VoIP- Telephony Basics

1.1What is Trunk circuit? Do telephony switches have addresses?

Trunk circuit is a single time slot, which is used to carry voice bits (PCM) on a trunk (or span or channel) between two switches. Telephony switches are addressable by Point Codes (The SS7 equivalence of IP address).

1.2What is SS7?

SS7 is the classic out-of-band signaling protocol for circuit switched. Mostly it does basic telephony signaling (ISUP), Intelligent network signaling (TCAP) and cellular signaling (MAP).

1.3What is bearer? What’s the difference between bearer and media?

'Bearer' is the circuit way of saying 'media'. In the circuit days media used to be mainly voice (and fax, data and video to a certain extent). In Internet Telephony the term media has much wider implication. It can be anything from white board images to real time text (including of course all legacy types of media).

1.4What is PCM? TDM?

PCM stands for Pulse Code Modulation. TDM stands for Time Division Multiplexing. Both pertain to the way analog signals, such as voice are sampled, encoded as digital bits, and transmitted over the digital wire (or wireless in some cases...). PCM/TDM is the main way circuit switches use to convey the bearer to each other.

1.5What is bandwidth? What is bit rate? What are codecs? any relation between those terms? 

Explanation taken from SIP Illustrated: Bandwidth and audio or video codecs are few of those buzz words, that everyone uses, but NOT too many people seem to know what they really mean…The lengthy explanation which follows, will try to reveal this mystery for you, but you will need to be brave and patient though…In order for any Voice/Media Over IP system to function, the process of Analog to Digital (A/D) conversion needs to take place (for instance, once the SIP signaling has setup a media session, your SIP phone needs to convert your voice/image to IP packets (e.g. G729/RTP/UDP/IP (voice); H261/RTP/UDP/IP (image)), which then need to be transmitted to the other SIP phone, and converted back by the other end point to audio/video information. Voice, audio and video are all sources of analog signal waveforms (i.e., Information, which is continuous in time and value), which need to be converted to digital information (discrete in time and value), so they can be transmitted as IP packets. The process of converting an analog signal to a digital representation (A/D) is consisted of three main phases: 1. Sampling (discretize time) 2. Quantizing(discretize value) 3. Encoding (digitize value).The inverse process of A/D, which is not surprisingly called the D/A process, is consisted of decoding the digital bits into the corresponding analog values and smoothing the rectangular waveform by applying a low pass filter width width Fmax. A device that can both encode an analog signal into a digital representation and decode a digital signal into its equivalent analog signal is called a coder-decoder, or codec for short. (Remember? codecs are these things, which are advertised by the SDP element in an SIP INVITE requests, for instance, “a=rtpmap:0 PCMU/8000”).In order to understand what bandwidth is, we need to realize that based on Fourier theorem any periodic function of a wave form (signal) is made up of sine waves each contribute one frequency component to the signal. The spectrum of these frequencies (to be precise. Fmax - Fmin) is called bandwidth. In the case of “lowpass” signal, i.e., message signals such as data signal, voice, audio or video, bandwidth is approximately equal to Fmax, since Fmin is negligible.This will help us to understand the connection between bandwidth and bit rate in digital systems. Now let’s indeed examine the relation between bandwidth and bit rate in digital systems. In the binary encoding phase of the A/D process, each of the binary digits, which is generated, needs to be transmitted to its digital destination by means of digital pulse. Since we have reciprocal relationship between wave width (period duration) and its frequency, we can say that the smaller the pulse width is - the higher its frequency, and thus the larger its bandwidth (F=Fmax=bandwidth). Since a digital signal is made up of multiple signal pulses of the same duration (and thus same frequency), its bandwidth is the same as the bandwidth of a single signal pulse (reminder: Bandwidth is the range of frequencies and for practical purpose may be approximated by the max frequency element. Therefore the bandwidth of a digital signal is actually determined by the signal pulse it uses (pulse width and shape). Since one pulse is used for one bit, the pulse width is proportional to the bit duration and inversely proportional to the bit rate. Therefore the conclusion is that in digital systems the bandwidth and bit rate are proportional. The general expression, which links bandwidth and bit rate in digital system is given by B = a*R when B is the bandwidth required for relaying a digital signal with a bit rate of R bits per second and 0.5 < a < 1 (depends on the pulse design). Once we digest all of the above, we can realize that codecs would mainly differ from each other by their sampling rate and step size. (To be honest, there are other differences as well (e.g., PCM encoding versus ADPCM encoding, different companding functions, linear quantization versus non-linear quantization, compression!), but we chose not say much about them, in order not to overwhelm you...). The higher the sampling rate and the smaller the step size the more accurate information we get, but the higher the number of steps (per second) is, i.e., the larger the number of quantums (symbols) to be encoded, and thus the longer the code words, i.e., large N, and thus the higher the bit rate, which means higher (PCM) bandwidth requirements. That means there is always a trade off between quality (accuracy) and bandwidth requirements. The more quality you want – the more bandwidth resources you have to throw in (pay!).

