Jan
16

A number of people had indicated that they thought my CCNA-to-CCIE-in-a-Year program was a bit aggressive for those with both family and job life to consider, so I've revised the plan into an 18-month plan, and also added a downloadable XLS spreadsheet to assist you in your studies. Simply plug in your starting date, and the spreadsheet should keep you on task on a week-by-week basis. Here's the link to my previous blog post where you'll find the updated plan to download, as well as links to all of the products used in the plan: From CCNA Voice to CCIE Voice in One Year – A Detailed Study Plan – REVISED Dec 31, 2012

Jun
13

Since it looks from here at Cisco Live like we won't be updating any CCIE Voice topics anytime soon (looks like roughly 12 months out - so rest easy and keep studying), then this post is most likely one that CCNP's will be focusing on for now - though CCIE's will still certainly benefit from this as well.

Here we will continue our series on Call Control Discovery via Service Advertisement Framework. Today we'll look at the 4th video in that series (video #56) from our current 62-hour CCNP Voice v8 bootcamp.

  • CCD via SAF :: Overview (29m)
  • CCD via SAF :: CUCM Inter-Cluster Call Routing (1h 32m)
  • CCD via SAF :: CUCM Call Routing with PSTN Failover (29m)
  • CCD via SAF :: CUCM Call Routing during SRST Fallback (48m)
  • CCD via SAF :: CUCM to CME Call Routing (54m)
  • CCD via SAF :: Inter-Cluster RSVP via SIP Preconditions (21m)

http://www.youtube.com/watch?v=EY0r9lu18io

Once we finish this series (probably in the next week), I will be starting a brand new long-running series around all of Cisco Unified Contact Center Express and advanced scripting. We'll start with the basics for every one to be able to follow along, but very quickly ramp up to the advanced scripting sessions, so stay tuned for that. It's something a lot of students I've run into here at Cisco Live asked for - and we love to listen to what you all want.

By the way, should the lab be updated anytime sooner - you can already see from this post that INE is way ahead of the development curve for advanced UCM topics, including UCM v9 and TelePresence topics. We've got you covered with the normal extremely in-depth knowledge that you need - not just to "pass a lab", but rather to truly make you an Expert!

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May
17

Continuing on in our series on CCD, today we'll look at the 3rd video (the shortest) in that series (video #55) from our current 62-hour CCNP Voice v8 bootcamp.

  • CCD via SAF :: Overview (29m)
  • CCD via SAF :: CUCM Inter-Cluster Call Routing (1h 32m)
  • CCD via SAF :: CUCM Call Routing with PSTN Failover (29m)
  • CCD via SAF :: CUCM Call Routing during SRST Fallback (48m)
  • CCD via SAF :: CUCM to CME Call Routing (54m)
  • CCD via SAF :: Inter-Cluster RSVP via SIP Preconditions (21m)

http://www.youtube.com/watch?v=sIkbQJUcOUI

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Apr
27

So we continue our series on CCD, something that is being talked about lately - in conjunction with UCM SME (Session Management Edition - think UCM as a Gatekeeper for other UCM clusters) - and something that is bound to be covered on the next version of the CCIE Voice blueprint (something that's received a lot of chatter lately with the advent of UCM v9 beta training this week). Also, a topic that you must know understand well today if you are studying for your CCNP Voice CIPT2 exam.

Today we'll look at the 2nd video in that series (video #54) from our current 62-hour CCNP Voice v8 bootcamp.

  • CCD via SAF :: Overview (29m)
  • CCD via SAF :: CUCM Inter-Cluster Call Routing (1h 32m)
  • CCD via SAF :: CUCM Call Routing with PSTN Failover (29m)
  • CCD via SAF :: CUCM Call Routing during SRST Fallback (48m)
  • CCD via SAF :: CUCM to CME Call Routing (54m)
  • CCD via SAF :: Inter-Cluster RSVP via SIP Preconditions (21m)

http://www.youtube.com/watch?v=ir4bpT72ztU

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Apr
24

Cisco keeps giving this week by allowing any desktop to join - for free - into any existing Telepresence video, and the quality of this video is really, really good. John Chambers recently said that Video is already the #1 traffic driver over Cisco's internal network, and that by 2013 it is expected that over 90% of all traffic over the internet will be based on Video. There's never been a better time to be training for Unified Communications!

