

session server-group 1

Multiple Incoming Patterns Under Same Incoming/Outgoing Dial-peer


CUBE

IP PSTN A

(408)100-1010

(510)100-1010

(919)200-2010

(919)200-2000

(510)100-1000

(408)100-1000

voice class e164-pattern-map 300

e164 919200200.

e164 510100100.

e164 408100100.


description Inbound DP via Calling

incoming calling e164-pattern-map 300

codec g729r8

voice class e164-pattern-map 400

url flash:e164-pattern-map.cfg


description Inbound DP via Called

destination e164-pattern-map 400

codec g711ulaw

! This is an example of the contents

of E164 patterns text file stored in flash:e164-pattern-map.cfg

9192002010 5101001010 4081001010

Site A

Site B

Site C

Site A

Site B

Site C

G729 Sites

G711 Sites

Provides the ability to combine multiple

incoming called OR calling numbers on

a single inbound voip dial-peer, reducing

the total number of inbound voip dial-

peers required with the same routing

capability

Up to 5000 entries in a text file

Destination Dial-peer Group

• Allows grouping of outbound dial-peers based on an incoming dial-peer, reducing existing outbound dial-peer provisioning requirements

• Eliminates the need to configure extra outbound dial-peers that are sometimes needed as workarounds to achieve desired call routing outcome

• Multiple outbound dial-peers are saved under a new “voice class dpg <tag>”. The new “destination dpg <tag>” command line of an inbound voip dial-peer can be used to reference the new dpg (dial-peer group)

• Once an incoming voip call is handled by an inbound voip dial-peer with an active dpg, dial-peers of a dpg will then be used as outbound dial-peers for an incoming call

• The order of outgoing call setups will be the sorted list of dial-peers from a dpg

Destination Dial-peer Group Configuration

voice class dpg 10000

description Voice Class DPG for DP Source SJ

dial-peer 1001 preference 1

dial-peer 1002 preference 2

dial-peer 1003

!


description DP Source SJ w/voice class dpg


destination dpg 10000


description DPG 10000



session target ipv4:10.1.1.1

!






!






1. Incoming Dial-peer

is first matched 2. Now the DPG associated

with the INBOUND DP is

selected

Media Manipulation

Audio Transcoding and Transrating

• Transcoding (12.4.20T)

• One voice codec to any other codec E.g. iLBC-G.711 or iLBC-G.729

• Support for H.323 and SIP

• CUCM 7.1.5 or later supports universal Transcoding

• Transrating (15.0.1M)

• Different packetizations of the same codec

• E.g. G.729 20ms to G.729 30ms

• Support for SIP-SIP calls

• No sRTP support with transrating

G.729 30 ms

CUBE

• Transcoding: G.711, G.723.1, G.726, G.728,

G.729/a, iLBC, G.722

• Transrating: G.729 20ms ↔ 30ms (AT&T)

Supported Codecs Packetization

(ms)

G.711 a-law 64 Kbps 10, 20, 30

G.711 µlaw 64 Kbps 10, 20, 30

G.723 5.3/6.3 Kbps 30, 60

G.729, G.729A, G.729B, G.729AB 8 Kbps

10, 20, 30, 40, 50, 60

G.722—64 Kbps 10, 20, 30

SP VoIP

Enterprise

VoIP

dial-peer voice 2 voip codec g729r8 bytes 30 fixed-bytes

iLBC, iSAC,

Speex IP Phones:

G.711, G.729 20 ms,

G.722

!Call volume (gain/loss) adjustment dial-peer voice 2 voip audio incoming level-adjustment x audio outgoing level-adjustment y

