CME Music On Hold

Ever since I've read past the CCNA:Voice section regarding Music on Hold (MOH) I've been trying to get it working ever since. The problem I ran into is that I couldn't get MOH to work when I placed a call from one IP Phone to another in my internal network, just dead silence. After some research, apparently this a known issue depending on how you configure MOH. In order for this to work between IP Phones in your internal network, you're supposed to use multicast rather than the default method. Even with multicast configured and every option I could think of, I still couldn't get this working.

Well after completely forgetting about MOH and eventually setting up my remote network weeks later, I was reviewing a few CBT's when it hit me that MOH should work when putting external callers on hold. I placed a call to the analog phone on the remote network and then put them on hold, sure enough it worked! I will admit though that the quality wasn't great because I was using the G.729 codec across my WAN and I'm betting that this analog phone is at least 15 years old!

The Smart Business Communications System

I finally knocked out the book for the CCNA:Voice this evening, the last two chapters were based on the Smart Business Communications System (SBCS). To be honest both chapters felt like one huge sales pitch but there was some useful information that I'm sure will be on the exam. As far as I understand, the SBCS suite is based around the Cisco UC500 product family and how they interconnect between each other.

The main player within the SBCS is the UC520 ISR which is pretty slick within itself, especially if you're a non-Cisco person. It enables you to have a router, firewall, POE switch, PBX, Wireless, and many other features within one small device. The UC520 itself seems fairly straightforward, especially with the web based GUI tool called Cisco Configuration Assistant (CCA). Now that I finish these last few chapters it's time to review over the next few weeks and then exam time!

Cisco Unity Express Hardware Limits

It's important to know that you're limited with voice mail size limits based on the Cisco Unity Express hardware itself. Below I have listed the four different hardware size's and their limitations that I'm assuming will be on the exam:

• Cisco Unity Express Advanced Integration Module (AIM-CUE)

50 mailboxes
14 hours of storage
4-6 ports for voice sessions depending on the router model

• Cisco Unity Express Network Module (NM-CUE)

100 mailboxes
100 hours of storage
8 ports for voice sessions

• Cisco Unity Express Network Module with Enhanced Capability (NM-CUE-EC)

250 mailboxes
300 hours of storage
16 ports for voice sessions

• Cisco Unity Express Enhanced Network Module (NME-CUE)

250 mailboxes
300 hours of storage
24 ports for voice sessions

Complete Voice Network

Today I wrapped up the last bit of configuration I needed to do to complete my CCNA:Voice lab setup (besides Unity Express). I am now able to place calls not only internally but also to my remote site via a PSTN connection. I have four more chapters to go and then it's time to review and prepare for the test. I want to take and pass the exam sooner than later because the certification is being completely revamped and I only have until February 28th to take to current version. I already feel pretty confident, especially after I was able to finally dig into dial-peers and the many ways you can manipulate them.

VoIP Trunking Protocols

With most business that integrate VoIP into their network, they not use VoIP technology within their own LAN but also across WAN's to remote sites, other businesses, ISP's, etc. There are a few popular voice trunking protocols to pick from and I'm starting to learn the history and details behind each. There are four voice gateway protocols that I hear and see configured quite a bit in the voice networks I've seen so far.

1. H.323: This is a pretty mature and older voice protocol that was created primarily for connecting to other networks using the ISDN connection. This doesn't surprise me as I see this used mainly on PSTN connections to ISP's. While the protocol is pretty stable, there are a lot of features newer protocols handle better along with being easier to manage. When I run the debug command debug isdn q931 on a gateway, the information is usually pretty cryptic and hard to undertstand.

2. MGCP: I see this protocol used all the time in the Cisco VoIP networks, apparently Cisco is really the only vendor that uses this protocol even though it's an open standard. MGCP is a centralized based configuration.

3. SCCP: Again this is only on Cisco VoIP networks because this is Cisco's proprietary protocol. This protocol is used specifically for communication with Cisco IP Phones and other Cisco endpoints such as ATA's.

4. SIP: This is the new kid on the block and it's still being developed constantly. SIP is basically a lighter and more feature rich version of H.323. Except that it is based on VoIP instead of ISDN, it uses a lot of the same standards that HTTP uses which makes it a lot simpler to develop. I haven't touched SIP yet but I've heard a LOT about it. There's actually a forum that is very involved with this new protocol, check it out at www.sipforum.org

First Hands on With QoS

As I dive deeper into the VoIP world, I'm starting to learn just how important QoS (Quality of Service) is in a network. QoS is a complicated topic and for good reason, not only because the configuration can be quite involved, but also their can be a lot of corporate politics involved. QoS comes into play when a network is congested, instead of having a router just drop random packets that it can't route you can prioritize which specific packets should be sent from most important to least important. With the least important being sent to the "bit bucket (gone forever)" most likely. With this said RTP voice traffic is very time sensitive, when there is latency it upsets the way it is sent to the receiving end and in turn usually upsets the end user as it can cause echoing, dropped calls, and a plethora of other issues.

You should prioritize voice and video traffic before other traffic in most environments...well unless the CEO steps in and demands that his e-mail and internet traffic is more important than any other traffic! There are essentially 3 steps to configuring QoS:

1. Identify and match packets with the class-map command
2. Prioritize traffic with the policy-map command
3. Assign the QoS policy to the impacted interface with the service-policy command

You can see a sample configuration in the picture above that I was messing around with in GNS3.

The Many CME Phone Features

Over the last few weeks I've been reading and toying with the many different phone feature settings that you can configure through CME. While there are still many options I haven't covered quite yet such as Intercom, Music On Hold, and Paging, I've learned a lot about what you can do using the ephone commands on the CME gateway. By far the most confusing part for me has been the configuration of hunt groups and how to route and stop the calls within hunt groups and to other hunt groups. It's a lot easier to understand when you're looking at a full Call Manager GUI rather than from ephone-dn commands!