Implementation & Management of Cisco Unified Border Element...Cisco Unified Border Element Address...

Implementation & Management of Cisco Unified Border Element (CUBE) Enterprise

BRKUCC-2934

Hussain Ali

Technical Marketing Engineer

© 2014 Cisco and/or its affiliates. All rights reserved. BRKUCC-2934 Cisco Public 3

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3

© 2014 Cisco and/or its affiliates. All rights reserved. BRKUCC-2934 Cisco Public

Agenda

• SIP Trunking and CUBE Overview

• SIP Trunking Design & Deployment Models

• CUBE Architecture

• Transitioning to SIP Trunking using CUBE

• Advanced features on CUBE

• CUBE Management & Troubleshooting

• Futures & Key Takeaways

4


SESSION

CONTROL

Call Admissions

Control

Trunk Routing

Ensuring QoS

Statistics and Billing

Redundancy/

Scalability

INTERWORKING

SIP - SIP

H.323 - SIP

SIP Normalization

DTMF Interworking

Transcoding

Codec Filtering

DEMARCATION

Fault Isolation

Topology Hiding

Network Borders

L5/L7 Protocol

Demarcation

SECURITY

Encryption

Authentication

Registration

SIP Protection

Voice Policy

Firewall Placement

Toll Fraud

Enterprise 1

IP SIP

CUBE

IP Enterprise 2

IP CUBE

SIP

Rich Media (Real time Voice, Video, Screenshare etc.. ) Rich Media

Why does an enterprise need an SBC ?

5


VXML

SRST

Cisco Unified Border Element

Address Hiding

H.323 and SIP interworking

DTMF interworking

SIP security

Transcoding

Note: An SBC appliance would have only these features

Unified CM Conferencing and

Transcoding

IP Routing & MPLS

WAN & LAN Physical

Interfaces

CUBE

Voice Policy

TDM Gateway

PSTN Backup

FW, IPS, QoS

Cisco Unified Border Element – Router Integration

6

An Integrated Network Infrastructure Service

Note: Some features/components may require additional licensing


CUBE (Enterprise) Product Portfolio

2900 Series ISR-G2 (2901,

2911, 2921, 2951)

ASR 1004/6 RP2

Active Concurrent Voice Calls Capacity

CP

S


CUBE Session Capacity Summary

8

Platform CUBE Sessions

NanoCUBE (8XX and SPIAD Platforms) 15 - 120

2901 100

2911 200

2921 400

2951 600

3925 800

3945 950

3925E 2100

3945E 2500

4451-X (IOS-XE 3.11) 4000

ASR1001-X 10000

ASR1001/1002-X 10000

ASR1004/1006 RP2 16000

Introduced in Oct 2013

Introduced in July 2013

Introduced in May 2014

For Your

Reference


CUBE ISR and ASR Licensing

Platform Single-Use Licenses Redundancy Licenses

( 1 SKU for Active/Standby Pair)

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

Cisco 2901, 2911, 2921 ISR G2

FL-CUBEE-5

FL-CUBEE-25

FL-CUBEE-100

FL-CUBEE-5-RED

FL-CUBEE-25-RED

FL-CUBEE-100-RED

Cisco 2951, 3925 ISR G2

FL-CUBEE-5

FL-CUBEE-25

FL-CUBEE-100

FL-CUBEE-500

FL-CUBEE-5-RED

FL-CUBEE-25-RED

FL-CUBEE-100-RED

FL-CUBEE-500-RED

Cisco 3945, 3925E, 3945E ISR G2

ISR 4451-X

FL-CUBEE-5

FL-CUBEE-25

FL-CUBEE-100

FL-CUBEE-500

FL-CUBEE-1000

FL-CUBEE-5-RED

FL-CUBEE-25-RED

FL-CUBEE-100-RED

FL-CUBEE-500-RED

FL-CUBEE-1000-RED

Cisco ASR1000

FLASR1-CUBEE-100P

FLASR1-CUBEE-500P

FLASR1-CUBEE-1KP

FLASR1-CUBEE-4KP

FLASR1-CUBEE-16KP

FLASR1-CUBEE-100R

FLASR1-CUBEE-500R

FLASR1-CUBEE-1K-R

FLASR1-CUBEE-4K-R

FLASR1-CUBEE-16KR

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

NEW

For Your

Reference

9

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


ISR G2 CUBE Ent ASR Parity

with ISR

ASR

CUBE Vers.