Configuration for SCCP based Transcoding (ISR-G2/4K)

voice-card 1

dspfarm

dsp services dspfarm

2. telephony-service configuration

telephony-service

sdspfarm units 1

sdspfarm transcode sessions 128

sdspfarm tag 1 CUBE-XCODE

max-ephones 10

max-dn 10

ip source-address

<CUBE_internal_IP> port 2000

3. sccp configuration

sccp local GigabitEthernet0/0

sccp ccm <CUBE_internal_IP> identifier 1 version 4.0

sccp

sccp ccm group 1

associate ccm 1 priority 1

associate profile 1 register CUBE-XCODE

4. dspfarm profile configuration

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729r8

maximum sessions 10

associate application SCCP

1. Enabling dspfarm services

under voice-card

For Your

Reference

Configuration for LTI based Transcoding (ISR-G2/4K & ASR)

voice-card 0/1

dspfarm

dsp services dspfarm

2. dspfarm profile configuration

dspfarm profile 1 transcode

codec g711ulaw

codec g711alaw

codec g729abr8

codec g729ar8

codec ilbc

maximum sessions 100

associate application CUBE

1. Enabling dspfarm services

under voice-card

Feature Notes:

• This uses Local Transcoding Interface to

communicate between CUBE and DSPs

• Also available on ISR-G2 starting IOS 15.2.3T

• Can only be used if CUBE invokes the DSP

for media services

• CUCM cannot invoke DSPs using this LTI

interface

Call Recording

CUBE Controlled Recording Option – Media Forking

• Call agent independent • Configured on a per

Dial-peer level to fork RTP

Cisco MediaSense

(authentication disabled w/o UCM)

Cisco Search/Play demo app

-or-

Partner Application

media class 1

recorder parameter

media-recording 20


description dial-peer that needs to be forked


media-class 1


description dial-peer pointing to MediaSense


session target ipv4:<Mediasense_IP>

• CUBE sets up a stateful SIP session

with MediaSense server

• After SIP dialog established, CUBE

forks the RTP and sends it for

MediaSense to record

• With XE 3.10.1, Video calls

supported and CUBE HA for audio

calls

SIP SIP

SIP

A

SP SIP

CUBE

RTP

RTP RTP

MediaSense

Needs to

match

Dial-peer based

Audio only Media Forking for an Audio/Video Call

media profile recorder 100

media-type audio media-recording 20


description dial-peer that needs to be forked


media-class 1


description dial-peer pointing to MediaSense


session target ipv4:<Mediasense_IP>

• MediaSense 10+ or any recording server can decline the video stream and choose to have only the audio

stream recorded by setting the video port as 0 in the SDP answer

• CUBE can be configured to offer only audio streams to be recorded even if the call that is being recorded

is an audio/video call

SIP SIP

SIP

A

SP SIP

CUBE

RTP

RTP RTP

MediaSense

CUBE Controlled Recording

media-class 1

recorder profile 100

• Support for forwarding any 3rd

party IP PBX GUID to the

recording server

CUCM (10.X or later) Controlled Recording

1. Enable HTTP on IOS ip http server

http client persistent

2. Enable the API on IOS uc wsapi

source-address [IP_Address_of_CUBE]

3. Enable XMF service within the API provider xmf

remote-url 1 http://CUCM:8090/ucm_xmf

no shutdown

Gateway/CUBE Recording

Enabled

1. 2.

3.

4.

5.

[1] – [3]: An external call is answered by user with IP phone

[4] – [5]: CUCM sends forking request over HTTP to CUBE, which

sends two media streams towards the Recording Server

UC Services API – Network Based Recording

• Selective Recording • Mobile/SNR/MVA Calls • Recording Call Preservation

• With XE3.13/IOS15.4(3)M, CUBE supports SRTP-SRTP, SRTP-RTP, RTP-SRTP recording. Feature on CUCM roadmap • CUBE HA not supported with CUCM controlled Recording

http://cucm:8090/ucm_xmf

High Availability

CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy

• All signaling is sourced from/to the Virtual IP Address

• Lower address for both the interfaces (Gig0/0 and Gig0/1) should be on the same platform, which is used as a tie breaker for the HSRP Active state

• HSRP Group number should be unique to a pair/interface combination on the same L2

• Both interfaces of the same group have to be configured with the same priority

• Multiple HSRP interfaces require preemption with interface tracking to be configured

• No media-flow around, SDP-Passthru, or UC Services API support for CUBE HA

CUBE HA Design Considerations on ISR-G2 for Box-to-Box Redundancy – Cont’d

• Both platforms must be connected via the same physical Switch across all interfaces for CUBE HA to

work. Cannot have WAN terminated on CUBEs directly or Data HSRP on either side

• TDM or VXML GW cannot be collocated with CUBE HA

• Both the CUBEs must be running on the same type of platform and IOS version and identical configuration. Loopback interfaces cannot be used for bind as they are always up.

