The Cisco CIUS

So I have had one of these little tablets for awhile now as it was a spare/demo that we were not using anymore.  I know Cisco discontinued this tablet since it was essentially a piece of crap that an iPad superseded in every way possible.  I had some time on my hands yesterday and wanted to register this thing to our CUCM 11 lab.  I ended up getting it registered and noticed that the Android version on it is dirt old, 2.2.2 old. 

One thing led to another and I realized that thing had Jabber on it but it was called "Chat" or something of the sort without the Jabber name.  Now the CIUS already comes with a built in call control system where you can easily dial with the touch of a button but I wanted to use Jabber since that is how things are done these days when using soft clients and tablets.  I went into the "Chat" client and tried to log in and kept getting a "Invalid username/password".  Ok, not possible, I just reset the password to something I couldn't fat finger and tried a few more times and got the same message.

At this point, I'm getting confused as to how this could be possible.  I put in the username@IP and nada.  It just keeps giving me that error.  After some research online, I found out that Jabber "Chat" program demands a URI of username@domain, in this case user@GTWON.TFE.COM (my special little domain inside the lab).  I hit login and bam, it worked...  Why?  Why did I "HAVE" to put that in there versus the IP which works fine on regular Jabber clients on windows and iPads?

So I was excited, I could make phone calls and chat with my other lab jabber clients and see their presence.  I went back into the call screen and noticed the users were not populating presence status yet the chat client was.  This can't be right, the subscribe CSS is correct and everything is allowed to pass through by default.  I still haven't figured this one out yet.

So leaving that behind, I went back into the "Chat" client and realized that this is all the application could really do, chat.  This tablet was made in an era where the call control and chat were separate still.  I tried to upgrade the application and couldn't.  I figured Google Play store might have been out of date, and it was.  I logged back in after the updates and couldn't find the "Cisco Jabber" application on the store front.  Again, I was like huh?

At this point I was confused but did some more reading and found out that the processor is intel based and not ARM based, not sure if that makes a difference, but still.  So no new Jabber client can be installed on this piece of garbage.  I tried to find out how to update the OS on the CIUS, again, nothing.  4.0 was promised as a free update for the CIUS but it never came to light since the tablet as dumped years ago. 

So there you have it, a tablet that is both outdated and worthless other than for lab use.  I wanted to try the camera out but don't have another video or telepresence device hooked up to the lab just yet.  Other than for looks, this tablet will sit there as a pure lab phone since I can do nothing with it for the time being. 

For those of you wondering how to register the CIUS, there is both a wireless and a 3g/wireless version.  The best way is to let it auto-register but you can also pick the right model based on what you have.  Also, do NOT use the wireless MAC to register, use the "Ethernet" MAC to register as the wireless one is not driving the system to CUCM apparently.  Not sure how that works but I found that out after some research too.  Well I hope this article on antiquated equipment gave you a chuckle.

H.323 Call Preservation

Since getting my NP Voice I figured I would delve deeper into some of the topics.  I feel that it is a good idea to write about this stuff to keep it in my head as well as help others who are studying or just want a deeper understanding on technology.  I know H.323 is old as dirt and is slowly being buried but it is still prevalent in the US since some areas do not have SIP yet.  This makes the T1 still a necessary evil and thus H.323 gateways or MGCP to follow.

MGCP aside, H.323 is where I wanted to talk a bit this go around.  H.323 if you don't already know is dirt old.  It's on revision 2 since last decade and that allows the fast start setup by putting the H.245 inside the H.225 setup.  Granted when this is done, it's purely for call setup since all other aspects still need negotiated but at least the codec and RTP stream can get going faster without waiting for a bunch of replies. 

H.225 is your basic call setup.  It doesn't negotiate media in itself as it is pure layer 2.  H.245 is you media negotiation that happens and is where codecs and all other capabilities are agreed upon.  By putting the H.245 inside the H.225 you get one big message that contains everything needed to setup the call quickly, hence the name fast start.  If the other side doesn't respond within the next three messages then slow-start is assumed and the call can still proceed.  Both sides have to support fast-start in order for it to occur.  In CUCM, this doesn't happen natively, or at least it didn't.  I haven't checked in version 10 and 11 as everything for us is SIP or MGCP. 

I had a pause for a moment and decided to take a look at the newer version of CUCM out there to see if setting up a H.323 gateway still required a check box.



CUCM as of version 11 still requires you to check the inbound and outbound H.323 fast start.  Crazy really, this is a old technology and Cisco must still think that even today there are devices out there that use slow-start.  I would have thought that the box would be pre-checked today.  Now I know of course that there are slow-starts still out there but the majority of everything is fast-start from what I have experienced these days.  Not a big deal but now you know where to setup the CUCM side of things for fast-start.

