Dane DeValcourt

Typically when deploying VoIP in the enterprise your going to be dealing with G.711 and/or G.729 codecs.  Both of these codecs sample at 10ms intervals.  The default is to usually send packets every 20ms, however in some situations one might see this set to every 30ms.

If your doing packets every 20ms this creates about 50 packets per second.  If your set to 30ms you will be creating 33.3 packets per second.

Lets focus on G.711 for a second.  G.711 has a sample size of 80 bytes.  So if we are set to use the default voice payload freq. of 20ms and we know that samples are taken every 10ms then our G.711 voice payload will be two samples of 80bytes = 160bytes of voice in every packet.

Thats only part of the equation though, you can’t forget about headers!

You will need to know what headers your going to be dealing with such as Layer 2, Layer 3 or both combined? And possibly WAN headers also.

In a typical ethernet packet your looking at about 58 bytes of header overhead.  18 bytes of ethernet (Layer 2) and 40 bytes of IP, UDP and RTP (Layer 3).  So with your 160 bytes of voice and 58 bytes of header overhead your dealing with 218 byte packets.

In this example we are using the default 20ms voice payload freq. which we said corresponds to about 50 packets per second right?  So we take our total packet size of 218bytes x 8bits x 50 packets per seconds = 87.2 Kb/s for every voice call.

Cisco has refreshed / updated the CCIE voice lab.  Listed below is the new equipment and software it will cover.

 

Lab Equipment:

• Cisco MCS-7845 Media Convergence Servers
• Cisco 3825 Series Integrated Services Routers (ISR)
• Cisco 2821 Series Integrated Services Routers (ISR)
• ISR Modules and Interface Cards

             VWIC2-1MFT-T1/E1 
             PVDM2 
             HWIC-4ESW-POE 
             NME-CUE

• Cisco Catalyst 3750 Series Switches
• IP Phones and Soft Clients

Software Versions

Any major software release which has been generally available for six months is eligible for testing in the CCIE Voice Lab Exam. 

• Cisco Unified Communications Manager 7.0
• Cisco Unified Communications Manager Express 7.0
• Cisco Unified Contact Center Express 7.0
• Cisco Unified Presence 7.0
• Cisco Unity Connection 7.0
• All routers use IOS version 12.4T Train.
• Cisco Catalyst 3750 Series Switches uses 12.2 Main Train

Network Interfaces

• Fast Ethernet
• Frame Relay

Telephony Interfaces

• T1
• E1

 

Source:  Cisco Learning Network

Some of the major changes are:

1) Remove analog devices (such as VG248, ATA)
2) Remove CatOS (Catalyst 65xx)
3) Replace CCM with CUCM 7 (Linux Appliance)
4) Replace Unity with Unity Connection 7 (Linux Appliance)
5) Add CUPS 7 (Linux Appliance)
6) Add SIP phones

Basic topology:

 

Source:  http://htluo.blogspot.com

So I was doing some upgrades this past week adding some CMM-ACT adapters to existing Cisco CMM modules that had active PRI’s on the modules.  I had one CMM module per Catalyst 6509. 

All my PRI’s are split between these two 6509′s, and the various PRI’s belong to trunk groups (hope I got the term right) from the telco.  So, basically I can unplug any given PRI from one 6509 and the other PRI on another 6509 will still be able to take calls.

With that said I can technically take down one CMM module from one 6509 and still make and recieve calls.

So that was the plan, take down a module, install the CMM-ACT put the module back in and check make sure everything is working and repeat on the other module.

The catch of course is that if their were any calls on a PRI attached to the particular CMM well they would go down when I unplugged the PRI.  So I started researching a way to ‘busy out’ my PRI’s on the CMM I was going to work on so that it would preserve existing calls but not accept any new calls forcing them to move to the other PRI’s on the other CMM.  Doing something like this should allow me to get all calls off the PRI’s so that I can do maintenance and not incur any downtime.

I didn’t get my answers in time for my maintenance.  But I did eventually find various sources of info related to this that I look forward to try at a later time.

