CCIE Training

Many businesses globally – large and small alike – have been converting calls from routing over traditional PSTN carrier trunks – such as E1 & T1 PRI or CAS – to much lower cost, yet still high performance, SIP ITSP (Internet Telephony Service Provider) trunks for years now. INE is no different than your business with regard to this – we have been using SIP trunks in lieu some traditional PSTN calling for years now as well. In fact, in response to a US Federal Communications Commission sub-commitee’s exploration on “PSTN Evolution” in December 2009, a representative from the US carrier AT&T described the traditional circuit-switched PSTN as “relics of a by-gone era”, and said that “Due to technological advances, changes in consumer preference, and market forces, the question is when, not if, POTS service and the PSTN over which it is provided will become obsolete” – source: Reuters [emphasis mine].

The challenge however, becomes that every SIP ITSP carrier has a slightly different way of implementing these sorts of trunks, and each has different provider network equipment that you, the customer, must connect to, and interoperate (properly) with. If you are a large national or multinational business, you may for instance sometimes even connect to two or three different types of provider network equipment, between possibly having multiple contracts with multiple carriers, and even sometimes having to deal with different provider equipment within a single carrier’s network.

Now while you both speak the same agreed upon language (namely SIP), it seems more often than not that you don’t always seem to speak exactly the same dialect of that language. This presents a major challenge in that calls and supplementary services (such as Hold, Resume, Blind Transfer, Semi-Attended Transfer, Fully-Attended Transfer, Forwarding, Faxing, etc) don’t behave as expected, or worse, that some functions don’t work at all.

It is not that SIP isn’t fully mature yet (it will turn 15 years old next year and has been widely used for over 10 years now), or that it is fully standardized and therefore governed by those IETF standards and working groups, it is simply that – as every one of us in the field for any respectable amount of time now knows – not every equipment vendor chooses to implement every single extension and option defined in every IETF RFC for SIP. I mean, have you ever actually looked at how many RFCs there are that deal with not just the core functions, but describe every option and extension to the SIP protocol? There are well over 100 RFCs! Therein lies the problem.

So what are we to do? Cisco Unified Border Element to the rescue! Today we will cover just a few of CUBE’s ability to perform SIP Normalization to allow optimum interoperability with many various SIP ITSPs.

At its base, CUBE consists of allowing both inbound and outbound call legs to be of the type VoIP. Here we first explore a very basic configuration where we have 2 Inbound/Outbound Dial-Peers (depending on which direction the call originates from) To/From the CUCMs in the cluster, and 2 Inbound/Outbound Dial-Peers To/From a fictional AT&T SIP ITSP trunk. We are allowing a codec negotiation and also possibly a DTMF relay internetworking between CUBE  and the CUCMs on Dial-Peer’s 101 & 102 (we needed both of these for another utility on this router using the SIP stack), while allowing for the codec of G.729 Annex B on the AT&T carrier side in Dial-Peers 1001 & 1002. We are also load balancing calls between both of the CUCM Subscriber servers and also between both of the SBCs on AT&T’s side that they have given us to peer with. We see this here:

!
ip domain retry 0
ip domain timeout 2
ip domain name ine.com
ip name-server 177.1.100.110
!
voice service voip
 address-hiding
 allow-connections sip to sip
 redirect ip2ip
 sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0
  header-passing error-passthru
  midcall-signaling passthru
  g729 annexb-all
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
!
dial-peer voice 101 voip
 description ** TO/FROM CUCM SUBSCRIBER 1 **
 destination-pattern 2065011...$
 voice-class codec 1
 session protocol sipv2
 session target ipv4:177.1.10.20
 incoming called-number .
 dtmf-relay sip-kpml rtp-nte
!
dial-peer voice 102 voip
 description ** TO/FROM CUCM SUBSCRIBER 2 **
 destination-pattern 2065011...$
 voice-class codec 1
 session protocol sipv2
 session target ipv4:177.1.10.25
 dtmf-relay sip-kpml rtp-nte
!
dial-peer voice 1001 voip
 description ** TO/FROM SIP ITSP - AT&T SBC 1 **
 destination-pattern +T
 voice-class sip localhost dns:corphqr1.ine.com
 session protocol sipv2
 session target dns:sip1.att.com
 incoming called-number 2065011...$
 dtmf-relay rtp-nte
 codec g729br8
!
dial-peer voice 1002 voip
 description ** TO/FROM SIP ITSP - AT&T SBC 2 **
 destination-pattern +T
 voice-class sip localhost dns:corphqr1.ine.com
 session protocol sipv2
 session target dns:sip2.att.com
 incoming called-number 2065011...$
 dtmf-relay rtp-nte
 codec g729br8
!
dial-peer hunt 1
!