1.6Are call setup and call establishment the same? What are they for?

Yes, these are just two synonyms of the same thing. It is needed for making sure the other end point is there and is willing to take the call. In addition it is needed for the telephony network to allocate all necessary resources for the call, make other necessary preparations (e.g. open billing records, open stat records, prepare lawful intercept hooks) and potentially run call setup time services (e.g. translation, call forwarding). In the VoIP world call setup also enables the end points exchange information about their media capabilities (e.g. codecs which can be supported) and media properties (e.g. IP address for media exchange, port number, protocol).

VoIP- IP Basics

1.1How does IP router deliver IP packets to an IP host?  

First we need to realize that IP router is a computer (host), which has multiple network adapters, each associated with a different IP address. These network adapters enable the router to be a physical member of multiple subnets (e.g. LANs), thus enabling it to forward IP packets from one subnet to another. You can think of a router as an electronic version of an octopus, each one of its legs is connected ("belongs") to a different subnet... Upon receipt of an IP packet, the IP router takes a look at the IP portion (header) of the packet (datagram). In particular it examines the destination IP address field in the IP header. It looks up its routing table (using the destination IP address as a key) and finds the closets matching entry. This entry tells the router what is the next hop to go to, and via which network adapter (AKA interface) to do so. The next hop could be as simple as a host on the neighbor subnet, which its IP address is the one indicated in the IP packet (and thus is the final destination of the packet), or it could be another router that needs to take the packet further down the road. In order to forward the packet to the next hop, the router needs to discover the MAC (Media Access Control) address of it by using ARP (see below). Then it lets its MAC software and hardware (e.g. Ethernet module and Ethernet network adapter) take care of the physical encapsulation and delivery of the packet to the MAC hardware/software of the destination (e.g. the Ethernet network adapter of the next hop). Note that neighbor routers ALWAYS share at least one subnet (so they can physically/directly forward packets to each other).

1.2What’s the connection/relation between the Internet and IP? 

IP stands for Internet Protocol. It is the basic protocol (set of rules; language) that ALL computers on the Internet (must) use in order to speak to each other. For instance, some computers on the Internet might be capable of sending and receiving emails; Some others might only be capable of downloading web pages. Still all of them MUST be capable of speaking IP. Recall that email messages and web pages are always encapsulated in IP packets, just like letters are 'encapsulated' in postal envelops. Same for SIP messages...

1.3What’s ARP? 

ARP stands for Address Resolution Protocol. It is the protocol/process used to map IP addresses to MAC (Media Access Control) addresses. When hosts want to communicate with each other on the same segment of network cable (subnet) they need to know the physical addresses (MAC addresses) of each other. To do so they either broadcast ARP queries on the network segment or use ARP cache. The input of the ARP query/cache is the IP address of the destination host. The output is the MAC addresses of it.

1.4What’s the relation between IP and Ethernet? How are IP datagrams and Ethernet frames different from each other? 