Apr
12

Cisco Unified Communications is a truly exciting field to be in, and one that absolutely everyone runs across today at some point during their work in any enterprise environment. We believe it's something that everyone should have access to learn be able to thoroughly understand. We also stand by our commitment to help people understand and attain certification in this highly lucrative field at a very affordable price. In keeping, we recently made our 25 hour CCNA Voice VoD product completely free to stream from our site as well as on YouTube, as well as creating the new Voice Scholarships. So to further this belief -and just as we recently did with our SPv3 Workbook- we have decided to combine the Volume I and Volume II Workbooks for our CCIE Voice track into one workbook, and also to lower the price. We certainly hope this just proves one more way that INE wants to earn your business by helping you to accomplish both your goals and dreams!

Also, we have added new 8-Hour labs with PDF-based solutions.

 

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Apr
03

INE is proud to announce a brand new 62+ hours of the latest v8 CCNP Voice bootcamp. Rerecorded completely from scratch using only UC v8 servers, this in-depth video series covers absolutely everything you need to know for the 5 CCNP Voice exams, with each exam clearly indicated by video title in the playlist. Also, don't forget about our Bookmark and Note-taking feature in our streaming playlists, as this will help you in your studies.


 

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Mar
19

I just finished up 2 weeks of a really great CCNP Voice bootcamp, covering all 5 of the latest 8.0 version exams from Cisco (CVOICE, CIPT1, CIPT2, CAPPS, & TVOICE). All in all we ended up with 62 completely brand-new hours of informative video that we are sure you will be excited to watch when they are posted to our streaming and download sites here in probably just about a week. We went fairly in-depth on most every topic, one of them being MGCP during our TVOICE section of class.

BTW, with this new 62 hours of CCNP Voice video, this brings INE to 320 hours of total CCNA-to-CCIE Voice video-on-demand training. Far, far more than any other vendor. And it is all up-to-date and taught by me, not by subcontracted instructors.

You may recall that in my last post related to MGCP Troubleshooting, we took a basic look at the MGCP commands that a Call-Agent (server) instructs a Gateway (client) to preform - something the RFC refers to as "verbs".

In this post, we are going to take a look at the output of the "debug mgcp packets" command for a single call, and then break down each section of the output into "transactions" (i.e. Command and Response).


So to begin with, here is the complete output a of single call from "debug mgcp packet":


ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x00FF
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA18381
Preferred, Channel 1
Progress Ind i = 0x8583 - Origination address is non-ISDN
Display i = 'Seattle US Phone'
Calling Party Number i = 0x4180, '2065015111'
Plan:ISDN, Type:Subscriber(local)
Called Party Number i = 0xC1, '2065011002'
Plan:ISDN, Type:Subscriber(local)

MGCP Packet received from 177.1.10.12:2427--->
CRCX 256 S0/SU0/DS1-0/3@corphq.voice.ine.com MGCP 0.1
C: D0000000028182cf000000F500000015
X: 3
L: p:20, a:G.729, s:off, t:b8, fxr/fx:t38
M: recvonly
R: D/[0-9ABCD*#]
Q: process,loop
<---

MGCP Packet sent to 177.1.10.12:2427--->
200 256 OK
I: 31

v=0
o=- 49 0 IN IP4 177.1.254.1
s=Cisco SDP 0
c=IN IP4 177.1.254.1
t=0 0
m=audio 18714 RTP/AVP 18 100
a=rtpmap:18 G.729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
<---

ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x80FF
Channel ID i = 0xA98381
Exclusive, Channel 1

ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x80FF
Progress Ind i = 0x8088 - In-band info or appropriate now available

MGCP Packet received from 177.1.10.10:2427--->
RQNT 257 S0/SU0/DS1-0/1@Branch2 MGCP 0.1
X: 1
R: D/[0-9ABCD*#]
S: G/rt
Q: process,loop
<---

MGCP Packet sent to 177.1.10.10:2427--->
200 257 OK
<---

ISDN Se0/0/0:23 Q931: TX -> CONNECT pd = 8 callref = 0x80FF
ISDN Se0/0/0:23 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x00FF