Now what if we have a carrier who wants to see our specific domain name (ine.com) after the @ in the Contact header of a SIP INVITE request message (so 2065011001@ine.com   vs.   2065011001@177.1.254.1), possibly for something like compliance with SIP Asserted-Identity? Let’s look at what the SIP INVITE might look like prior to any modification to the above configuration:

Sent:
INVITE sip:+12065015111@sip2.att.com:5060 SIP/2.0
Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK2BAFFD
Remote-Party-ID: "Jack Shepherd" <sip:2065011001@corphqr1.ine.com>;party=calling;screen=yes;privacy=off
From: "Jack Shepherd" <sip:2065011001@corphqr1.ine.com>;tag=8074E2B0-20E7
To: <sip:+12065015111@sip2.att.com>
Date: Fri, 20 Aug 2010 02:34:27 GMT
Call-ID: 9FE12628-A81511DF-8700FC78-AA8D9DEB@corphqr1.ine.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2682052728-2819953119-2264595576-2861407723
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1281926067
Contact: <sip:2065011001@177.1.254.1:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 292

v=0
o=CiscoSystemsSIP-GW-UserAgent 5117 3857 IN IP4 177.1.254.1
s=SIP Call
c=IN IP4 177.1.254.1
t=0 0
m=audio 16532 RTP/AVP 18 100 19
c=IN IP4 177.1.254.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=rtpmap:19 CN/8000
a=ptime:20

!
voice class sip-profiles 1
 request INVITE sip-header Contact modify "<sip:(.*)@(.*):5060>" "<sip:\1@ine.com:5060>"
!
dial-peer voice 1001 voip
 voice-class sip profiles 1
!
dial-peer voice 1002 voip
 voice-class sip profiles 1
!

Now let’s take a look at what that did to the contents of our Contact header in a new call, and thus a new SIP INVITE message that we send out to AT&T:

Sent:
INVITE sip:+12065015111@sip2.att.com:5060 SIP/2.0
Via: SIP/2.0/UDP 177.1.254.1:5060;branch=z9hG4bK2BAFFD
Remote-Party-ID: "Jack Shepherd" <sip:2065011001@corphqr1.ine.com>;party=calling;screen=yes;privacy=off
From: "Jack Shepherd" <sip:2065011001@corphqr1.ine.com>;tag=8074E2B0-20E7
To: <sip:+12065015111@sip2.att.com>
Date: Fri, 20 Aug 2010 02:34:27 GMT
Call-ID: 9FE12628-A81511DF-8700FC78-AA8D9DEB@corphqr1.ine.com
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE:  1800
Cisco-Guid: 2682052728-2819953119-2264595576-2861407723
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1281926067
Contact: <sip:2065011001@ine.com:5060>
Expires: 180
Allow-Events: telephone-event
Max-Forwards: 69
Session-Expires:  1800
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 292

v=0
o=CiscoSystemsSIP-GW-UserAgent 5117 3857 IN IP4 177.1.254.1
s=SIP Call
c=IN IP4 177.1.254.1
t=0 0
m=audio 16532 RTP/AVP 18 100 19
c=IN IP4 177.1.254.1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=rtpmap:19 CN/8000
a=ptime:20

Excellent! It did exactly what we asked it to!

There are many other things that the Cisco’s UBE can do for us, and we have only covered one small one here in this article. For a lot more great information on this product check out INE’s Class-on-Demand CCIE Voice Deep Dive for CUBE. By the way, Cisco’s implementation of what others in the industry might label a “SBC” (Session Border Controller), goes far beyond what other industry SBCs are able to offer in terms of both features and scalability (CUBE hardware support ranges from ISRs for SMBs, up through ISR-G2s and ASRs for Enterprises, up to the 12000 series routers for SPs). I will cover many more of the offered features of the CUBE in follow-up postings, so stay tuned!

I will leave you with a great Cisco article describing some basic functionality of CUBE and SIP Normalization, and also a lot of great Cisco configuration examples from live SIP ITSP trunks that Cisco has installed and tested with in their RTP labs, as well as live PBX integrations that they have performed, and subsequently written up these “recommended practice” documents.

A common question for the CCNA Voice candidate is – “Just how can we translate our analog voice waveform into the digitized form that is required for Voice over IP transmission through the converged network?” These active reading questions tell the story – enjoy!

Me Singing in the Shower!

Questions Only

In its natural form, what signal type is the human voice?

To send voice as a series of zeros and ones is known as what type of encoding?