• LTI based transcoding called flows including SRTP/RTP interworking preserved starting 15.5(2)T. Requires same PVDM3 chip capacity on both active and standby in the same slot/subslot

• Upon failover, the ACTIVE CUBE goes through a reload

redundancy inter-device scheme standby SB voice service voip mode border-element allow-connections sip to sip redundancy ipc zone default association 1 no shutdown protocol sctp local-port 5000 local-ip 10.10.1.12 remote-port 5000 remote-ip 10.10.1.11

Define Redundancy scheme: Creates

interdependency b/w CUBE redundancy & HSRP

CUBE 1 CUBE 2

Turn on CUBE Redundancy

IPC configuration : Allows

the ACTIVE CUBE to tell

the STANDBY about the

state of the calls. CONFIG

SHOULD BE APPLIED on

the LAN SIDE to avoid

SPLIT BRAIN

CUBE Configuration on ISR-G2 Box-to-Box Redundancy

redundancy inter-device scheme standby SB voice service voip mode border-element allow-connections sip to sip redundancy ipc zone default association 1 no shutdown protocol sctp local-port 5000 local-ip 10.10.1.11 remote-port 5000 remote-ip 10.10.1.12

interface GigabitEthernet0/0 ip address 10.10.1.11 255.255.255.0 standby version 2 standby 1 ip 10.10.1.13 standby delay minimum 30 reload 60 standby 1 name SB standby 1 preempt standby 1 track 2 decrement 10 standby 1 priority 50 interface GigabitEthernet0/1 ip address 128.107.60.71 255.255.255.0 standby version 2 standby 10 ip 128.107.60.73 standby delay minimum 30 reload 60 standby 10 preempt standby 10 track 1 decrement 10 standby 10 priority 50

interface GigabitEthernet0/0 ip address 10.10.1.12 255.255.255.0 standby version 2 standby 1 ip 10.10.1.13 standby delay minimum 30 reload 60 standby 1 name SB standby 1 preempt standby 1 track 2 decrement 10 standby 1 priority 50 interface GigabitEthernet0/1 ip address 128.107.60.72 255.255.255.0 standby version 2 standby 10 ip 128.107.60.73 standby delay minimum 30 reload 60 standby 10 preempt standby 10 track 1 decrement 10 standby 10 priority 50

Outside

interfaces:

HSRP group 10

Inside

interfaces:

HSRP group 1

CUBE 1 CUBE 2




description TO SERVICE PROVIDER

destination-pattern 9T


session target ipv4:y.y.y.y

voice-class sip bind control source-interface GigabitEthernet0/1

voice-class sip bind media source-interface GigabitEthernet0/1

!


description TO CUCM

destination-pattern 555….



voice-class sip bind control source-interface GigabitEthernet0/0

voice-class sip bind media source-interface GigabitEthernet0/0

!

ip rtcp report interval 3000

!

gateway

media-inactivity-criteria all

timer receive-rtcp 5

timer receive-rtp 86400

!

track 1 interface Gig0/0 line-protocol

track 2 interface Gig0/1 line-protocol

Bind traffic destined to the outside (SP SIP trunk)

to the outside Physical interface.

This ensures that all RTP and SIP packets are

created with the virtual IP associated with the

respective physical interface.

CUBE HA does not work with loopback interfaces

as they are always up

Configuration on Active and Standby

Bind traffic destined to the inside (CUCM or IP

PBX) to the inside Physical interface.

This ensures that all RTP and SIP packets are

created with the virtual IP associated with the

respective physical interface.

Configure media inactivity feature to clean up any

calls that may not disconnect after a failover

Configure Interface Tracking

• All signaling is sourced from/to the Virtual IP Address

• Lower address for all the interfaces (Gig0/0/0, Gig0/0/1, and Gig0/0/2) should be on the same platform

• Redundancy Interface Identifier, rii (HSRP Group number) should be unique to a pair/interface combination on the same L2

• Configuration on both the CUBEs must be identical including physical configuration and must be running on the same type of platform and IOS version. Loopback interfaces cannot be used as bind as they are always up.

