Avaya Static On Calls

Analogy of the Troubleshooter

Static_On_Calls_By_Wellington

Avaya Static On Calls

Analogy of the Troubleshooter

In this post “Avaya Static On Calls”, see how you can optimize your troubleshooting skills by following these easy steps which I call the “Analogy of the troubleshooter”. Learn how to prioritize and eliminate issues related to static on calls by understanding the technology that the customer has, and following some of the troubleshooting techniques mentioned here.

Not so long ago I got the chance to help one of our customers who were occupying different floors connected via fiber links and Cisco switches; I reached out to the IT Administrators to help me find the cause of the “static on calls”. During the troubleshooting phase we found that the problem was related a misconfigured switch, and the problem was fixed after matching the CDP settings to all the switches.

Here I’m providing you the 3 steps which helped me figured out the problem and fix it.

  • 1.- Prioritize and eliminate
  • 2.- Understand the technology
  • 3.- Find, test, and Fix

1.-Prioritize and eliminate

When arriving to the job site, it is your responsibility to sit-down with the customer and investigate which problems that they are experiencing. Replicate the issues as they happen if possible.

In most cases you will collect different types of issues as you go through your findings. All of which have the same level of urgency to the customer, because of this it is a good idea to prioritize, eliminate by asking, and compering them. One way to demonstrate issues to your customer is by showing which ones carry the biggest priority, by doing this exercise they should be able to decide which one should be resolved first.

2.- Understand the technology

The Avaya Product line has multiple ways to integrate services offered by the Provider. As a service technician it is your responsibility to know the technology implemented to better serve your customer(s). These are some of of them=

Loop-Start Trunks – This copper trunk connects from the main frame provided by your Telephone Company using CAT3 wire. A pair of wires carries a positive and negative electric charge producing dial-tone also known as TIP/RING.

There are great limitations when using these type of trunks, from the amount of conference calls allowed, EC500/Twinning, to manipulating outbound caller ID information.

In order to connect CO trunks into the Avaya gateway/PBX you need a trunk card or module. The MM711, TN747, and the IP Office Analog Trunk 16 are some of the ones available.

H.323 Trunks – Also known as IP Trunks, where a connection exists between two call servers or gatekeepers. Provisioned through the customer’s network or directly to service provider’s equipment, supporting voice and video . The main purpose of these trunks is to network systems together.

C-LANs, MEDPROs, and IP Media Servers are needed with the Avaya Aura, and IP Office uses VCM Modules or Combo cards.

SIP Trunks – These type of trunks are brought to the customer premises via a Gateway installed by the ISP providing telephony services through a packet switched network, hence the name of ITSP (Internet Telephony Service Provider). An Ethernet connection is required to connect to your Avaya PBX. SIP is responsible for controlling call signaling, video, instant messaging, and presence, utilizing the same VoIP resources to covert the signals from TDM/DIG/IP.

ISDN T1-PRI Trunks – 23 B-Channels are use for voice transmission, and one D-Channel for signaling, carrying caller ID, and other information. The TELCO delivers a Smart-Jack for you to connect the T1-Card using a CAT5 cable. It is recommended to use a CSU to sync the T1 with the service provider’s clock-source.

For Avaya Aura MM710 and the TN-464 are the most used, or the ISDN-PRI Daughter card for the IP Office.

3.- Find, test, and Fix

I have documented some troubleshooting techniques to help you find the most common “static on calls” related issues.

Loop-start Trunks Troubleshooting Steps

Start by connecting your test-set (butt-set) directly to the TELCO demarcation point and check for dial-tone. If you happen to find issues there, it is time to contact your telephone company and arrange for further troubleshooting. But before you get to this point make sure that there is nothing connected from the block to your system, the only wires connected to he demarc should be the ones coming from the MPOE.

Common Loop-start Trunk Problems

Loss – It is the user’s awareness of how high or low the telephone connection is. Measured in decibels, loss can be detected when cables are installed too far from the local loop (service provider) causing attenuation issues.

What to do – Have the LEC come out and test and replace the wire.

Crosstalk – it is when a user hears another user’s conversation causing a coupling of voice signals between the voice paths. Personally I have seen this when near radio stations with old wire laying on top of each-other.