2900/ 3900 FCS CUBE Vers.

IOS XE Release FCS

8.7 15.1.4M Apr 2011 ~50% 1.4.2 3.4 15.1(3)S July 2011

8.8 15.2.1T July 2011 ~70% 1.4.3 3.5 15.2(1)S Nov 2011

8.9 15.2.2T Nov 2011 >80% 1.4.4 3.6 15.2(2)S Mar 2012

9.0 15.2.3T/

15.2.4M Mar 2012 >85% 9.0 3.7 15.2(4)S July 2012

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

10.0.3 15.5(1)T Nov 2014 >95% 10.0.3 3.14 15.5(1)S Nov 2014

CUBE Software Release Mapping

10


Agenda








11

© 2014 Cisco and/or its affiliates. All rights reserved. BRKUCC-2934 Cisco Public 12

Cisco Session Management & CUBE: Essential Elements for Collaboration

• CUBE provides session border control between IP networks

– Demarcation

– Interworking

– Session control

– Security

• Cisco SME centralizes network control

– Centralizes dial plan

– Centralized applications

– Aggregates PBXs

12

Video

Mobile

SIP TRUNK TO CUBE

3rd Party IP

PBX

TDM PBX

CUBE

Cisco Session Management IM, Presence,

Voicemail

Cisco B2B


CUBE Deployment Scenarios

SIP Trunks for PSTN Access

Network-based Media

Recording Solution

SIP

H.323

SP VOIP

Services SBC

TDM

SIP Trunk

Partner API MediaSense

CUBE

SIP

RTP

SIP

Active

Standby

SP IP

Network SBC

Extending to Video and High Availability for Audio Calls

IVR Integration for Contact

Centers SIP

CVP vXML Server

Media Server

SP IP

Network SBC

Business to Business

Telepresence

SP IP

Network

SIP SIP

SBC

CUBE

CUBE

CUBE CUBE

CUBE

13


The Centralized Model

14

Characteristics of Centralized Operational Benefits Challenges

• Central Site is the only location with

SIP session connectivity to IP PSTN

• Voice services delivered to Branch

Offices over the Enterprise IP WAN

(usually MPLS)

• Media traffic hairpins through

central site between SP and

branches

• Centralizes Physical

Operations

• Centralizes Dial-Peer

Management

• Centralizes SIP Trunk

Capacity

• Increased campus bandwidth, CAC, latency; media optimization

• HA in campus

• Survivability at branch (PSTN connection at the branch)

• Emergency services

• Legal/Regulatory

IP PSTN

Centralized

Enterprise IP WAN

CUBE

Site-SP Media


Characteristics of Distributed Operational Benefits Challenges

• Each site has direct connection

for SIP sessions to SP

• Takes advantage of SP session

pooling, if offered by SP

• Media traffic goes direct from

each branch site to the SP

• Leverages existing branch

routers

• No media hair-pinning thru any

site

• Lower latency on voice or video

• Built-in Redundancy strategy

• Quickest transition from

existing TDM

• Distributed dial-peer

management

• Distributed operational overhead

• IP addressing to Service Provider

from branch

Site-SP Media

Enterprise IP WAN

Distributed

IP PSTN

CUBE

CUBE CUBE CUBE CUBE

The Distributed Model

15


Characteristics of Hybrid Benefits

• Connection to SP SIP service is determined on a site by

site basis to be either direct or routed through a regional

site.

• Decision to route call direct or indirect based on various

criteria

• Media traffic goes direct from site to SP or hairpins

through another site, depending on branch configuration.