Converting analog voice into digital data begins with taking “snapshots” of voice very frequently. This is called what?

If you sample voice too infrequently, the listener might hear a different sound. This is called what?

What is the reason that you would not want to oversample voice?

What theorem teaches that the sample rate needs to be twice as high as the highest frequency being sampled?

Based on the Nyquist Theorem, how often should we sample voice?

What is the initial process of voice sampling called?

What is the process of taking PAM amplitudes and assigning them a number?

What type of scale is used in voice quantization?

Rounding off during quantization can cause a “hiss” on the line. This is called what?

When is quantization error more noticeable?

What approach to the quantization logarithmic scale is commonly used in North America?

What approach to the quantization logarithmic scale is commonly used outside North America?

If VoIP equipment connects from different countries, what quantization logarithmic scale is used?

What is the breakdown of the 8 bits in a voice sample?

When considering voice sampling only- how much bandwidth is required to send voice and how is the value arrived at?

The process of encoding and decoding a wave form to save bandwidth is accomplished by what type of technology?

What type of codec does not actually compress the waveform and what is an example?

What type of codec sends the difference in the current sample versus the previous sample and what is an example?

What type of codec dynamically builds a codebook based on speech patterns and what is an example?

What is the most popular codec in the Cisco VoIP environment for sending voice over the WAN and why?

What is the bandwidth for a voice call required under the G.711 codec?

What is the bandwidth required for a voice call under the G.729 codec?

What type of codec is very similar to CS-ACELP but uses a smaller codebook and what is an example?

What are the bandwidth and delay characteristics of LDCELP?

What codec is typically used in the LAN in a Cisco VoIP environment?

What variation of G.729 uses a less complex algorithm?

What variation of G.729 enables VAD?

What is VAD?

What are four other factors that impact the size of a voice packet?

What VoIP quality measurement uses a trained ear to rate quality on a scale of 1 to 10?

What quality measurement digitally measures the difference in the original signal and the signal after it passes through the codec?

What variation of PSQM attempts to match the measurement with MOS?

Calculating the number of telephone calls during the busiest time of day is referred to as what?

What is an Erlang?

What is the formula for calculating the number of call minutes a corporate phone system uses during the busiest hour of the day?

What is the percentage of call to reject during the busiest hour of the day referred to as?

Questions and Answers

In its natural form, what signal type is the human voice?

Analog

To send voice as a series of zeros and ones is known as what type of encoding?

Binary

Converting analog voice into digital data begins with taking “snapshots” of voice very frequently. This is called what?

Sampling

If you sample voice too infrequently, the listener might hear a different sound. This is called what?

Aliasing

What is the reason that you would not want to oversample voice?

Requires too much bandwidth

What theorem teaches that the sample rate needs to be twice as high as the highest frequency being sampled?

The Nyquist Theorem

Based on the Nyquist Theorem, how often should we sample voice?

Every 125 ms

What is the initial process of voice sampling called?

PAM (Pulse Amplitude Modulation)

What is the process of taking PAM amplitudes and assigning them a number?

Quantization

What type of scale is used in voice quantization?

Logarithmic

Rounding off during quantization can cause a “hiss” on the line. This is called what?

Quantization error

When is quantization error more noticeable?

At lower volumes

What approach to the quantization logarithmic scale is commonly used in North America?

Mu-Law

What approach to the quantization logarithmic scale is commonly used outside North America?

a-Law

If VoIP equipment connects from different countries, what quantization logarithmic scale is used?

a-Law

What is the breakdown of the 8 bits in a voice sample?

1 Polarity Bit; 3 Segment Bits; and 4 Step Bits

When considering voice sampling only- how much bandwidth is required to send voice and how is the value arrived at?

8000 samples per second * 8 bits per sample = 64 Kbps

The process of encoding and decoding a wave form to save bandwidth is accomplished by what type of technology?

A Codec

What type of codec does not actually compress the waveform and what is an example?

Pulse Code Modulation (PCM); an example is G.711

What type of codec sends the difference in the current sample versus the previous sample and what is an example?

Adaptive Differentiated PCM (ADPCM); an example is G.726

What type of codec dynamically builds a codebook based on speech patterns and what is an example?

Conjugate Structure Algebraic Code Excited Linear Prediction (CS-ACELP); G.729 is an example

What is the most popular codec in the Cisco VoIP environment for sending voice over the WAN and why?

G.729; decent quality with lower bandwidth requirements

What is the bandwidth for a voice call required under the G.711 codec?

64 Kbps

What is the bandwidth required for a voice call under the G.729 codec?