So back to the call preservation since I kind of got off on a side tangent there.  I felt that it needed explained in order to fully understand what happens.  Basically, when an H.323 gateway gets a call setup, either from CUCM or elsewhere a logical channel is opened on the H.225 side.  If a WAN failure happens and CUCM dumps, the call can still go on if preservation is configured.  The reason you want to configure preservation is if CUCM comes back online during the call.  Since CUCM lost the TCP connection, it clears the socket and see that the phone is still connected.  CUCM at this point just ends the call because it thought the call was either over or something else happened.  If you enter the "call preserve" command, the gateway will ignore the call teardown request and keep the call active until either the media goes inactive or the call is finished.

Here is Cisco's SRND on the entire process:

 "Note that, after the media is preserved, the call is torn down later when either one of the parties hangs up or media inactivity is detected. In cases where there is a machine-generated media stream, such as music streaming from a media server, the media inactivity detection will not work and then the call might hang. Cisco Unified CM addresses such conditions by indicating to the gateway that such calls should not be preserved, but third-party devices or the Cisco Unified Border Element would not do this.

Flapping is defined for this feature as the repeated and temporary loss of IP connectivity, which can be caused by WAN or LAN failures. H.323 calls between a Cisco IOS gateway and Cisco Unified CM may be torn down when flapping occurs. When Unified CM detects that the TCP connection is lost, it clears the call and closes the TCP sockets used for the call by sending a TCP FIN, without sending an H.225.0 Release Complete or H.245 End Session message. This is called quiet clearing. The TCP FIN sent from Unified CM could reach the gateway if the network comes up for a short duration, and the gateway will tear down the call. Even if the TCP FIN does not reach the gateway, the TCP keepalives sent from the gateway could reach Unified CM when the network comes up. Unified CM will send TCP RST messages in response to the keepalives because it has already closed the TCP connection. The gateway will tear down H.323 calls if it receives the RST message.

Configuration of H.323 call preservation enhancements for WAN link failures involves configuring the call preserve command. If you are using Cisco Unified CM, you must enable the Allow Peer to Preserve H.323 Calls parameter from the Service Parameters window.

The call preserve command causes the gateway to ignore socket closure or socket errors on H.225.0 or H.245 connections for active calls, thus allowing the socket to be closed without tearing down calls using those connections."

So as you can see, the gateway does as told by CUCM since CUCM is there for call setup and teardown.  This is actually an ingenious design that I had not fully understood until I read the SRND.  Even with NP Voice, knowing the material isn't enough, you need to dig deeper and understand the inner workings of everything!

It's official I'm CCNP Voice certified

It's been a rough year of study.  Granted I didn't study hard every day since I do this for a living but it's still been some dedication to get to where I am now.  After ten years in the business I'm finally certified at the NP Voice level.  For all of you aspiring engineers, you have it a little bit better as you won't need CCNA Data anymore to obtain a CCNA Collaboration (the new voice track).  that comes at the expense of you need to know more stuff and missing out on the data side is going to hinder you which is why I would still recommend having a CCNA so you at least know what a gateway and switch are. 

With all of that aside, I feel that the exams were mostly difficult but not impossible.  CIPT2 was the hardest one to pass despite me failing CIPT1.  I think it's more of where you are in your career and what you mess with on a daily basis that defines what topics are hard.  For instance, SAF and CCD are hardly ever used, in fact SAF is being phased out as it is no longer developed from the discussions that I have read.  ILS is going to be taking its place however, so don't think you are skipping by on something.

I think the biggest topics you really need to understand are the dial-plans and the media resources between CIPT1, CIPT2, and TVOICE.  If you arn't familiar with these aspects you will surely fail as they are the meat and potatoes of the Call Manager.  CVOICE and the gateways are equally important but with the new tracks out, they are starting to take a backseat as the focus is on GUI driven objects and understand how they work.  My biggest issue with this is, new engineers won't know how to configure a gateway for SIP, H.323, or MGCP.  I know T1s are going the way of the dinosaur but that doesn't mean you can skip on that information, especially if you are in a MGCP environment.

TVOICE this week

Well, I am here at last.  The final exam and I will be taking it this week then I am CCNP Voice certified.  Once that is complete and the new 300 exams come out for CIPTV2, I will be getting that exam knocked out as well to convert the NP Voice over to NP Collaboration.  It's been ten years since I started in the industry and it's strange that I just now decided to get these exams done.  It's been a long road and while not the end, there is still a nice little vacation spot for me knowing I completed a milestone.  I will continue to post relevant data here on voice as that is my career but I will also be veering off into other adventures too.  I may go CCNA Wireless then obtain my CCNP Wireless as that is another hot industry. 