• Multiple HSRP interfaces require interface tracking to be configured

CUBE

CUBE

CUBE-1

CUBE-2

GE 0/0/1 – 20.20.1.1

GE 0/0/1 – 20.20.1.2

20.20

.1.3

GE 0/0/0 – 10.10.1.1

GE 0/0/0 – 10.10.1.2

10.1

0.1

.3

CUCM

10.10.1.10

SP IP

Network

Y.Y.Y.Y

red

un

da

ncy

ri

i 1red

un

da

ncy

rii 2Keepalives

LAN Virtual IP

WAN Virtual IP

GE 0/0/2 – 3.3.1.1

GE 0/0/2 – 3.3.1.2

CUBE HA Design Considerations on ASR1K/ISR-4K for Box-to-Box Redundancy

CUBE HA Design Considerations on ASR1K/ISR-4K for Box-to-Box Redundancy – Cont’d

• Transcoding (LTI based) call flows are preserved starting XE 3.15. Requires same SPA-DSP capacity/placement

• Both platforms must be connected via the same physical Switch for CUBE HA to work. Cannot have WAN terminated on CUBEs directly or Data HSRP on either side

• Upon failover, starting XE3.11, the ACTIVE CUBE can be moved to PROTECTED state to avoid reload

• It is mandatory to use separate interface for redundancy (RG Control/data, Gig0/0/2). i.e interface used for traffic cannot be used for HA keepalives and checkpointing.

• CUBE B2B HA on ASR is not supported over a crossover cable connection for the RG-control/data link or across data centers

CUBE SIP Trunk Monitoring with OOD Options message

CUCM SIP Trunk SP SIP Trunk SP SIP

CUBE

A

OOD Options

200 OK

OOD Options

INVITE

DP 100 = ACTIVE

Timeout – no response

DP 100 = BUSYOUT

OOD Options

503 Service Unavailable

OOD Options

INVITE INVITE

200 OK 200 OK


voice-class sip options-keepalive

up-interval 20 down-interval 20 retry 3

Three timers that can be configured:

• up-Interval: OPTIONS keepalive

timer interval for UP endpoint

• down-interval: OPTIONS keepalive

timer interval for DOWN endpoint

• retry: Retry count for OPTIONS

keepalive transmission

Warning: • Each dial-peer that has options

message configured sends out a

separate message.

• EEM Script can be used to busyout

other dial-peers

OOD OPTIONS Ping Keepalive Enhancement

• Each dial-peer that has OPTIONS message configured sends out a separate message, even if the session targets are same

• Network bandwidth and process runtime are wasted in CUBE and remote targets to sustain duplicate OOD OPTIONS Ping heartbeat keepalive connection

• Consolidate SIP OOD Options Ping connections by grouping SIP dial-peers with same OOD Options Ping setup

• New CLI : “voice class sip-keepalive-profile <tag>” is used to define OOD OPTIONS Ping setup

• Consolidated SIP OOD Options Ping connection will then be established with a target for multiple SIP dial-peers with the same target and OOD Options Ping profile setup

CUCM SIP Trunk SP SIP Trunk SP SIP

CUBE

A

OOD Options (DP 100)

200 OK

DP 100 : Session Target IPv4:1.1.1.1

INVITE INVITE (DP 100)

200 OK 200 OK

DP 200: Session Target IPv4:1.1.1.1


200 OK



200 OK



200 OK

OOD OPTIONS Ping Keepalive Enhancement - Configuration

• With OOD Options Ping Keepalive group, an options ping keepalive connection is established on per remote target base as opposed an options ping keepalive connection established per dial-peer basis

• Up to 10,000 “voice class sip-options-keepalive <tag>” can be defined per system