8 Kbps

What type of codec is very similar to CS-ACELP but uses a smaller codebook and what is an example?

Low-Delay Conjugate Excited Linear Prediction (LDCELP); G.728

What are the bandwidth and delay characteristics of LDCELP?

It reduces delay, but at the cost of higher bandwidth

What codec is typically used in the LAN in a Cisco VoIP environment?

G.711

What variation of G.729 uses a less complex algorithm?

G.729a

What variation of G.729 enables VAD?

G.729b

What is VAD?

Voice Activity Detection – devices will not send “the sound of silence” with VAD

What are four other factors that impact the size of a voice packet?

Media, Tunneling, Header Compression, Codec used

What VoIP quality measurement uses a trained ear to rate quality on a scale of 1 to 10?

Mean Opinion Score (MOS)

What quality measurement digitally measures the difference in the original signal and the signal after it passes through the codec?

PSQM (Perceptual Speech Quality Measurement)

What variation of PSQM attempts to match the measurement with MOS?

PESQ (Perceptual Evaluation of Speech Quality)

Calculating the number of telephone calls during the busiest time of day is referred to as what?

Traffic Engineering

What is an Erlang?

One solid hour of phone usage

What is the formula for calculating the number of call minutes a corporate phone system uses during the busiest hour of the day?

[Monthly_Call_Minutes/22] * .15

What is the percentage of call to reject during the busiest hour of the day referred to as?

GOS (Grade of Service)

As I mentioned in a blog article a few months back, INE installed 7961 phones in all of our Voice Racks used for customer rental, and simultaneously struck up an exclusive deal with the company called VoIP Integration so that our students could buy their fantastic Remote Phone Control Software at quite a significant discount. More students than I ever imagined would, have contacted me and received the INE Student Edition of this software for use during their studying sessions. In fact, many that already have phones of their own, bought the software just so that they could travel and stil practice labs – while on the road – with their phones and without having to pack up and take their phones with them.

This is the very tool we use during all of the live online and recorded CCIE Voice Deep Dive’s to show you exactly what is happening in real-time on the actual phones (which happen to be in front of me so that you can hear what is going on as well) for any given task we are all working through configuring or testing & troubleshooting.

Well, now VoIP Integration has updated their Remote Phone software to version 2.1 and brought quite a significant update in features. Standing out clearly in front of all of them – even in front of the great performance improvements – is support for IP Phones registered to Cisco Unified Communications Manager Express and IOS CME as SRST!

I have provided the basic configuration here that you would need to support them in CME.  Important NOTE: Don’t forget the “type” command on your ephone. Without it, CME doesn’t actually build the CNF files for that phone, and while they will register using the Default.cnf.xml file, that file doesn’t contain the necessary Authentication URL.

!
ip http server
no ip http secure-server
ip http path flash:gui
!
ixi transport http
 response size 4
 no shutdown
 request outstanding 1
!
ixi application cme
 no shutdown
!
telephony-service
 max-ephones 1
 max-dn 1
 ip source-address 177.1.254.3 port 2000
 url authentication http://177.1.254.3/CCMCIP/authenticate.asp admin cciecisco
 log password cciecisco
 create cnf-files
!
!
ephone-dn  1 dual-line
 number 3001
!
!
ephone  1
 mac-address 001B.5452.D7BD
 type 7961
 button  1:1
!

Download it and give it a spin, and read more about the support (found in Appendix C) of their latest Administration Guide.

Also, If you would care to purchase a copy of INE’s Student Edition of the VoIP Integration Remote Control software, please ping me at msnow at ine dot com.

The author and poet Maya Angelou said “Words mean more than what is set down on paper. It takes the human voice to infuse them with deeper meaning.”. Well that is certainly what we have attempted to do with the CCIE Voice Deep Dive self-paced Class on Demand series – that is to bring the human instructional voice element to infuse deeper meaning to what is already fantastic Cisco Documentation. Anyone that has set out and determined to undertake the task of studying for and ultimately passing any CCIE Lab exam, knows that at some point during your studies, the words on paper (Cisco Docs, RFCs, books) – while a absolute phenomenal source of information – can at times seem to loose their impact. Perhaps you have been studying too long, read one too many docs, have the time pressure of your family and friends waiting for you to return to be a part of their life, or perhaps you are just starting out on your adventure and don’t know where to begin. Whatever stage you are at or whatever the case may be, it is certainly helpful to have a tutor and mentor there beside you at times, assisting you in understanding what each complex technology’s documentation is trying to teach you, in possibly a deeper and more insightful way than you can manage on your own.