• Adaptable to site specific requirements

• Optimizes BW use on Enterprise WAN

• Adaptable to regional SP issues

• Built-in redundancy strategy

Enterprise IP WAN

IP PSTN Hybrid

CUBE CUBE

CUBE CUBE CUBE

.. and the Hybrid Model

16


In-Depth Explanation of SIP Deployment Models

Educate your customer on SIP Deployment Models

New White Paper will be posted by the end of January at the following URL:

www.cisco.com/go/cube

17

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


Agenda








18


ASR & ISR-G2 Architecture Comparison

19

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 (IOS-XE based) Architecture

RP IOS

CPU

I/O I/O

Control

Plane

Data Plane

ISR G2 Architecture


ASR & ISR-G2/4451-X Feature Comparison

20

General SBC Features ASR ISR-G2 4451-X

High Availability Implementation Redundancy-Group

Infrastructure HSRP Based

Redundancy-Group

Infrastructure

TDM Trunk Failover/Co-existence Not Available Exists Exists

Media Forking XE3.8

(Thousands of calls)

15.2.1T

(Upto 1250 calls) XE3.10

Software MTP registered to CUCM (Including

HA Support) XE3.6 Exists Exists

DSP Card SPA-DSP PVDM2/PVDM3 PVDM4

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

Transcoder Implementation Local Transcoder

Interface (LTI)

SCCP or LTI (starting IOS

15.2.3T) SCCP and LTI

Embedded Packet Capture Exists Exists Exists

Web-based UC API XE3.8 15.2.2T Exists

Noise Reduction & ASP Exists 15.2.3T Exists

Call Progress Analysis XE3.9 15.3.2T Exists

CME/SRST and CUBE co-existence Not Available Exists XE3.11

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


Agenda








21


High-density Dedicated

Gateways

Transitioning to Centralized SIP Trunking...

22

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

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

22

Enterprise

Branch Offices

Enterprise Campus


• 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...

23

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

24

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

25

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

26

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 ipv4 10.1.1.50 ipv4 20.20.20.20 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

27

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


LAN Dial-Peer Configuration

dial-peer voice 100 voip

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

incoming called-number 9T

session protocol sipv2

codec g711ulaw

dtmf-relay rtp-nte

Outbound Dial-Peer for calls from CUBE to CUCM


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

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


session target ipv4:

codec g711ulaw

dtmf-relay rtp-nte

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

Inbound Dial-Peer for calls from CUCM to CUBE

28

CUCM sending 9

+ All digits dialed

SP will be

sending 10 digits

inbound


WAN Dial-Peer Configuration


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

incoming called-number [2-9].........


codec g711ulaw

dtmf-relay rtp-nte

Outbound Dial-Peer for calls from CUBE to SP


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

translation-profile outgoing Digitstrip

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


voice-class sip bind control source gig0/1

voice-class sip bind media source gig0/1


codec g711ulaw

dtmf-relay rtp-nte

Catch-all for

all inbound

PSTN calls

Dial-peer for

making long

distance calls

to SP

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

Emergency, Informational calls etc.

Inbound Dial-Peer for calls from SP to CUBE

29


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

30

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 "" ""

request INVITE sip-header To modify "" ""

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


Normalize Outbound SIP Message (Example 1)

31


request INVITE sip-header Diversion modify “sip:(.*>)” “sip:[email protected]>”

request REINVITE sip-header Diversion modify “sip:(.*>)” “sip:[email protected]>”


description Incoming/outgoing SP

voice-class sip profiles 400

Sent:

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

………

User-Agent: Cisco-SIPGateway/IOS-15.2.3.T

………

Diversion: ;privacy=off;

reason=unconditional;screen=yes

……...

m=audio 6001 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

……...

Configure

SIP Profiles

Apply to

Dial-peer or

Globally

Sent:


……….

User-Agent: Cisco-SIPGateway/IOS-15.2.3.T

……….

Diversion: ;

privacy=off;reason=unconditional;screen=yes

……….

m=audio 32278 RTP/AVP 18 8 101


………..

voice service voip

sip

sip profiles 400

SIP Provider

Requirement For Call Forward & Transfer scenarios back to PSTN, the Diversion

header should match the registered DID of your network

SIP INVITE that CUBE sends SIP INVITE that Service Provider expects

For Your

Reference


Normalize Inbound SIP Message (Example 2)

32


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


description Incoming/outgoing SP

voice-class sip profiles 400 inbound

Sent:


………

User-Agent: SP-SBC

………

Diversion: ;privacy=off;

reason=unconditional;screen=yes

……...

m=audio 6001 RTP/AVP 0 8 18 101


……...