IP packets (datagrams) are encapsulated in Ethernet frames. Computers (hosts) that share network segment (IP subnet) can talk to each other through their network interfaces (adapters). Every network adapter in the world has its own unique Ethernet (MAC) address. The logic built into it enables it to accept only frames whose destination MAC address match its own MAC address (except for broadcast frames which are always accepted). Perhaps now is the right time to realize that when a web client sends a request to a web server for a web page, the (HTTP) request is encapsulated in an IP packet. This IP packet may travel many routers until it finally reaches its destination (the web server). Every leap on its journey is consisted of ARP operation, MAC (e.g. Ethernet) encapsulation and MAC delivery between two neighbor network adapters (e.g. the adapters of two routers along the routing path, or the adapters of the web client and the next hop router). When a network adapter receives and accepts an Ethernet frame it throws away the bits that constitute the Ethernet (MAC) header and hands the remaining bits (IP header and its payload) up the stack to the IP module. This is called decapsulation and stack propagation.

1.5Is TCP reliable? Is UDP reliable? 

Yes, TCP is a reliable transport protocol. Its reliability is achieved by means of acknowledgements and segments retransmission. UDP is not reliable, but is very simple and is a good fit for packets that contain real time media information, such as voice or video. Loosing a single voice frame might not be that critical, but having to delay the transmission of the next voice frame due to lack of acknowledgement for its predecessor frame, might be critical and thus makes TCP a bad choice for real time communication.

1.6What is UDP? 

UDP stands for User Datagram Protocol. It is the simple standardize way to encapsulate a message and identify its application layer destination (by using port #). UDP header contains the source and target ports, message length and optional checksum.

1.7What is TLS? 

TLS stands for Transport Layer Security. It is a mechanism that provides hop to hop transport-layer security over connection-oriented protocols (e.g. TCP); It provides data integrity (no one has tampered with it), data confidentiality (no one has seen it or was able to understand it) and authentication (i.e., the sender is who he claims he is). TLS starts with a handshake phase that negotiates an encryption algorithm (e.g., AES, IDEA) and keys, and authenticates the server to the client (and vice versa) using certificates and trusted known CAs (Certification Authorities). Once the handshake is complete and data transmission begins, the data is encrypted using the keys and algorithm negotiated during the handshake phase (Symmetric cryptography).

1.8What is port number? Why is it called port? 

Port number is a logical identifier for a sender or receiver application. It has nothing to do with the physical port devices/connectors at the back of the computer... It enables multiplexing of IP packets between different applications sharing the same platform. In other words: A computer can run several applications at the same time. Each one of these may send/receive IP packets to/from different or identical sources. Each one of the incoming IP packets will find its way (up in the IP stack) to the correct waiting application based on the port number associated with it. The port number is added to a message by the application layer and is part of the transport header (e.g. UDP, TCP).

1.9What is an IP Switch? Is it the same as telephony switch? 

IP Switch is a new marketing buzz word. Normally it is used to describe an IP telephony server. Another word which is used interchangeably with the word 'IP switch' is Soft Switch. The IP switch (or Soft Switch) is different from the legacy telephony switch by the strict separation it keeps between the application logic (controller) unit and the media switching unit. This is in contrast to the centric way switching is done by circuit telephony switches.

1.10What is DNS? Any relation between Internet domains and IP subnets? 

DNS Stands for Domain Name System. It enables us to use names which are easy to remember (e.g. www.nba.com) rather than long and meaningless IP addresses (e.g. 10.32.111.2). Internet domain may contain many different IP subnets. Domain is a logical term, and its geographical meaning might be loose in many cases. For instance the domain 'intel.com' describes a virtual network that is consisted of many subnets in different geographical locations.

1.11What is IP multicast? Do I need to add a network adapter to my PC to be able to do multicast? 

By Bryan McLaughlin - Cisco: IP multicast is the ability for a host and the network to enable delivery to a group of interested receivers. A host can dynamically choose to listen/join a multicast group (using the same network adapter). IP routers that support multicasting, take care of duplicating the original IP packet and delivering the copies to all listeners to the group. In short IP multicast enables an unlimited number of hosts to receive a single data stream with 'no additional' load on the source or the network

IMS Technology

IMS enables a packet -based Network to provide multiple services on single  control/service layers via different access networks.