Wait no longer for such help to arrive! INE is happy to announce that each Live-Online Deep Dive course that we have taught has been recorded, and you have the ability to access these extensive repositories of knowledge at any time.

Here are a couple of great demo’s of just a portion of the latest Deep Dive session we held on Globalization & Localization in order to whet your appetite:

Demo 1: Globalization Prezi – Theory and Reasons

Demo 2: Inbound Calling Party Localization

For each complex topic we have held — or will soon hold (listings to follow below) — a separate online class where we dive down deep and explore all the concepts, practical application and troubleshooting associated with each technology topic. We then allow you to purchase each module individually (if you like) so that you can either try small sections of the product, or so that those who only need to plug in small gaps of knowledge can do so at a very deep, intense level – either one without committing to purchase the entire product series.

The general format for each Class-on-Demand Deep Dive module spends between 4-7 hours on the given topic for that day, and during that time follows this outlined training methodology:

• Collectively discuss and teach all concepts involved in the technology
• Whiteboard concepts to further deepen every participant’s understanding
• Define a specific set of tasks to be accomplished
• Demonstrate how the tasks and concepts are implemented and properly configured
• Test the configuration thoroughly
• Vary the configuration to understand how different permutations effect the outcome
• Debug and trace the working configuration to understand what should be seen
• Break the configuration and troubleshoot with debugs and traces to contrast from the working set

Thus far, we have held 10 online sessions – each with a median recorded runtime of 6 hours. We have almost 60 hours of Class on Demand content, and we’ve only just begun! We conservatively estimate that by the time we complete our more than 30 planned modules, that we will have at over 200 hours of Deep Dive recordings.

Below is a detailed index from the 10 currently available sessions:

Module 1 :: Network Infrastructure with LAN Quality of Service

• Catalyst 3560/3750 Classification and Marking
• Catalyst 3560/3750 Conditional Trust
• Catalyst 3560/3750 Ingress Interface Mapping
• Catalyst 3560/3750 Ingress Interface Queuing
• Catalyst 3560/3750 Ingress Interface Expedite Queue
• Catalyst 3560/3750 L2 CoS to L3 DSCP Mapping
• Catalyst 3560/3750 Egress Interface Mapping
• Catalyst 3560/3750 Egress Interface Queuing
• Catalyst 3560/3750 Interface Queue Memory Allocation
• Catalyst 3560/3750 Egress Queue-Set Templates
• Catalyst 3560/3750 Weighted Tail Drop (WTD) Buffer Allocation
• Catalyst 3560/3750 Egress Interface Expedite Queue
• Catalyst 3560/3750 Egress Interface Sharing
• Catalyst 3560/3750 Egress Interface Shaping
• Catalyst 3560/3750 Scavenger Traffic Policing

Module 02 :: CUOS GUI and CLI Admin

• CUCM WebUI: Service Activation and Stop/Start/Reset
• CUCM WebUI: Bulk Administration Tool (Import/Export, Phone Reports, etc)
• CUCM WebUI: DB Replication Status
• CUCM WebUI: Trace Files
• CUOS CLU: TFTP Files Management
• CUOS CLU: Status and Hostname
• CUOS CLU: DB Replication Assurance
• CUOS CLU: DB Replication Repair and Cluster Reset
• CUOS CLU: Trace Files
• CUOS CLU: RIS DB Search
• CUOS CLU: Performance Monitor (PerfMon)
• RTMT: Trace Files
• RTMT: Performance Monitor (PerfMon)

Module 03 :: CUCM System and Phone – SCCP and SIP Fundamentals

• CUCM Services
• UC Servers and Groups
• Date/Time with NTP Reference
• Regions and Codecs
• Location-Based Call Admission Control
• SRST References
• Device Pools
• System Parameters
• Enterprise Parameters
• Phone Button Templates
• Softkey Templates
• SCCP Phone Basics
• SIP Phone Basics

Module 04 :: Users, Credentials, Multi-Level Roles and LDAP Internetworking

• CUCM User Credentials and Policies
• LDAP Synchronization for CUCM and Unity Connection
• LDAP Authentication for CUCM and Unity Connection
• CUCM End Users
• CUCM User Roles
• CUCM Multi-Level Administration
• CUCM Device/Phone/Line User Association
• UCCX and CUP Basic Users

Module 05 :: Call Features – In-Depth

• SCCP and SIP Phone Display
• Phone Firmware
• Phone Logging
• Ring Settings
• Basic and Advanced Call Forwarding Display
• Auto-Answer Options
• CallBack (Camp-On)
• Intercom
• Advanced Call Hold Options
• Call Park
• Directed Call Park
• Advanced Call Park Settings
• Call Pickup
• Group Call Pickup
• Other Call Pickup
• Directed Call Pickup
• Call Pickup Attributes
• Shared Line
• Barge and cBarge (Conference Barge)
• Privacy
• Built-In IP Phone Bridge

