Author Archives: Andy Vassar
Author Archives: Andy Vassar
The jabber-config.xml file is an essential piece of configuration for the Jabber client. Sure, the client has the ability to operate just fine without this file. Video calling, deskphone control, instant messaging, etc. all work flawlessly. However, if you need to add any additional options, policies, or directory integrations, the jabber-config.xml file becomes necessary. Within the realm of the CCIE Collaboration certification, we are specifically concerned about two different configurations: UDS Directory Integration and SIP URI Dialing.
User Data Service (UDS) simply put, is the name for the End User database within Cisco Unified Communications Manager (CUCM). It contains all relevant information about that user, as would any other directory. UDS, however, is not enabled by default on the Jabber client. In fact, Jabber is geared towards integration with an LDAP source “out of the box”. This means that we must instruct the Jabber client to use UDS if we would like to be able to search the CUCM database to communicate with other users. Since this will have to be done by using the jabber-config.xml file, we must first determine how to create it. Thankfully, the Cisco documentation does not disappoint in this regard. From the Continue reading
The implementation of redundancy in any technology is of paramount importance, whether you’re studying to achieve a CCIE certification or designing a network for a client. So it goes without saying that this is a concept with which you should become intimately familiar.
In this blog, we’ll turn our focus specifically to redundancy in IOS
dial-peers. Of course,
dial-peers come in two different flavors: POTS and VoIP. POTS
dial-peers deal exclusively with PSTN connectivity while VoIP dial-peers can be used for several purposes, as long as the communication takes place over IP.
Let’s take the example of a call routed inbound from the PSTN, destined toward the HQ CUCM cluster using the H.323 protocol. The configuration on the gateway appears as shown below.
As you can see, we are accepting calls inbound from the PSTN using
dial-peer voice 1 pots and translating the incoming called number to a 4-digit DN. From there, we have two separate
dial-peers with the ability to send the call to the HQ CUCM cluster. As you know, the
dial-peer with the lowest preference (default 0) is chosen as the first routing option. If for some reason, that option is unavailable, the next possible
dial-peer Continue reading
Finally, it’s the blog you’ve all been waiting for! Yes, that’s right folks; the time has come to discuss the benefits of Real-Time Monitoring Tool (RTMT) in CUCM and CUC. All right, I know it’s not the most exciting subject to discuss all the topics on the CCIE Collaboration lab blueprint, but it can help you perform troubleshooting tasks in a very efficient manner. The goal for this blog is to point out a couple useful features of RTMT to give you a nice boost when tackling different lab topics.
For those that are not familiar with RTMT, it can be used to pull traces (log files) for troubleshooting in all Cisco UC servers, monitor real-time platform statistics, check syslog messages, and display a host of “Performance” parameters that can assist the engineer in gathering system information. While those are all great features worthy of our attention, I’d like to focus specifically on a new RTMT feature available in CUCM 9.x called “Session Trace Log View.” This feature is an excellent troubleshooting tool, especially when used with SIP. Essentially what this does for us is organize the traces in such a way as to provide a cohesive view Continue reading
Just as was recently announced for the CCNA Voice and CCNA Video, the CCNP Voice has now gone the way of the dinosaur. It’s replacement? The highly-anticipated CCNP Collaboration certification, which of course will now be adding video to its laundry list of topics.
To attain the CCNP Collaboration certification, you must now pass four different exams. This is actually a nice bit of news, since we had to pass five separate exams to achieve the CCNP Voice certification. Don’t get too excited though; Cisco is sure to have packed each of these four exams full of enough content to account for the loss! On that note, Cisco has not yet released the exact details regarding the topics for each exam. So we must wait a little while to let the full picture develop.
The first of the four exams is called “Implementing Cisco IP Telephony and Video, Part 1” and corresponds to exam number 300-070 CIPTV1. This exam will mostly likely introduce the majority of the necessary Cisco IPT concepts while laying a solid foundation to build upon. The second is called “Implementing Cisco IP Telephony and Video, Part 2” which corresponds to exam number 300-075 CIPTV2. For this Continue reading
Well folks, it has finally been announced! Cisco has retired the CCNA Voice and CCNA Video certifications in favor of a new, all-encompassing CCNA Collaboration certification. It will be comprised of two separate exams—one with a focus on Unified Communications solutions and one emphasizing the implementation and troubleshooting of video infrastructures.
The first exam is called “Implementing Cisco Collaboration Devices” and corresponds to exam number 210-060 CICD. Topics to note within this exam will be call signaling and media flows, VoIP quality implications, user account creation and modification, calling privileges, IM and Presence, RTMT and CDR/CMR-based reporting, and typical end user support scenarios. The focus of this exam is now on becoming a well-rounded engineer, with knowledge in more than just CUCM. For more information, visit this link here.
The second exam is called “Implementing Cisco Video Network Devices” and corresponds to exam number 210-065 CIVND. Topics on this exam that should carry great importance will be streaming video, media convergence, desktop and immersive systems, troubleshooting methodologies, media quality, and multi-point control units. With the focus here being solely on video and its integration with voice networks, candidates for the exam will gain a breadth of knowledge during the studying Continue reading
The time has come, CCIE Collaboration hopefuls, to focus my blog on Quality of Service (QoS). I know, it’s everyone’s favorite subject, right? Well, you don’t have to like it; you just have to know it!
I would specifically like to focus on WAN QoS policies as they are going to be an essential piece of the lab blueprint to understand. Typically, the goal on a WAN interface is to queue traffic in such a way as to prioritize certain types of traffic over other types of traffic. Voice traffic will usually be placed in some type of expedited or prioritized queue while other types of traffic (video, signaling, web, etc.) will use other queues to provide minimum bandwidth guarantees. Policies such as this will utilize the Modular QoS Command Line Interface (MQC) for implementation.
To begin, let’s use our three-site topology (HQ, SB, and SC) to provide a backdrop for this example. The HQ site (R1) has a Frame Relay connection to both the SB (R2) and SC (R3) sites through the same physical Serial interface, which has a total of 1.544 Mbps of bandwidth available. Assume that both R2 and R3 have connections to R1 using Continue reading
In the CCIE Collaboration lab, understanding dial-peers is extremely important. Lack of knowledge in this area can yield devastating results in your lab score report since they can be found in so many different sections of the exam. We must be thoroughly prepared to tackle every aspect of this technology should we be presented with it at some point.
I recently got a great question in our forums about digit manipulation within POTS dial-peers and how they interact with translation rules and profiles. I figured that since this is such an important topic, my answer to his question bears repeating so it can reach a wider audience.
Let’s begin with the simple example of dialing the number “123” from a CUCME phone. Of course, the POTS dial-peer must be created to support the desired behavior.
When this pattern is selected, all digits will be stripped automatically since they are explicitly defined. This is due to the “automatic POTS dial-peer digit strip” feature in IOS. See below for the ISDN Q.931 debug output (no Called Party Number).
Since we are not currently sending a Called Party Number, we’ll need some way to add the digits back to the string to Continue reading