What to do – Follow the same procedures already mentioned above. You can also provide new cable drops and replace all connections if necessary.

Impedance Mismatch and DB-Levels – IP Office has a feature where you can run a test per analog loop-start trunk or card to match the impedance value read through the copper wire. Another factor to consider is the DB Levels in the trunk. Contact your service provider and match those settings, or connect a Voltmeter to get the readings off the block.

Common H.323 Trunk Problems and Troubleshooting Steps

Start by checking the Avaya Aura, or IP Office system configurations. then move on to the physical components. Check the cable connections. Run a packet tracer such as Wireshark which allows you to see VoIP traffic, assessing the customer’s network. While running the trace you are looking for high volume of traffic with the same protocol.

Local Echo – based on the “Avaya’s Analog Trunk Measurements document”, Local echo is produced by signal reflections at impedance discontinuities, which occur at the 2-wire to 4-wire interfaces on the analog port circuits, and where impedance mismatches occur on the two wire loops.

Talker echo – When a caller can hear himself while talking, generated by the delayed of the talker’s words. and Listener echo, which is produced when the signal passes through two reflection points and is added to the transmitted speech causing feedback and the voice to sound hollow.

You need to start by troubleshooting the voice echo cancellers first before troubleshooting the customer LAN.  You should be aware that  the Voice-echo-cancellers have to learn about echo first, before it can be detected.

MPs and VCMs – For the Media Processors (MPs) you can check the health and performance via the console shell emulation  and the ASA application.

ASA Commands to help you troubleshoot echo related issues

list measurements ip dsp-resource summary today-peak
list measurements ip dsp-resource summary yesterday-peak
status media-processor all
status media-processor board boar-location

Media Gateway Shell Commands

show voip-parameters
test voip-dsp
show echo-cancellation

Auto Gain Controls – These can help to eliminate the echo on Handsets, Headsets, and Speakers by adjusting the volume. The AGC can be adjusted via each physical phone by going through the settings menu.

Jitter, Delay and Packet loss

As you should know when dealing with packet-switched networks, Jitter, Delay and Packet loss are factors that sooner or later will be experienced. Here we are learning what these factors are and how to treat them.

  • Delay is the time that a packet take to traverse the network from point A to point B. When latency occurs a caller might originate a new conversation and talk over the recipient or callee.
  • Jitter is related to latency or packets traversing different paths on a connectionless packet switched network.
  • When a packet is sent and can’t make its destination it’s called “Packet Loss”.

Common SIP Trunk Problems and Troubleshooting Steps

SIP uses the same mechanism to encode-decode the voice traffic as the H.323 Trunks. Both IP Trunks (SIP & H.323) use the hardware for the most part. When troubleshooting “static on calls” with SIP, you should consider verifying the following system configurations=

Shuffling and Hairpinning on CM – These NRs settings use the VoIP network carrying the phone call from end to end without the need of TDM resources, once the call has been setup and teardown. For those using different NRs check the  Intra and Inter IP-IP Direct Audio settings.

Keep in mind that announcements, messaging, and music on-hold will use a TDM Bus in the process.

CLAN, MedPros, and IPSIs – Its hard to troubleshoot static on calls when the hardware has been working for a long period of time. If the problem starts after a software update, or the installation of new SIP phones, it is a good idea to compare the Avaya Compatibility Matrix to see what firmware/software level you should be using. You can also try replacing these boards with new versions.

Codecs and QoS – Both SIP and H.323 require that the same codec type is implemented in both ends, as well as QoS markings from cradle to grave (including the Service Provider).

Before contacting the ITSP – For those with Session Border Controllers interfacing with the ITSP, it is a good practice that you do preliminary testing before engaging the Service Provider, and check the QoS Markings as the SBCE won’t pass them through to the ITSP.

Have the Service Provider analyze the traffic during peak hours, and monitor calls preferably with Wireshark to see if there are any mismatches in the configuration.

Question – What procedures do you follow when troubleshooting static on calls?

Other Resources

Cisco TechNotes on QoS with voice samples

SBCE Maintenance and Troubleshooting

 

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