• Either legacy “sip options-keepalive” or the new “sip options-keepalive profile <tag>” can be configured on a dial-peer

voice class sip-options-keepalive 1

description UDP Options consolidation

down-interval 49

up-interval 180

retry 7

transport udp





voice-class sip options-keepalive profile 1





voice-class sip options-keepalive profile 1

Single OOD Option

Ping Group applied

to multiple dial-peers

with same session

targets

Sample Show command output

CUBE#sh voice class sip-options-keepalive 1

Voice class sip-options-keepalive: 1 AdminStat: Up

Description: UDP Options consolidation

Transport: udp Sip Profiles: 0

Interval(seconds) Up: 180 Down: 49

Retry: 7

Peer Tag Server Group OOD SessID OOD Stat IfIndex

-------- ------------ ---------- -------- -------

1 4 Active 9

2 4 Active 10

OOD SessID: 4 OOD Stat: Active

Target: ipv4:10.104.45.253

Transport: udp Sip Profiles: 0

Contact Center Features

Mid-call Xcoder Insert/Drop – Codec Renegotiation

Call arrives on G.729 SIP trunk

CVP connects call to speech recognition server that requires G.711. Since provider does not support G711 CUBE inserts transcoder

CVP xfers call to a remote agent that uses G.729

CUBE drops xcoder and e2e call becomes G.729 again

1

2

3

4

Transcoder Inserted

Transcoder Dropped

SP SIP

CVP

SIP

G.711

G.729

G.729 / G.711

1

Provider supports only

G.729 codec

CUBE

G.711

2

Call Xfer (signaling only)

3

G.729

4

REFER Handling for Contact Centers

• Enables CUBE to handle REFER messages more efficiently in contact center deployments

• CUBE can operate in either consume mode or pass-through mode

SIP SP

CVP

1. REFER

3. INVITE

Based on “Refer-To” header,

CUBE does outbound dial-peer

match and sends out an INVITE

message No supplementary-service sip refer

supplementary-service media-renegotiate

CUBE will pass across the

Refer message “as-is” without

any modification

A

CUBE

2. INVITE

SIP SP

CVP

1. REFER

A

CUBE

2. REFER

REFER Consumption

REFER Pass-through (Default mode)

REFER Handling Enhancement

• A new CLI, “refer consume”, has been added to the SIP dial peer.

• The final decision to consume or pass-through REFER is determined based on this new CLI option configured on the Refer-To dial-peer.

“supplementary-service sip refer”

Configured globally or

at inbound dial-peer

“refer consume”

Configured at dial-

peer that matches

‘refer-To’

Outcome

Yes (default) No (default) REFER Pass-through

Yes (default) Yes REFER Consume

No No (default) REFER Consume

No Yes REFER Consume







Agenda

Serviceability

New CUBE Serviceability Features

Histogram for Call rate

Histogram for Concurrent calls

Histogram for Call duration

Histogram for SIP message rate

High/Low watermark for Call Rate

High/Low watermark for Concurrent calls

High/Low watermark for SIP message rate

Histogram for Call Failure Rate

High/Low watermark for Call Failure Rate

1122222357676678753222211111122247545789774322213311112245654598843333222

10

9 * *

8 * ** ***

7 * * *** * ***** * ##*

6 ******** * ***** ** *##*

5 *########* #* *####* *######*

4 *########* *#***####** *########*

3 **########** *#########** ** *########*****

2 ******#########***** ****##########**** ** ***########********

1 *######################################################################*

0....5....1....1....2....2....3....3....4....4....5....5....6....6....7..

0 5 0 5 0 5 0 5 0 5 0 5 0

Call switching rate / CPS (last 72 hours)

* = maximum calls/s # = average calls/s

Example:

show call history stats cps

Call Arrival Rate

Call History Stats – Graphical or Tabular form

show call history stats connected [table]

Last 60 sec, 60 minutes, 72 hours

Ability to sort dial-peers show run dial-peer sort

dial-peer (default) dial-peer sort dial-peer sort descending

dial-peer voice 4020 pots


port 0/2/0

!


destination-pattern 5...



!


incoming called-number 1...

port 1/0/0:23



port 1/0/0:23

!



port 0/2/0

!