In CallManager it seems there is a service parameter that would help accomplish what I was trying to do.  The service parameter is:  ”Change B-Channel Maintenance Status 1 – 5″
Apparently this should allow you to take the B-channels out of service for up to five MGCP gateways without disrupting calls.

Source: Cisco CallManager Best Practices
Google Book Search Preview:  http://tinyurl.com/c5lop7

Catch is the PRI’s have to be pre-configured with “unchecking the inhibit restarts at PRI intialization” and “check enable status poll” then restart the gateway.

This Cisco doc seems to explain it fairly well.

Busy-Out ISDN B-Channels in Cisco CallManager Configuration Example
Source Cisco Website:  http://tinyurl.com/2675fo

 

Someone else also suggested, actually they asked, if the “mgcp block-newcalls” gateway config command could possibly be used in this situation.  I don’t know myself but its worth trying one day.  Still waiting to see if anyone comments on his question in which case I will add that info here as well.

Came across a source for some free “virtual lectures” today related to the various CCIE exams.  Only briefly tried them out and I think they have potential so I might try to go through them all soon.

• CCIE Related Topics: All Tracks
  • The Psychology of the CCIE lab and how to plan an attack.
• CCIE R&S Related Topics
  • Basic Multicast Design/Operations
  • Frame Relay
  • OSPF 
  • Layer 2 Tunneling Techniques 
  • Spanning-Tree
  • Troubleshooting on the CCIE Lab
  • Multicast- Anycast RP
  • Binary Math: Subnetting / access-lists
• CCIE Voice Related Topics
  • SRST
  • Unity
    
  • H323 Gatekeeper Basics
  • IPMA
  • WAN QoS
  • Digit Manipulation on CallManager 4.1(3) & CME 3.3
  • Troubleshooting in the CCIE Lab
  • Advanced Call Routing
  • Unity, Unity Express, & VPIM
  • Basic-ACD: Part 1, Part 2
  • Campus QOS
• CCIE Security Related Topics
  • DMVPN
  • Binary Math: Subnetting / access-lists
• CCIE Service Provider Related Topics
  • ATM Operations and Configuration

I really enjoy reading NetworkWorld.com due to both my career and simply out of my love of all things network related. I am a telecommunications junkie I suppose.  Anyway.. Dennis Hartmann has being doing a fair amount of posts on QoS lately that I have really enjoyed.

http://www.networkworld.com/community/blog/6202

QoS is so very important in the way networks function and it’s something I have yet to be able to master myself.  I constantly relearn some things when it comes to QoS to be honest, I have a love / hate relationship with it.  Use it, but its kinda been one of those things you set it up, fight to get it to do exactly what you want and then forget about it until a later date. 

I need to master it soon though.  Its a big part of the CCIE Voice.  Yeah!

20% of standard G.711 packet is used by IP, UDP and RTP headers.  G.711 payload consists of 160 bytes and IP, UDP and RTP headers total roughly 40 bytes.

QSIG is a peer-to-peer signaling system used in corporate voice networking.  Internationally, QSIG is known as Private Signaling System No. 1 (PSS1).  This open standard is based on the ITU-T Q.9XX series of recommendations for basic service and supplementary services.  Therefore, as well as providing inter-PBX communications, QSIG is compatible with public and private ISDN.

QSIG also has one important mechanism known as Generic Functional Procedures (QSIG GF).  This mechanism provides a standard method for transporting features transparently across a network.

Source:  Voice over IP Fundamentals (Second Edition) – Cisco Press

Wiki – QSIG

Light travels through a vacuum at a speed of 186,000 miles per second and electrons travel through copper or fiber at approximately 125,000 miles per second.  A fiber network stretching halfway around the world (13,000 miles) induces a one-way delay of about 70 ms.  Although this delay is almost imperceptible to the human ear, propagation delays in conjunction with handling delays can cause noticeable speech degradation.

Source:  Voice over IP Fundamentals (Second Edition) – Cisco Press

Wiki – Propagation Delay