Module 06 :: Media Resources – MTPs, Conf Bridges, Annunciator and Music on Hold

• IOS Software MTP
• IOS Conference Bridge
• IOS Transcoding
• Media Preference and Redundancy
• Meet-Me Conferencing
• Ad-Hoc Conferencing
• Annunciator
• Unicast Music on Hold
• Traditional Multicast Music on Hold
• Alternate Multicast Music on Hold

Module 07 :: Expert Gateways & Trunks

• ISDN Switch Types and Advanced CNAM options
• ISDN Information Elements
• SIP Trunks – Fundamental and Advanced Options
• H.323 Gateways – Fundamental and Advanced Options
• MGCP Gateways – Fundamental and Advanced Options

Module 08 :: Expert H.323 Gatekeeper

• Provisioning IOS H.323 Gatekeeper
• Registering CUCM with H.323 Gatekeeper
• Registering CUCME with H.323 Gatekeeper
• Routing Calls from CUCME to CUCM via Gatekeeper in Multiple Zones with Dynamic E.164 Aliases
• Routing Calls from CUCM to CUCME via Gatekeeper in Multiple Zones with Multiple Tech Prefixes
• Routing Calls from CUCME to CUCM via Gatekeeper in Multiple Zones with Multiple Tech Prefixes
• Routing Calls from CUCME to CUCM via Gatekeeper in Multiple Zones with Static E.164 Aliases
• Routing Calls from CUCM to CUCME and Back via Gatekeeper in One Zone with One Tech Prefix
• Gatekeeper Call Admission Control
• Routing Calls from CUCM to CUCME and Back via Alternate Gatekeeper Clustering in Multiple Zones with Multiple Tech Prefixes using GUP

Module 09 :: Dial Plan – Line Device Approach and the Not-So-Basic Fundamentals

• Class of Service: Calling Search Spaces and Partitions
• Gateways, Route Groups, Local Route Groups/Device Pools
• Route Lists and Standard Local Route Groups
• Route Patterns and Translation Patterns
• Digit Manipulation: Calling & Called Party Transformations and IOS Dial Peers
• Private Line Automatic Ringdown (PLAR)

Module 10 :: Dial Plan – Globalization & Localization of both the Calling and the Called Numbers, and with Mapping the Global Number to the Local Variant

• Inbound PSTN Calls (Ingress from PSTN, Egress to Phones): Calling Party Globalization :: GW Incoming Calling Party Settings
• Inbound PSTN Calls (Ingress from PSTN, Egress to Phones): Calling Party Localization :: Phone Calling Party Transformations
• Outbound PSTN Calls (Ingress from Phones, Egress to PSTN): Called Party Globalization :: PSTN Patterns – a.k.a. “Translation Patterns are the *New* Route Patterns”
• Outbound PSTN Calls (Ingress from Phones, Egress to PSTN): Called Party Localization :: Digit Manipulation: Calling & Called Party Transformations and IOS Voice Translation Rules & Dial Peers
• Mapping the Global Number to the Local Variant :: + Dialing and One-Button Missed Call DialBack

So stay tuned to this blog as we will shortly post the upcoming modules soon to be held online and recorded.

Summer is here and it’s time to get certified!  Join us during the Summer of Success by attending one of our bootcamps and save up-to $1000.  Get $500 off any one week on-site bootcamp or $1000 off any two week on-site bootcamp when you purchase during this limited offer.  This special promotion applies to CCIE Routing & Switching, CCIE Voice, CCIE Service Provider or CCIE Security.

To take advantage of this special promotion, use discount code 1WEEKBOOTCAMP or 2WEEKBOOTCAMP when you check out at http://www.ine.com.

Coming June 7th, 2010 – CCIE Voice Deep Dive

INE knows Voice. As the only CCIE vendor on the market employing 4 CCIE Voice Instructors, we are constantly trying to pool our collective brain-trust to find a better way to communicate what we know to you, to help you achieve your goals in passing this rigorous practical exam. We’ve all been mulling over for some time now the best possible way to get this information from our heads, to yours. We gave the idea a go where each of us would perform a vulcan mind-meld with the student, however there are still a few roadblocks in the way of us being able to achieve success with this method. The biggest issue we kept running into was that once Petr would step into the room and begin his knowledge transfer, the candidate’s brain would often overheat and we ended up taking a few too many to the hospital to have their brains cooled down with liquid nitrogen. Needless to say – we haven’t quite worked out all of the technical glitches out of our method just yet.