!



port 0/2/0

!



port 1/0/0:23

Dial Peer tag

New CUBE Serviceability Features

Router# show call active total-calls

Total Number of Active Calls : 10

Total Number of Active Calls

A single call can have multiple call-legs. To determine the total number of active calls from call-legs is challenging

CLI added to display the value of current number of active (connected) calls on CUBE

The table defines the relation between call-legs and number of active calls

Call Flow Call-legs Connected

call

Basic call (audio/video) 2 1

Transferred call (Refer

handling)

3 2

Transcoded call (SCCP) 4 1

Calls after rotary/hunt 2 + x 1

Forwarded calls (CUBE

handling)

3 1

Forked call (media forking) 3 2

Forked call (signaling forking) 2 1

Avoiding Non-call-context Debug Logs

• Many times SIP debugs contain unrelated debugs that are not useful in debugging issues related to call failures

• Starting CUBE 10.0.1, non-call-context debugs will not be printed when debug ccsip is issued

• If a message is not part of any call, that debug will not be printed

• Affected messages: OPTIONS, REGISTER, SUBSCRIBE/NOTIFY

• To see the above messages in debugs, issue the following command

debug ccsip non-call

Debugging Made Easier

Router# debug ccsip feature < audio | cac | config | control | dtmf | fax | line | misc |

misc-features | parse | registration | sdp-

negotiation | sdp-passthrough | sip-profiles

| sip-transport | srtp | supplementary-

services | transcoder | video >

Categorize Debugs based on Functionality

Categorization based on Functionality

1. Audio/video/sdp/control

2. Configuration /sip-transport

3. CAC

4. DTMF/FAX/Line-side

5. Registration

6. Sdp - passthrough

7. Sip-profile/SRTP/transcoder

Example: enabling DTMF and audio debugs only with default log level is considered.

CUBE#sh debugging

CCSIP SPI: SIP info debug tracing is enabled (filter is OFF)

CCSIP SPI: audio debugging for ccsip info is enabled (active)

CCSIP SPI: dtmf debugging for ccsip info is enabled (active)

May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/verbose/32/sipSPI_ipip_store_channel_info: dtmf negotiation

done, storing negotiated dtmf = 0,

May 21 17:54:53.377: //444/5FE632EB8479/SIP/Info/info/2/sipSPIUpdateCallEntry:

Call 444 set InfoType to SPEECH

DTMF(32) debug code

Audio(2) debug code

Debugging Made Easier

CUBE# show cube debug category codes

Categorize Debugs based on Functionality

This CLI is used to collect the predefined debug features category codes , which helps in analysis of debugs manually.

|-----------------------------------------------

| show cube debug category codes values.

|-----------------------------------------------

| Indx | Debug Name | Value

|-----------------------------------------------

| 01 | SDP Debugs | 1

| 02 | Audio Debugs | 2

| 03 | Video Debugs | 4

| 04 | Fax Debugs | 8

| 05 | SRTP Debugs | 16

| 06 | DTMF Debugs | 32

| 07 | SIP Profiles Debugs | 64

| 08 | SDP Passthrough Deb | 128

| 09 | Transcoder Debugs | 256

| 10 | SIP Transport Debugs | 512

| 11 | Parse Debugs | 1024

| 12 | Config Debugs | 2048

| 13 | Control Debugs | 4096

| 14 | Mischellaneous Debugs| 8192

| 15 | Supp Service Debugs | 16384

| 16 | Misc Features Debugs| 32768

| 17 | SIP Line-side Debugs | 65536

| 18 | CAC Debugs | 131072

| 19 | Registration Debugs | 262144

|-----------------------------------------------


• CUBE Architecture





Agenda

Key Takeaways • It is a manageable transition from existing TDM based networks to SIP

networks using these network design techniques

• Enterprise SBC (Cisco Unified Border Element - CUBE) is an essential component of a UC solution providing;

• Security, Session Management, Interworking, Demarcation

• An unmatched set of features and functionality

• Proven interoperability with 3rd party PBX vendors and different service providers around the world (more than 160 countries)

• Now is the time to deploy SIP Trunking in either a Centralized or a Distributed solution to save money, simplify your topology and setup your infrastructure for future services

• Complete feature Presentation, Lab Guide & Application Notes :

https://cisco.box.com/cube




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Implementation & Management of Cisco Unified Border · CUBE (Enterprise) Product Portfolio 2900...

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