MGCP Packet received from 177.1.10.12:2427--->
MDCX 258 S0/SU0/DS1-0/3@corphq.voice.ine.com MGCP 0.1
C: D0000000028182cf000000F500000015
I: 31
X: 3
L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38
M: sendrecv
R: D/[0-9ABCD*#], FXR/t38
S:
Q: process,loop

v=0
o=- 49 0 IN EPN S0/SU0/DS1-0/3@corphq.voice.ine.com
s=Cisco SDP 0
t=0 0
m=audio 18214 RTP/AVP 0
c=IN IP4 177.1.11.26
a=X-sqn:0
a=X-cap:1 image udptl t38
<---

MGCP Packet sent to 177.1.10.12:2427--->
200 258 OK
<---

ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x00FF
Cause i = 0x8290 - Normal call clearing

MGCP Packet received from 177.1.10.12:2427--->
MDCX 259 S0/SU0/DS1-0/3@corphq.voice.ine.com MGCP 0.1
C: D0000000028182cf000000F500000015
I: 31
X: 3
M: recvonly
R: D/[0-9ABCD*#]
Q: process,loop
<---

MGCP Packet sent to 177.1.10.12:2427--->
200 259 OK
<---

MGCP Packet received from 177.1.10.12:2427--->
DLCX 260 S0/SU0/DS1-0/3@corphq.voice.ine.com MGCP 0.1
C: D0000000028182cf000000F500000015
I: 31
X: 3
S:
<---

MGCP Packet sent to 177.1.10.12:2427--->
250 260 OK
P: PS=974, OS=155840, PR=981, OR=156960, PL=1, JI=3, LA=0
<---

 
 
So now just looking at the first transaction, we see an Inbound call (thanks to the inclusion of the 'debug isdn q931' output) and that the call-agent (UCM) instructs the gateway to "CRCX" or "CreateConnection", and the Gateway responds with very simple "200 OK", while also including the SDP (IETF Session Description Protocol) pertaining to such things as audio codec and DTMF. Notice how after each call-agent instructed verb (command) and gateway response there is a 3 digit number that corresponds. This is the transaction number, and use of it essentially ensures that the response is specific to the command (as there may be many commands and responses being issued on a heavily populated MGCP gateway). We can clearly see in this initial command that the call-agent is instructing the use of the "a:G.729" audio codec on the "L" line, and the gateway is obliging with the SDP response that will use the RTP/AVP (Audio Video Profile) codec number 18, G.729 without Annex B. More on this line reveals other things about the call such as "p" being the packet or sampling size of 20ms, "s" being VAD, "t" being the RFC 2474 ToS/DSCP bits (b8 is hex for 10111000 or DSCP EF), and fax capabilities. You also might note that the gateway was instructed to "M: recvonly" or as far as the "connectionMode" is concerned, that the gateway should only Receive packets, not send them. The "C" line is the global CallID and the "X" line is essentially the local callID. The "R" line is the supported DTMF.


MGCP Packet received from 177.1.10.12:2427--->
CRCX 256 S0/SU0/DS1-0/3@corphq.voice.ine.com MGCP 0.1
C: D0000000028182cf000000F500000015
X: 3
L: p:20, a:G.729, s:off, t:b8, fxr/fx:t38
M: recvonly
R: D/[0-9ABCD*#]
Q: process,loop
<---

MGCP Packet sent to 177.1.10.12:2427--->
200 256 OK
I: 31

v=0
o=- 49 0 IN IP4 177.1.254.1
s=Cisco SDP 0
c=IN IP4 177.1.254.1
t=0 0
m=audio 18714 RTP/AVP 18 100
a=rtpmap:18 G.729/8000
a=fmtp:18 annexb=no
a=rtpmap:100 X-NSE/8000
a=fmtp:100 200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 200-202
a=X-cap: 2 image udptl t38
<---

 
 
The next transaction shows us the ISDN is telling us the phone out on the PSTN is now alerting (ringing) and the MGCP Request Notify tells the gateway with the "S" line to play the "rt" or ringback tone.


ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x80FF
Progress Ind i = 0x8088 - In-band info or appropriate now available

MGCP Packet received from 177.1.10.10:2427--->
RQNT 257 S0/SU0/DS1-0/1@Branch2 MGCP 0.1
X: 1
R: D/[0-9ABCD*#]
S: G/rt
Q: process,loop
<---

MGCP Packet sent to 177.1.10.10:2427--->
200 257 OK
<---

 
 
In this next transaction we see the ISDN output showing that the call is now connecting and the MGCP output that the call-agent us informing the gateway to "MDCX" or "ModifyConnection" again, this time to change its RTP connectionMode to send and receive packets "M: sendrecv". This is where the call was answered, and audio now commences.


ISDN Se0/0/0:23 Q931: TX -> CONNECT pd = 8 callref = 0x80FF
ISDN Se0/0/0:23 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x00FF

MGCP Packet received from 177.1.10.12:2427--->
MDCX 258 S0/SU0/DS1-0/3@corphq.voice.ine.com MGCP 0.1
C: D0000000028182cf000000F500000015
I: 31
X: 3
L: p:20, a:PCMU, s:off, t:b8, fxr/fx:t38
M: sendrecv
R: D/[0-9ABCD*#], FXR/t38
S:
Q: process,loop

v=0
o=- 49 0 IN EPN S0/SU0/DS1-0/3@corphq.voice.ine.com
s=Cisco SDP 0
t=0 0
m=audio 18214 RTP/AVP 0
c=IN IP4 177.1.11.26
a=X-sqn:0
a=X-cap:1 image udptl t38
<---

MGCP Packet sent to 177.1.10.12:2427--->
200 258 OK

 
 
In this next transaction we see from the ISDN output that the call is being disconnected and the call-agent is informing the gateway to "MDCX" or "ModifyConnection" again, this time to change its RTP connectionMode back to receive only. This is the call-agent prepping the gateway that the call is being torn down.


ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x00FF
Cause i = 0x8290 - Normal call clearing

MGCP Packet received from 177.1.10.12:2427--->
MDCX 259 S0/SU0/DS1-0/3@corphq.voice.ine.com MGCP 0.1
C: D0000000028182cf000000F500000015
I: 31
X: 3
M: recvonly
R: D/[0-9ABCD*#]
Q: process,loop
<---

MGCP Packet sent to 177.1.10.12:2427--->
200 259 OK
<---

 
 
In this final (and immediately following the previous) section, we see the call-agent instructing the gateway to "DLCX" or "DeleteConnection". The gateway obliges, and also provides some useful statistics (ConnectionParameters to be specific) about the call, namely "P: PS=974, OS=155840, PR=981, OR=156960, PL=1, JI=3, LA=0". (PS=PacketsSent, OS=OctetsSent, PR=PacketsReceived, OR=OctetsReceived, PL=PacketsLost, JI=Jitter, LA=Latency). BTW, if we had anything such as no audio or a one-way audio issue, we could clearly see that by either both PS & PR or in the case of only one-way audio only PS or PR having a value of 0 (no packets sent and/or received).


MGCP Packet received from 177.1.10.12:2427--->
DLCX 260 S0/SU0/DS1-0/3@corphq.voice.ine.com MGCP 0.1
C: D0000000028182cf000000F500000015
I: 31
X: 3
S:
<---

MGCP Packet sent to 177.1.10.12:2427--->
250 260 OK
P: PS=974, OS=155840, PR=981, OR=156960, PL=1, JI=3, LA=0
<---

 
 
Seeing as we have just covered SDP, it will only make sense to move on to looking as SIP, as it uses SDP for audio information as well.

 

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Mar
02

We are streaming our entire CCNA Voice 5-Day bootcamp free for the month of March (ends 31 March 2012). Just register here and watch.

Also we are streaming Day 1 of our brand new 10-Day CCNP Voice bootcamp, and if you'd like to join for the entire two weeks, we've reduced the price.
Click here to watch live on 5 March or here for more details on the reduced price class.

 

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Feb
18

I recently finished up recording a brand new CCNA Voice v8 online bootcamp, and it is available for both streaming and download. I also spent a number of hours today revising the extremely popular blog post outlining INE's "CCIE Voice in a Year" study plan.

We are still taking enrollment for our brand new CCNP Voice v8 live online bootcamp, so be sure to register if you haven't already.

 

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