So instead the idea was collectively reached that we should instead develop a brand new product that allows us to perform a sort of “Deep Dive” with each candidate. The concept of this “deep-dive” method goes far beyond what most candidates expect and subsequently receive when purchasing some fashion of Self-Paced On-Demand or Classroom Bootcamp style training. Instead we take one subject, whatever that subject may be for the day, and we vet it fully. To that end I mean that we don’t stop collectively learning until the concept is grasped fully by all candidates.

The format of each class module will be focused around roughly a 4-5 hour window of instruction per topic, however some topics will inevitably go longer. If a topic spans more than the time allotted in a single day’s module, then and there another day to complete the materials will dynamically be scheduled, and any attending students will automatically be added to the continued new module.

The specific breakdown for each module will look like this:

• I will lead the discussion for each and every class module
• We will collectively discuss and fully understand all concepts involved in the technology topic for a given day
• We will then define a very specific set of  Tasks to be accomplished
• We will whiteboard the Tasks, notes about them, and how they will be logically implemented*
• We will then demonstrate with live, hands-on interaction, how the concepts are implemented and properly configured
• We will test the configuration thoroughly
• While testing, I will vary the configuration so that we all can see how different permutations effect the outcome
• We will Debug and Trace the working configuration to understand what we ’should be seeing’
• We will then break the configuration and Troubleshoot with more Debugs and Traces to contrast from the working set

*Each whiteboard sketching will available for download after each module as separate JPG or PDF documents

Engaging students for 4-5 hours per day on a single technology topic allows us to cover that technology in much more depth than could ever be accomplished in a normal classroom setting, where an instructor typically has to get through all of the technologies covered by the CCIE Lab in 4-5 days. Also by not deviating from a single topic from day-to-day, allows the student to truly focus-in on the single technology topic of the day, and truly master it. This also gives students the ability to span their education out over multiple days and weeks and still accomplish normal business demands that are, of course, inevitable.

Every module will be recorded, which will give the attending students the ability to revisit any topic at any time, or even to miss the occasional class day if need be, without being left behind when attending future classes. Each module once recorded, will also available for individual purchase as well, giving students the ability to ‘brush-up’ on a few technologies, without having to commit to purchasing the entire series.

The schedule for these Deep Dive’s can be found by clicking here. And what’s better is that each class module can be purchased individually. If you think you only need to brush up or possibly “go deeper” into a single subject? Just sign up for that one day. More on pricing can be found by clicking here.

Many more modules and weeks are being added to the schedule early next week, don’t think for a minute that this is it. As mentioned, there won’t be a single stone or topic left unturned. Once we finish covering all of the topics that can possibly be tested on for the practical exam environment, we will continue on with practical “real-world” scenario topics in Unified Communications that aren’t covered by the lab blueprint, and then continue on with “real-world” design modules.

Hope to see you in a class soon!

As you know, purchasers of the new INE 5-day QoS bootcamp receive the class in four different modalities. There is the interactive self-paced version, the live class, the recorded live class, and an audio bootcamp.

If you would like to check out a sample lesson of the audio bootcamp, tune into W-INE Internet radio, or visit the course’s Samples page. The lesson is also going to appear on our iTunes podcast channel by Saturday, May 22, 2010. Just search the iTunes store for INE.

Enjoy, and I look forward to “seeing you” in class soon.

That is a very true saying – in fact one that we believe strongly in here at INE. However we also understand just how expensive it can be to undertake studying for any CCIE Lab exam. That is why, whenever we can, we try to reduce the load on you – the students – to bear this cost. Take for instance our CCIE RS Volume II for Dynamips – we do all we can to provide you the best available instruction while trying to reduce, or sometimes even be able to eliminate the hardware costs associated with studying.