Configure Inbound

SIP Profile to add a

dummy user part

Apply to Dial-peer

or Globally

Sent:


……….

User-Agent: SP-SBC

……….

Diversion: ;

privacy=off;reason=unconditional;screen=yes

……….

m=audio 32278 RTP/AVP 18 8 101


………..

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

For Your

Reference

Enable Inbound SIP

Profile feature voice service voip

sip

sip-profiles inbound


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)

33


Agenda








34


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

35

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

36


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" ;tag=1

To: ABC

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



37


CUBE

A


IP PSTN


Outbound Calls


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:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

Priority

A

B

C

D



38


CUBE

A


IP PSTN


Outbound Calls




answer-address 555




host ipv4:10.1.1.1


host ipv4:10.2.1.1









Received:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

A

B

C

D

Priority



39


CUBE

A


IP PSTN


Outbound Calls




answer-address 555




host ipv4:10.1.1.1


host ipv4:10.2.1.1









Received:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

A

B

C

D

Priority



40


CUBE

A


IP PSTN


Outbound Calls




answer-address 555




host ipv4:10.1.1.1


host ipv4:10.2.1.1









Received:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

A

B

C

D

Priority





41


CUBE

A


IP PSTN


Outbound Calls


answer-address 555




host ipv4:10.1.1.1


host ipv4:10.2.1.1









Received:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

A

B

C

D

Priority



42


CUBE

A


IP PSTN


Outbound Calls




answer-address 555




host ipv4:10.1.1.1


host ipv4:10.2.1.1









Received:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

A

B

C

D

Priority



43


CUBE

A


IP PSTN


Outbound Calls




answer-address 555




host ipv4:10.1.1.1


host ipv4:10.2.1.1









Received:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

1

2

3

4

A

B

C

D

Priority


Understanding Outbound Dial-Peer Matching Techniques

44


Number & carrier-id target

Match Based on URI of


& carrier-id target

Received:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

Match based on URI of an



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



45



carrier-id target orange


host ipv4:10.2.1.1


destination uri 2001




host ipv4:10.2.1.1




45

Received:



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

From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........


CUBE

A


IP PSTN


Inbound Calls

Outbound Calls

1

2

3

4

Priority



46





host ipv4:10.2.1.1






host ipv4:10.2.1.1




46

Received:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........


CUBE

A


IP PSTN


Inbound Calls

Outbound Calls

1

2

3

4

Priority



47





host ipv4:10.2.1.1






host ipv4:10.2.1.1




47

Received:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........


CUBE

A


IP PSTN


Inbound Calls

Outbound Calls

1

2

3

4

Priority



48





host ipv4:10.2.1.1






host ipv4:10.2.1.1




Received:




From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........


CUBE

A


IP PSTN


Inbound Calls

Outbound Calls

1

2

3

4

Priority



49


Number & carrier-id target

Match Based on URI of


& carrier-id target

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" ;tag=1

To: ABC


CSeq: 1 INVITE


Supported: timer

Max-Forwards: 70



Content-Length: 226

........

Match based on URI of an



number


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

50

Match based on DIVERSION Header of incoming INVITE

Match Based on URI of incoming INVITE message with or without carrier-id target

Received:




From: "555" ;tag=1

To: ABC


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.

51


Dial-peer Provision Policy Configuration 1. Define Voice Class Dial-peer Provision Policy

52

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

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

CUBE(config-class)#preference ?