Things have been slow lately.  I have had time to study (which is good), but also come across some relatively different things while working on internal company stuff for our telepresence and voice upgrades.  I think the biggest thing is licensing and how much of a pain in the butt it is to deal with at times.  We are in the process of getting new equipment and licenses and the TMS is one of those things that I will be rebuilding from scratch.  Yes, I could technically export the SQL database, in fact I already have but decided I would rather build it from the ground up since this server is something I need to deploy to customers.

I know TMS and understand it fairly well, but from a design perspective and installing/configuring from scratch, well not so much.  Our biggest issue right now is the TMS is on a physical Windows Server 2003 box.  That box is EOS/EOL and has been for a long time.  The new one will be virtualized and thus we just need to convert the licenses from hardware to software and then bring it up.  I don't think it will be very hard but I may record the entire process for those that would like to see it from beginning to end. 

CIPT2 Cleared

     Well I took my CIPT2 exam yesterday and cleared it, just barely.  That exam was much harder than CIPT1 and covered less material.  I think the thing about CIPT2 is SAF and CCD is big.  Even though I have never used and will likely never use SAF and CCD, especially since it is no longer being developed, it is on the test.  Globalized dial plans are fairly simple and straight forward and understanding TEHO should be second nature to anyone in the VoIP business.

With all that being said here is a list of things you really need to be on the top of your game on.

1. CCD/SAF configuration and theory.
2. Multisite dial-plans and how to normalize/globalize the numbers inbound and outbound.
3. Media Resources played a small role for me on my exam.
4. Dial-plans...did I say dial-plans?  Please make sure you understand every aspect of a dial-plan.
5. CAC both gatekeeper based and local to CUCM as well as the bandwidth for common codecs to include your layer 2 and layer 3 portions.
6. Mobility and how logging in one location can be different than another based on criteria such as location and subnet.  This is a key portion to understand.

     I know most of that is the blueprint on Cisco's site but it really gets detailed.  I kept hitting diagram after diagram on scenarios which took some considerable time.  The problem is, the screen is not big and you can't fit everything on one page which makes you click between the different diagrams and configurations.  This can get a bit irritating since you have to keep bouncing around so the more you know and are comfortable with the material the better.

     That being said, I feel that CIPT2 would be great for a design engineer in a multi-site environment or someone supporting that style deployment.  I have almost never dealt with intercluster trunks as everything is centralized to keep things simple.  Even when I worked at Cisco and we had over 3000 telepresence endpoints alone across the globe, we were centralized.  Sure we might have had a CUCM elsewhere within the cluster but the bandwidth was so highspeed that it was under the 80ms required for database replication.

     CBT Nuggets did a pretty good job at preparing me for the small things I didn't already know.  Some of the stuff was incredibly easy since I do this for a living while the other stuff like CCD/SAF with the appropriate trunking was a bit more difficult.

BAT and 64-bit Excel

Lately, I've been seeing some people have issues when trying to use BAT on CUCM.  When you open the excel file you have macros and all that jazz needed to start whipping up a spreadsheet for import.  Normally this works without issue but now that systems are going pure 64-bit, there is really no need to make 32-bit applications.

So what is the problem you might ask?  The biggest issue is that Cisco has not developed their BAT file to work on 64-bit Excel.  This can cause headaches as large phone deployments suck without BAT.  Fear not however, there is a work around that a clever gentlemen on another website had posted to fix the issue.  The only problem with this solution is that you have to keep making the change to the visual basic file to get everything to work each time you open a BAT sheet.

Go here: 64-bit Excel Fix

Again, this isn't my work but a solution I found and posted on the Cisco support forums when the issue came up.  It makes life easy because hand jamming 283 phones sucks and you will be in a hand brace the next day if you go that route.  Oh and by the way, I passed my CIPT1 exam.  Two left and I am preparing to take both of them back to back.

CIPT1 again

Tomorrow marks my second attempt at CIPT1.  If I pass, I'll go schedule CIPT2 since I've at this point combined study on both.  Assuming I pass, that would leave two exams left before the October deadline, and none of the rest are going to be an issue.  Ideally, I'm CCNP Voice by the end of June or beginning of July.  Then I can take the CIPTV2 exam which is the new 10.6(10.5(2)) track and convert to a CCNP Collab cert.  Here goes to hoping I pass and get one exam closer to my goal so I can move onto other things.