Are you looking for job as a VoIP engineer/administrator?

Or are you thinking of leaving your

current position for a new job as a VoIP engineer/administrator with a new company in a Unified
communication networking?

If you answered yes to either of those questions, this article is for you.

A Network Engineer/Administrator (VoIP) position is a higher-level position, often with a

“junior” or “senior” prefix. The major responsibility of a VoIP engineer is to design and
implement both the hardware and software technologies needed for a VoIP setup. They have to
customize VoIP networks as per the organization’s needs, such as adding software and hardware,
performance monitoring, troubleshooting, logging errors, backing up and restoring data,
assigning permissions to users, and helping users for VoIP network issues.

Before facing any interview for a VoIP engineer position, make sure that you have enough
knowledge on the below technologies:

General Network concepts

 * Fundamentals of OSI & TCP/IP model

 * IP Addressing & Summarization
 * Basics of routers and routing (EIGRP,OSPF,BGP)
 * Basics of LAN Switching (VLANs, Inter-VLAN routing, STP)
 * Network Services (Telnet, SSH, NTP, DHCP, Syslog)

VoIP Topics

 * IP phone boot process

 * Phone registration through CME (IOS based)
 * CME features- call park, call pick-up, paging, intercom
 * Understanding of clusters, ISDN configuration
 * VoIP dial peers and POTS
 * Cisco ip phone registration through CUCM (Manual/Auto/BAT)
 * Understanding and Implementation of voice gateways MGCP,H.323,SIP
 * Signaling and Flow of voice gateways
 * Call routing elements : route group, route list, SLRG
 * Partition & CSS
 * Media Recourses
 * Integration of CUCM with CUC via SCCP & SIP
 * Mobility : device/extension mobility, MUA, Single Number reach (SNR)
 * Survivable Remote Site Telephony (SRST)
 * Call manager fall back
 * Resource Reservation Protocol (RSVP)
 * Automated Alternate Routing (AAR)
My Best Questions for an Interview for VoIP Engineer/Administrators:
All of the questions below are very common and must be prepared for before facing any
interview for the unified communication environment.

Q. What is VoIP?

A: Voice over Internet Protocol (VoIP) is the technology to send your voice (analog data) over
the internet (digital data) to an end user. It enables users to use the Internet as the transmission
medium for voice calls at a very low cost.

Q. How does VoIP (voice over Internet protocol) work? What makes it different from
traditional phone service?

A: In VoIP, phone conversations are converted to packets that flit all over the Internet or private
networks, just like e-mails or Web pages, though voice packets get priority status. The packets
get reassembled and converted to sound on the other end of the call but in traditional phone
service, a phone conversation is converted into electronic signals that traverse an elaborate
network of switches, in a dedicated circuit that lasts the duration of a call.

Q. What are some disadvantages of using VoIP?

A: VoIP is far better than traditional telephony but it has some drawbacks as listed below:

 * Some VoIP services don’t work during power outages and the service provider may not
offer backup power.
 * Not all VoIP services connect directly to emergency services through 9-1-1.
 * VoIP providers may or may not offer directory assistance/white page listings.

Q. What basic set-up equipment is needed for VoIP?

A: For general VoIP set up we require the following things:

 * Broadband connection
 * VoIP phone
 * Nexton soft-switches
 * Router
 * Audiocodec
 * Astric server

Q. What is PVDM and what is the use of it?

A: PVDM stands for Packet Voice DSP (digital signal processor) Module and it enables Cisco
Integrated Services Routers to provide high-density voice connectivity, conferencing, and
transcoding capabilities in Cisco IP Communications solutions.

Q. What is VoIP gateway and explain the basic features of it?

A: A VoIP gateway works as a bridge between an IP network and the PSTN. It converts analog
telephony signals to digital.

VoIP gateways include the following features:

 * Call routing, packet processing and control signalling managementVoice and fax
compression/decompressionExternal controller interfaces, for example to a soft switch,
billing system or network management system

Q. What is the difference between MGCP & H.323 Voice gateways?

A:

MGCP

 * Uses clear text for call controlUses a client-server modelIdeally positioned for service
providers (centrally located call agents)Centralized management and control (Dial plan,
etc.)Enhanced call survivabilityBetter feature interaction with capabilities like Caller-
IDSupport of QSIG supplementary services with CCM

H.323

 * Uses Abstract Syntax Notation 1 for call control messages. Uses a peer-to-peer model.
Scales well in an enterprise. Fractional PRI support. . Caller-ID support on analog FXO.
Many more TDM interface types and signaling. Gateway-resident applications like TCL
and VXML. CAC network design with H.323 Gatekeepers. No release dependencies
between GWs and CCM. Call preservation for SRST on PRIs. NFAS support

Q. What is the difference between Transaction, Dialog and Session?

A: Transaction: A Transaction refers to a fundamental unit of message exchange, between the

SIP user agents.; It basically includes a request-response cycle.

Dialog: A peer-to-peer relationship between two use agents. It is usually created through
generations of SUCCESSFUL final response.

Session: A Session refers to the exchange of media between two or more endpoints.