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

description “Match outbound dial-peer based on criteria defined here”

preference 1 first-attribute second-attribute

preference 2 first-attribute second-attribute

For Your

Reference


Dial-peer Provision Policy Configuration – Cont’d

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

53


description Inbound Dial-peer

destination provision-policy

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

For Your

Reference


Dial-peer Provision Policy Configuration – Cont’d

54

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


Dial-peer Provision Policy Example – Match on FROM

55


user-id 555


host 10.2.1.1


description "Inbound dialpeer. Choose outbound based on DPP 10"

destination provision-policy 10




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



From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


........

voice class dial-peer provision-policy 10

description "Match outbound dialpeer on both From AND To Headers"

preference 1 from to

!


description "Match outbound DP based on FROM first, if no match

select based on TO"

preference 1 from

preference 2 to


description "Outbound dialpeer based on FROM"

destination uri-from 10


description "Outbound dialpeer based on TO"

destination uri-to 20


description "Outbound dialpeer based on FROM and TO"





56


user-id 555


host 10.2.1.1










From: "555" ;tag=1 To: ABC


CSeq: 1 INVITE


........




voice class dial-peer provision-policy 20 description "Match outbound DP based on FROM first, if no match

select based on TO"

preference 1 from preference 2 to













57


user-id 555


host 10.2.1.1










From: "555" ;tag=1 To: ABC


CSeq: 1 INVITE


........





select based on TO"

preference 1 from preference 2 to












Dial-peer Provision Policy Example – Match on TO

58


user-id 555


host 10.2.1.1










From: "555" ;tag=1

To: ABC Call-ID: [email protected]

CSeq: 1 INVITE


........





select based on TO"

preference 1 from

preference 2 to




shutdown









Dial-peer Provision Policy Example – Match on TO

59


user-id 555


host 10.2.1.1










From: "555" ;tag=1


CSeq: 1 INVITE


........





select based on TO"

preference 1 from

preference 2 to




shutdown









Dial-peer Provision Policy Example – Match on FROM & TO

60


user-id 555


host 10.2.1.1










From: "555" ;tag=1

To: ABC


CSeq: 1 INVITE


........






select based on TO"

preference 1 from

preference 2 to













61


user-id 555


host 10.2.1.1










From: "555" ;tag=1


CSeq: 1 INVITE


........

voice class dial-peer provision-policy 10 description "Match outbound dialpeer on both From AND To Headers"




select based on TO"

preference 1 from

preference 2 to













62

voice class uri 10 sip user-id 555

voice class uri 20 sip host 10.2.1.1










From: "555" ;tag=1


CSeq: 1 INVITE


........

voice class dial-peer provision-policy 10 description "Match outbound dialpeer on both From AND To Headers"




select based on TO"

preference 1 from

preference 2 to




shutdown









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

63

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 Destination-Patterns Under Same Outbound Dial-Peer

64


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 100

e164 919200200.

e164 510100100.

e164 408100100.


destination e164-pattern-map 100

codec g729r8

session target ipv4:10.1.1.1


url flash:e164-pattern-map.cfg


destination e164-pattern-map 100

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

destination-patterns targeted to the

same destination to be grouped into a

single dial-peer


Multiple Incoming Patterns Under Same Incoming Dial-peer

65


CUBE

IP PSTN A

(408)100-1010

(510)100-1010

(919)200-2010

(919)200-2000

(510)100-1000

(408)100-1000


e164 919200200.

e164 510100100.

e164 408100100.


description Inbound DP via Calling

incoming calling e164-pattern-map 100

codec g729r8


url flash:e164-pattern-map.cfg


description Inbound DP via Called

incoming called e164-pattern-map 200

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


URI Based Dialing Overview

Existing CUBE behavior:

• In CUBE URI based routing (user@host), the “user” part must be present and must be an E164 number

• The outgoing SIP ‘Request-URI’ and ‘To header URI’ are always set to the session target information of the outbound dial-peer

• For Req-URIs with same user name e.g. [email protected], [email protected], two different dial-peers are configured with the respective session targets

66

Enterprise

xyz.com Enterprise

abc.com

CUBE SBC

INVITE sip:[email protected]



URI Based Dialing Enhancement – URI Pass Through

• By default, the host portion is replaced with the session target value of the matched outbound dial-peer

• Enhancement : Outgoing INVITE has same request URI as received in Incoming INVITE. This can be achieved by configuring ‘requri-passing’ in the outgoing dial-peer or globally.