IMS Requirements:

1. IP Multimedia sessions
2. QOS
3. Service control
4. Roaming
5. Internetworking
6. Rapid service creater
7. Multiple access

IMS Protocols:

1. SCF(Session Control Function)

       Cirtuit Switched network :-

                          TUP (Telephony User Part)

                           ISUP(ISDN User Part)

                           BICC(Bearer independent call control)

     Packet Switched Network:-

                                 SIP

                                   H.323

 2) AAA(Authentication, Authorization and Accounting)

                                Diameter

 3) Other protocols

                            Megaco(H248)

                                  RTP/RTCP

IMS was originally standardized by 3gpp.

Key components or Nodes of IMS architecture:

1. HSS and SLF
2. CSCF
3. AS
4. BGCF
5. Media Gateways
6. MRF

1. HSS & SLF :

HSS(Home Subscriber System)========:

• It is a database of all subscriber and server data.
• It is an Evolution of HLR(Home location register) which is in GSM.
• It contain User Profiles used by control layer
• It contain subscription information used by service layer

User profile contains

• . User identity
• .allocated s-cscf name
• . Registration information and roaming profile
• .authentication parameters
• .control and service information

SLF (subscriber location function):

• An SLF is needed to map user address when multiple  HSSs are used.
• Network with Single HSS do not need SLF, On other hand networks with more than one HSS require SLF.
• Both HSS and  the SLF  communicate through the Diameter protocol

2. CSCF(Call session control function):This is a sip server. There are  three types of CSCFs , depending on Functionalities they provide :-

1. PCSCF(proxy)
2. SCSCF(Serving)
3. ICSCF(interrogating)

PCSCF:

1. It is the first point of contact between  IMS terminal (UE) and IMS network.
2.  Its main functionalities are:-

• It establishes number of IP sec security associations ( the ability to detect the content of message has changed  since its creation) towards the IMS terminal.
• It verifies the correctness of sip requests sent by IMS terminal and  forwards sip messages to SCSCF.
• it forwards registration requests  received from UE  to I-CSCF
•  It forwards requests  and answer to the UE.
• It also Includes Compressor  and de-compressor of SIP messages.
• It authenticate  the User  and asserts the identity of the user  to  other nodes in the network.
•   It  also include PDF(policy decision Function).It is integrated with PCSCF or Standalone unit.PDF authorizes media plane and manages Quality of service over media plane.

 3) The PCSCF may be located either in Visited network or Home network.

ICSCF:

I CSCF  is Sip proxy located at the edge of  an administrative domain.

• It's IP Address is published in the DNS of the domain(using NAPTR and SRV type of DNS records)
• It has an interface  to SLF and HSS.
• It Queries the HSS using Diameter cx Interface to retrieve the user location.
• It  also implements interface to Application servers, to route requests that are addressedto services rather regular users.
• It may optionally encrypt  the parts of sip messages that contain sensitive information about the domain, DNS names and capacity. This functionality is called THIG(Topology hiding inter-network gateway.
• It is located in Home Network , In some special cases such as ICSCF(THIG) it may be located in visited network as well.

SCSCF:

S CSCF is the central node of the signalling plane.

• It is a sip server  always located  in home network.
• It Uses Diameter  cx and DX to upload or download user profiles, it has no local storage. All necessary information is stored in HSS.
• It handles SIP registrations, Which allows to bind User location/IP address and SIP address.
• It sits on path of Signalling message and can inspect every message.
• It decides to which application servers the sip message will be forwarded, in order to provide services.
• It  provide Routing Services typically using ENUM lookups.
• It enforces the policy of the network operator.

MRF:(Media Resource Function)    It provides a source of media in the home network .

• It is used for playing Announcements(audio/Video)
• It is used for Multimedia Conferencing( ex: Mixing audio streams)
• It is used for TTS(text-speech Conversion) and Speech recognition.
• It is used for transcoding between different codec
• It is used for obtain statistics and do any sort of media analysis.

It is mainly divided into two types:

MRFC and MRFP

MRFC(Media resource function controller):

• It is a signalling plane node that  acts as a Sip user agent for S-cscf and which controls the MRFP with a H.248 interface.

MRFP:(Media resource function processor):

• It is media plane node that implements all media related  functions, such as playing and mixing media.