Q. What you will do if you get a crackling sound on VoIP line while talking?

A: Often we get this kind of issue with analog phones which create a crackling sound when you
talk, but is not common for VoIP provider or internet connection as digital systems don’t
generate sounds like this. It’s only possible that it could be from the person on the other end who
is also on an analog telephone line but if this happens with different people then it’s likely your
phone. Try to replace the phone.
Q. What is SLRG & why do we use it?

A: SLRG (Standard Local Route Groups) eliminates the pairing between the gateway and the
Route Pattern, thus creating a more flexible method of selecting a PSTN gateway.

Because it reduces the number of route patterns that need to be created per country, a huge
amount of administrative overhead can be saved, especially for organizations with a large
number of sites.

Q. What is the basic difference between VoIP and POTS (Plain old telephone systems) dial-
peers?

A: VoIP dial peers route calls to other VoIP systems via IP protocol where POTS dial peers
route calls to legacy PBX systems via local ports which can be analog (like FXS, FXO) or digital
(like E1/T).

Q. What is sip trunk and what are the basic benefits of using SIP trunk?

A: SIP Trunk is a voice call connection placed over your Internet connection. This VoIP “trunk”
(or phone line) connects to a provider who routes your calls through their gateway and usually
has very reasonably lower rates on long distance calls, international calls and in-bound toll free
calls.

Additionally, SIP trunks can carry instant messages, multimedia conferences, user presence
information, and Enhanced 9-1-1 (E9-1-1) emergency calls.

 * Eliminate BRI and PRI subscription fees because SIP trunk connected directly to an
Internet telephony service provider
 * Eliminate IP-PSTN gateways (or even your entire PBX)
 * Low cost long distance calls, international calls, etc.
 * Expansion of lines is dependent on bandwidth, which can easily be increased if needed.
It means that with SIP trunking you don’t need to buy lines in blocks of 24 or 32. Instead,
you can buy the bandwidth you need in smaller increments.

Q. What is the difference between E1 and T1 Link? In which country are we using t E1 and
T1 for digital transmission?

A: The main difference between E1 and T1 is the data rate. T1 has a data rate of 1.544 mbps and
E1 has a data rate of 2.048 mbps.

Other differences between T1 and E1 lies in the number of channels (E1-32 Channel and T1-24
channels) but speed (64 kbps) remains the same for both links: may be for inter – connection
between the E1 and T1 lines. This is interconnected because it is used for international
connectivity purposes.
T1 is used mainly in the United States, Canada, Hong Kong and Japan. E1 is mostly used in
Europe.

Q. How many channels are in E1 and T1 link? What is the operational bandwidth of each
channel in each link?

A: E1 link contains 32 channels where T1 link contains 24 digital channels and each channel of
both T1/E1 gives you 64 kbps bandwidth.

An E1 link consists of 30B channels and 1D channel and each channel has 64k Bw.

So for E1 – you get 30*64 = 1920kbps

A T1 link consists of 23B channels and a D channel and each channel has 64k Bw. So for T1 –
you get 23*64 = 1472kbps

Q. What is the signalling and what is the difference between CAS and CSS?

A: Signalling is a way of information exchange to establishment and control of a

telecommunication circuit and the management of the network.

Common channel signalling (CAS) uses a dedicated channel for the signalling where Channel
Associated Signalling (CAS) conveys signalling information relating to multiple bearer channels.
These bearer channels therefore have their signalling channel in common.

CCS with E1 =30 B channels and 1 D channel

CCS with T1 = 23 B channels and 1 D channel

CAS with E1 = 31 B Channels

CAS with T1 = 24 B Channels

B=Bearer channels responsible for carrying voice signal

D=Data channel responsible for signalling control

Q. What are the basic differences between G711 and G729 codec?

A: G729 is a compressed audio codec with better tolerance for packet loss and jitter than G711.
G729 uses 33 Kbps of bandwidth whereas G711 uses 87 Kbps. G729 is compressed but still
sounds very good in poor network but G711 sounds better only with good network conditions.

Q. Can we configure trunking between Cisco and Avaya PBX?

A: QSIG PRI trunking configuration can be used to establish trunking between Cisco and Avaya
PBX.
Q signalling (QSIG), a protocol for Integrated Services Digital Network (ISDN) communications
based on the Q.931 standard, is used for signalling between digital PBXs.

The questions above are very tricky and important from the standpoint of clearing any
interview for a VoIP network engineer/administrator position. It is not possible for anyone
to explain all kinds of questions, but you can get more frequently asked interview questions
for VoIP Network Engineering Jobs from the download link posted here. If you find any
difficulty in answering any questions, then you can write me @ Comment section.

Tips for Preparing for an Interview

 * Study: Before an interview, take a quick recap of relevant technologies.

 * Updated Resume: Read your resume through; don’t copy and paste your resume. You
must be aware of your strengths and weaknesses.
 * Professional Certifications: One of the best ways to prove the technical skills
mentioned in your resume is through certifications. This gives a new employer an easy
way to understand your knowledge level.
 * Updated LinkedIn Profile: Update your LinkedIn profile regularly; make sure that
your work experience, qualifications, and project details match with your resume.

With this article I have tried to help/guide candidates/Job seekers for their interview preparation
in VoIP domain.