• Allows for peer-to-peer calling between enterprises using URIs

67


INVITE sip:[email protected] CUBE


incoming uri request 1 dial-peer voice 200 voip

session protocol sipv2 destination uri 1

voice-class sip call-route url session protocol sipv2


voice-class sip requri-passing


host cisco.com

For Your

Reference


URI Based Dialing Enhancement – ‘User’ portion non-E164 format

• By default, alphanumeric/non-E164 users were not allowed

• Enhancement : User part in Incoming INVITE Req-URI can be of Non-E164 format. e.g. sip:[email protected]. Outgoing INVITE will have user portion as it is received i.e. ‘hussain’ (unless SIP profiles are applied).

• Useful for video calls

68









host cisco.com

For Your

Reference


URI Based Dialing Enhancement – ‘User’ portion absent

• By default, call is rejected with “400 Bad Request”

• Enhancement : Incoming INVITE with no user portion (e.g. sip:cisco.com.) is supported. Dial-peer matching will happen based on ‘host’ portion. Outgoing INVITE Req-URI will not have any user portion in this case (unless sip-profiles are applied).

• If user portion is present in incoming INVITE ‘To header’, it is retained in outgoing INVITE ‘To Header’

• If ‘voice-class sip requri-passing’ is not configured, INVITE will go out as sip:10.1.1.1

• REFER and 302, both consume and pass-through cases supported as well

69

INVITE sip:cisco.com

INVITE sip:cisco.com CUBE






voice-class sip requri-passing


host cisco.com

For Your

Reference

sip:10.1.1.1


URI Based Dialing Enhancement – Deriving Target host from Incoming INVITE Req-URI

• For different hosts with the same ‘user’, multiple outgoing dial-peers had to be configured

• Enhancement : To support URIs with the same user portion but with different domains, only one dial-peer per can be configured. Outgoing dial-peer needs to be configured with ‘session target sip-uri’ instead of regular session target configuration. This will trigger DNS resolution of the domain of incoming INVITE Req-URI and dynamically determine the session target IP.

70







session target sip-uri


user hussain

user .*

Skype

Facebook Video

For Your

Reference


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 ”. The new “destination dpg ” 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

71


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




!






!






1. Incoming Dial-peer

is first matched 2. Now the DPG associated with the INBOUND DP is

selected

72


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

73

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-G2/4451-X)

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

port 2000

3. sccp configuration

sccp local GigabitEthernet0/0

sccp ccm

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

74

For Your

Reference


Configuration for LTI based Transcoding (ISR-G2/4451-X & ASR)

75

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


Mid-call Codec Renegotiation Transcoder Insert/Drop

76

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


Media Forking – Network Based Recording Solution

77

• Call agent independent

• Configured on a per Dial-peer level

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



• 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

MediaSens

e

Needs to

match

Dial-peer based


CUCM 10.X Recording UC Services API

Also see BRKUCC-2250

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

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

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

http://cucm:8090/ucm_xmfhttp://cucm:8090/ucm_xmf


CUBE Phone Proxy

• Enables B2BUA line side support in CUBE for CUCM

• Allows you to have phones on the Internet connected to a CUBE at the edge of the enterprise, replacing the need for ASA Phone Proxy by providing Secure RTP and TLS based communications on the leg away from CUCM

• CUBE Phone Proxy must have a Public IP Address and cannot be behind a NAT

• IP Phones can be behind a NAT

• Access Side : Connection between Phone and CUBE

• Core Side : Connection between CUCM and CUBE

TLS/SRTP

Public Network Users

SIP Line-Side

CUBE

SIP/RTP SIP

H.323 SP VOIP

Services SIP Trunk Side

SIP Line-Side CUBE

SBC

Phone Registration without VPN

79

ACCESS SIDE CORE SIDE


CUBE SIP Trunk Monitoring with OOD Options message

80

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

• Out-of-dialog OPTIONS message sent to check the status of the SIP Trunk

• The dial-peer is “busyout” if it does not receive a response within a configurable time period

• For an INVITE that matches a “busyout” dial-peer, CUBE sends “503 Service Unavailable”

• If there is a secondary dial-peer configured, the call will be re-routed the secondary path


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

81

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 ” 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

82


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 ” can be defined per system

• Either legacy “sip options-keepalive” or the new “sip options-keepalive profile ” 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

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