MRF is located in Home network

BGCF(Break Out Gateway Control Function) :

• It is a Sip server  used for routing  Calls between the IMS terminal and PSTN phone.
• It routes based on Telephone numbers.
• It break out occurs in same network as the BGCF  then the BGCF select a MGCF that will be responsible  for  internetworking  with the PSTN  and forwards the signalling to MGCF. Other wise it forwards signalling to BGCF  of another operator.
• The MGCF then receives the signalling from BGCF and  manages the internetworking with PSTN network.

The PSTN/CS Gateway:

The internetworking with CS network is realized by several components for signaling, media and control functions.

SGW(Signalling Gateway):

• It is an interface  with signalling plane of CS network.
• It performs Lower layer protocol conversion.
• It transforms ISUP over MTP  into ISUP over SCTP/IP.

MGCF(Media Gateway Control Function):

• It performs call control protocol conversion between Sip and ISUP.
• It interfaces SGW over SCTP.
• It controls MGW with a H.248(Megaco) interface.

MGW: (Media Gateway)

• It is an interface with Media plane of CS network.
• It converts RTP to PCM
• It also performs media transcoding when Codecs doesn't match.

Application Server(AS):

              AS is a sip entity that hosts and executes services .

• It interface with  the S-CSCF and I-CSCF using Sip and HSS using Diameter.
• This allows third party  providers  and easy integration and deployment of their value added services to the IMS infrastructure.

There are three different types of Application servers:-

 SIP AS :- It hosts and executes IP multimedia services based on sip.

 OSA-SCS(open service Access-service capability server):-  It inherits OSA capabilities to access the IMS securely from external network.

IM SSF(IP multimedia Service switching system Function):    It allows a GSM SCF(GSM service control function)to control an IMS session. IMS SSF provides intelligent gateway functionality between sip based IMS network an IN systems that use  protocols such as CAMEL,INAP,AIN and MAP.

VoIP Interview Questions and Answers -1

What is VoIP?

VoIP stands for Voice Over Internet Protocol or Voice Over IP. VoIP technology makes it possible to convert analog voice signal into digital data and transmits it over the Internet. (There are more likely possible pronunciations, as well as vo-ipp, have been used, but generally, the single syllable - voyp, as in voice - may be the most common within the industry.)

Why VoIP is better than traditional phone services?

Due to its cost efficiency, VoIP is more and more popular largely over traditional telepone networks. VoIP cuts companies’ monthly phone bill by approximately fifty percent. In addition to its cost efficiency, VoIP technology ensures many advanced features, like conference calling, IVR, call forwarding, automatic redial, call recording, etc. without extra fees.
VoIP offers cheaper international long distance rates that are generally one-tenth of what is charged by traditional phone companies.
Due to its portability VoIP is a really good option to avoid expensive hotel phone and cell phone roaming charges. Only a high speed broadband connection (and a plugged adapter) is needed and anyone can reach you at your local number - independently of your location. Most of the times in-network calls to other VoIP service subscribers are free even if the calling parties are located in different parts of the world.
By using Internet connection for both data traffic and voice calls, it is possible to get rid of one monthly payment that are usually charged by most Internet service providers. In addition, the Internet-based voice and data transmission enables to avoid wireless roaming fees and long distance rates.

What are the advantages of VoIP?

In addition to its cost efficiency, the feature-rich services and the metaphorical disappearance of geographical boundaries as that were mentioned above, VoIP has many other benefits as follows:

• VoIP technology enables to detect and process touch tones and DTMF responses
• VoIP systems can be automated easily
• VoIP systems allow to use more than one codec
• VoIP provides rich media service as more file formats can be used with these systems
• VoIP ensures a much more flexible system than hardware based solutions
• Most VoIP service providers provide a user control interface, typically a web GUI, to their customers so that they can change features, options, and services dynamically. 
• VoIP protocols run on the application layer and are able to integrate or collaborate with other applications such as email, web browser, instant messenger, social-networking applications, etc. 
  

How does VoIP work?