So now we have taken to task trying to do the same for our CCIE Voice track products. We can’t quite virtualize all of the routers used as voice gateways (pesky DSP’s and TDM trunk cards that dynamips won’t ever be able to support since we actually need hardware for the drivers to be able to trigger the signaling), but we thought we would try to reduce the hardware cost for you, the student, in any way we can with the necessary hardware. Anyone having decided to study for the CCIE Voice lab exam has no doubt realized that even if you decide not to take on the enormous cost of building your own rack to practice with, and instead, to rent rack time from any vendor on the market, you still must purchase your own hardware 7961 IP Phones along with some sort of a hardware VPN solution (such as an ASA 5505 or 851 ISR router) at a minimum in order to be able to practice for all of the most important features tested in the lab. This is quite simply due to the fact that the much older 7960’s and all current SCCP Software Client phones (including Cisco CIPC, IPBlue VT-GO*, etc) don’t support any of the newer features – those that are most critical to studying for the latest lab exam. Even if you can manage to find refurbished 7961 IP Phones from eBay for roughly $150/phone and $500/ASA5505 – you still have to invest over $1,000USD just in hardware before you are ready to rent the rack! Seeing as how the 7961 phones are already a generation behind the current ones, and the possibility that when you pass your lab 6-12 months from now that they will likely be 2 generations old and harder to sell for the same price you paid for them – it becomes a very expensive venture to undertake!

To that end, we have made the decision to equip each of our voice racks with three 7961 phones (one at each “site” within the rack topology). Now the only way adding these phones to our racks makes any sense for our students is if we give them some way of controlling them. So we decided to form an exclusive partnership with the most premier remote phone control software company on the market – VoIP Integration. They recently upgraded their very popular Remote Phone software and have added a bunch of features, not to mention made the screen refresh rate a lot faster. Anyway – onto the good stuff. If you are a current (or future) Voice customer of INE, our strategic partnership with them will allow you to purchase their normally $199 Remote Phone software for a significantly reduced cost. One license will allow you to run multiple instances of the software in order to allow you to control all of the phones connected to your rack from a single PC.

If you have the ability to procure these hardware 7961 phones, they are still a great study option, and we still of course still support both IPSec EzVPN in Network Extension Mode and SSL VPN for everyone of our 14 Voice racks, just as we have done for well over a year now. And of course every live Voice course that we teach uses six hardware 7961 IP Phones – that won’t change.

Students outside the US who have experienced higher latency and trouble getting their remote hardware phones to stay registered to CUCM or CUCME may also wish to consider this option and ultimately find this to be a very acceptable alternative.

One final reason that drove us to form this partnership and invest in phones for our racks was that we’ve been asked by so many of our Voice students to provide an option for them to study with 7961 phones as they travel from site to site, and don’t wish to, or in many cases simply cannot, take their hardware phones with them. This gives them the inexpensive option to study using either our phones directly connected to our racks or else their own phones at their main study location connected to our racks via VPN, remotely controlled with this software.

Either way you choose, you now have an equally suitable option for studying for your Voice lab. The only unfortunate part about this is that you know have less of a reason not to be studying for your exam.

Only thing you have to do to qualify for this deeply-discounted version of VoIP Integration’s Remote Phone software is to have 1) purchased any one of INE’s Voice Products, and 2) send me an email expressing your wish to purchase a copy of their discounted software. VoIP Integration will then in turn send you an email to purchase the software at the discounted rate.

Finally – stay tuned for a new announcement approximately one week from now on a brand new Voice product that I personally am very excited to deliver! Something that I have been wanting to do now for a very long time, and something that so far, every live classroom student that I’ve run the concept by, has emphatically communicated back to me how deeply needed this type of instruction is in the Cisco UC marketplace, well beyond the help for the CCIE exam that it will provide. I wish I could give you more info right now – but my flight is landing for my next week’s training class, and I am being told to put my laptop away. So watch this blog early next Friday for the announcement of when this product will (so very soon) launch!

In the (modified) words of Napolean Dynamite:

I hope all of your wildest (studying) dreams come true,

Mark Snow

*IPBlue VT-GO can register to CUCM as a 7961/62 phone, however it does not actually act like one. Many of the most important features tested in the lab, specific to the 7961, don’t work with the IPBlue soft-client – such as Globalization and Mapping Globalized to Localized, G.722 codec negotiation, and many of the softkeys needed.

As a follow up to today’s Cisco announcement, removing the Core Knowledge section from both CCIE Routing & Switching and CCIE Voice exams, INE is offering all our  customers a $99 credit that can be applied to any purchase over $500. Since the Core Knowledge section of the CCIE Exam was announced, we worked hard to deliver you a simulation that would give you the confidence to pass.  We would like to thank all those who used the Core Knowledge Simulator and we were thrilled to hear how it helped you pass.  With the Core Knowledge section being removed from the exam, we would like to use this time to give back to you.  Please, take this $99 credit as our way to say thank-you, and to celebrate this portion of the exam getting removed.  To redeem this credit, simply use promo code INE-OEQ . Remember, INE’s got you covered.  Act now, this offer expires soon.

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• CCIE Routing & Switching Training Solutions
• CCIE Voice Training Solutions
• CCIE Security Training Solutions
• CCIE Service Provider