VoIP is usually based on the SIP system that is the recognized standard. Any SIP compatible device can talk to any other. Any SIP telephone can call another over the Internet - any additional equipment is not needed for this. You only need to plug your SIP phone into the Internet connection, configure it then dial the other person. You can also connect traditional analog phones to your VoIP network – in this case and ATA device is needed.
In VoIP systems, your analog voice is converted into packets of data (as little files), and then transmitted to the recipient through the Internet and decoded back into your voice at the other end. To make it quicker, these packets are compressed before transmission, a bit like zipping a file (it will be decompressed of course at the other party).
The advantages of converting analog signals into digital data can be summarized as follows: Digital format can be better controlled as it can be compressed, routed, converted, etc. In addition, digital signals are more noise tolerant than analog signals. Quality of Service (QoS) ensures real-time errorless data streaming that allows interactive data voice exchange as well.

What is the actual cost of VoIP telephony?

If you only want to use VoIP to communicate with other users in your VoIP network, you can do that free of charge. If however you want to be able to use VoIP to make and receive calls to/from people who are out of your VoIP network or do not have VoIP, you will need to subscribe to a VoIP service provider plan, and a gateway service may be also needed that provides a bridge between VoIP and the conventional phone networks.

Is it possible to replace the current traditional corporate PBX with a VoIP one?

Definitely yes. VoIP is a very cost-effective option for those companies who want to upgrade their old PBX systems and VoIP ensures new features that traditional PBX systems simply do not. To change to a VoIP system, companies can buy an IP PBX, but it is also possible to add some VoIP functionalities into an existing phone system.

What kind of equipments do I need for creating a VoIP system?

Getting started with VoIP is fairly simple. Assuming that you already have the 2 most important ingredients (a Windows PC or Mac computer and a broadband Internet connection), all you need to get started is the following:

• Some telephone or messaging software
• A microphone
• Headphones or speakers
  

You can use a headset of course rather than a microphone and speaker to leave your hands free.
In order to choose which software to use, it is worth to consider the followings. Using voice chat in G-Talk or Yahoo Messenger could be regarded as VoIP, so could the highly publicised Skype; but these are all proprietary systems. You can download them free of charge, but to talk to someone using G-Talk, the person at the other end also needs G-Talk. The same applies to Yahoo and, to a great extent, to Skype. They use their own special system that is not open and will not connect to other systems easily. So – especially for corporate using – it is rather recommended to use such a softphone as ConterPath X-Lite with a SIP enabled IP PBX or access to an Internet Service Provider.

Which protocols describe VoIP connections?

VoIP has been implemented in various ways using both proprietary protocols and protocols based on open standards. You can see the VoIP protocols below:

• H.323
• Media Gateway Control Protocol (MGCP)
• Session Initiation Protocol (SIP)
• H.248 (also known as Media Gateway Control (Megaco))
• Real-time Transport Protocol (RTP)
• Real-time Transport Control Protocol (RTCP)
• Secure Real-time Transport Protocol (SRTP)
• Session Description Protocol (SDP)
• Inter-Asterisk eXchange (IAX)
• Jingle XMPP VoIP extensions
• Skype protocol
• Teamspeak
  

The most commonly used one is SIP.  Session Initiation Protocol is IETF signaling protocol used for VOIP and other text and multimedia communication sessions such as voice and video calls over Internet Protocol (IP).
SIP can be used for creating, modifying and terminating two-party (unicast) or multiparty (multicast) sessions. These sessions include Internet telephone calls, multimedia distribution, multimedia conferences, instant messaging, file transfer and online games.

How to get started on VoIP programming?
The best way for creating any VoIP application is using a VoIP development kit. These SDKs are intended to provide a background support for your VoIP project by offering prewritten VoIP components. It is quite effective and comfortable to use these prewritten components, as you can save time and money as well. (During the softphone development that will be described below, Ozeki VoIP SIP SDK has been used for this purpose, that supports all the .NET programming languages, so C# as well.)
For using these toolkits, you have to add your preferred SDK as reference in you IDE. After you have added it, you can reach all VoIP components that are needed to be able to define the behaviour of such VoIP applications as softphones, call recorders, IVR menu systems, software-based IP phone systems (PBX), etc.

How to make VoIP calls?

In order to be able to make voice calls by using your own software application you need to connect your system to the telephone network. This can be done in three ways:

• Option 1: Use a VoIP telephone adapter
  A VoIP telephone adapter is a hardware device that can be connected to your Ethernet LAN or to your PC. There are VoIP telephone adapters for GSM lines, for standard analog telephone lines and for ISDN lines. When you connect this hardware to your Ethernet LAN, it will receive an IP address. You need to configure this IP address in your VOIP SDK.
• Option #2: Use a SIP Account provided by VoIP telephone service provider
  There are many VoIP telephone service providers worldwide that offer phone service over the Internet. You need to subscribe to their service, and you will receive a SIP account (including an IP address, a username and a password). You need to configure the SIP account details in your VoIP SDK.
• Option #3: Use your existing office PBX if it is a VoIP system.
  If you already have an IP telephone system, you need to connect your VoIP SDK to that over the LAN. The SDK can log in to the phone system by using a SIP account and it can make telephone calls just like any other
  

8x8 VoIP Test

This Online Testing Utility will open a socket-connection to your browser and pass simulated VoIP Traffic to your home/office computer. This test measures the quality and performance of your Internet connection between your home/office network and the 8x8 servers.
click on below link:

8x8 voip Test 

Internet and Voip Speed test

This speed test measures the quality and performance of Internet connections for Voice over IP by simulating real VoIP sessions between our server and your computer. VoIP transmission consists of Session Initiation Protocol (SIP) signaling and Real Time Protocol (RTP) udp data stream. We test only real-time part as the most important factor of call quality. But first your Internet connection (download and upload) is tested. After the test you can see results and comments depending on your connection measured parameters. You may share your results in forums and webpages. You will see instructions how to do it when your test will be complete. So let's select server near you and go to speed test in your location. You can also perform our Ping test and India Speed Test.

If you have any problems with VoIP test, check if the JAVA SUN is installed in your web browser. Click here to check your JAVA installation. It is strongly recommended to be up-to-date with the newest JAVA version. Sometimes your browser (Mozilla Firefox, Chorome) may ask to permin VoIP TEST applet to run with JAVA - you ought to allow your browser to do it. Otherwise, the "js/java problem" will appear in jitter and packet loss result fields.

How to install VM player on Ubuntu 14.04

How to install  VM player on Ubuntu 14.04

Introduction

VMware Player allows you to run entire operating systems in a virtual machine, which runs on top of Ubuntu or Windows. To the guest operating system (the one running inside the virtual machine), it appears as though it were running on its own PC. The host operating system runs the VMware Player, which provides the guestwith resources like network access. It can be downloaded for free from VMware.

Virtual machines configured with an operating system and applications ready to perform a specific function are called virtual appliances. An appliance can be created using certain VMware products, or you can download ready-made appliances. A wide variety of appliances (both certified and other-wise) are available from VMware's Appliance Marketplace.

If you are a Windows (or other operating system) user looking for an official Ubuntuappliance to run, you will want to read only the last section.

If you are an Ubuntu user who wishes to install or use the VMware Player software, continue reading.

Installing VMware Player on Ubuntu 14.04

1. Install required packages build-essential and linux-headers:
1. sudo apt-get install build-essential linux-headers-$(uname -r)
2. Download the latest VMware player e.g. VMware-Player-6.0.2-1744117.x86_64.bundle (download the bundle version, not the rpmone) and run it as root using gksudo. You'll get a graphical installer that installs VMware player for you.

• gksudo bash ~/Downloads/VMware-Player-6.0.2-1744117.x86_64.bundle

Note: this assumes the location of your Downloads folder is /home/$USER/Downloads. *If nothing appears, you may need to make the file executable. You can do so with this command:

chmod +x ~/Downloads/VMware-Player-6.0.2-1744117.x86_64.bundle

(again, with the assumption of your Downloads folder location). After completion, VMware player is installed and should show up in the menu under Applications → System Tools → VMware Player (for Unity users, it should come up in the search results for vmware player).

As well, you may notice that when trying to create a new virtual machine, vmware player will complain on the terminal output(if it was started from the terminal as vmplayer) that:

VMware Player is installed, but it has not been (correctly) configured for your running kernel. To (re-)configure it, your system administrator must find and run "vmware-config.pl". For more information, please see the VMware Player documentation.

vmware-config.pl is not present anymore in the latest vmware-player versions (seems to have been superseded by vmware-modconfig). If you have this problem you may instead need to check if you have a /etc/vmware/not_configured file and, if so, delete it.