Voip information Introduction
VOIP is an acronym for Voice Over Internet Protocol, or in more common terms phone service over the Internet.
If you have a reasonable quality Internet connection you can get phone service delivered through your Internet connection instead of from your local phone company.
Some people use VOIP in addition to their traditional phone service, since VOIP service providers usually offer lower rates than traditional phone companies, but sometimes doesn’t offer 911 service, phone directory listings, 411 service, or other common phone services. While many VoIP providers offer these services, consistent industry-wide means of offering these are still developing.
How does VOIP work?
A way is required to turn analog phone signals into digital signals that can be sent over the Internet.
This function can either be included into the phone itself or in a separate box like an ATA .
VOIP Using an ATA
Ordinary Phone —- ATA —- Ethernet —- Router —- Internet —- VOIP Service Provider
VOIP using an IP Phone
IP Phone —– Ethernet —– Router —- Internet —- VOIP Service Provider
VOIP connecting directly
It is also possible to bypass a VOIP Service Provider and directly connect to another VOIP user. However, if the VOIP devices are behind NAT routers, there may be problems with this approach.
IP Phone —– Ethernet —– Router —- Internet —- Router —- Ethernet —- IP Phone
Applications using VOIP
Traditional telephony applications, such as outbound call center applications and inbound IVR applications, normally can be run on VOIP.
Why use VOIP?
There are two major reasons to use VOIP
- Lower Cost
- Increased functionality
In general phone service via VOIP costs less than equivalent service from traditional sources. This is largely a function of traditional phone services either being monopolies or government entities. There are also some cost savings due to using a single network to carry voice and data. This is especially true when users have existing under-utilized network capacity that they can use for VOIP without any additional costs.
In the most extreme case, users see VOIP phone calls (even international) as FREE. While there is a cost for their Internet service, using VOIP over this service may not involve any extra charges, so the users view the calls as free. There are a number of services that have sprung up to facilitate this type of “free” VOIP call. Examples are: Free World Dialup and Skype for a more complete list see: VOIP Service Providers
VOIP makes easy some things that are difficult to impossible with traditional phone networks.
- Incoming phone calls are automatically routed to your VOIP phone where ever you plug it into the network. Take your VOIP phone with you on a trip, and anywhere you connect it to the Internet, you can receive your incoming calls.
- Call center agents using VOIP phones can easily work from anywhere with a good Internet connection.
Plain explain: IP-phone
We all strive to get a quality telephony, but when we do some analyze of the voice quality issues and most common reasons that may lead to the problems with the VoIP service quality we should take into account the VoIP equipment, including IP-phones.
Looking closer: what is an IP-phone
VoIP phone is a phone with built-in IP port (Ethernet), and it supports the transport protocols that are used to connect and transport data to/from IP-phone to/from VoIP service provider (or IP-PBX or SIP server) – this allows you to make and receive voice (and in some cases video) calls.
IP phone is a very hike in form and functionality to the usual traditional phones to which we are accustomed to and can see in any office on the desktop.
Fine, but all the same, what is the “IP”, it’s sounds so familiar to me, one guy from the IT Department is constantly mention the word…
If you think that you have heard of the term “IP” before ” you’re right. IP stands for Internet Protocol, a way of data packets passing through a local network (LAN) or Wide Area Network (WAN).
This is the same technology that is used for data transmission in the data networks and we use that every day. You found this site and got all the information that you are reading now using IP.
This is also where the term VoIP comes from. VoIP stands for Voice over IP (Internet Protocol) and is a widely used term nowadays, associated with voice calls over the Internet.
How an IP-phone (VoIP phone) works?
Briefly: VoIP phone (aka IP-phone which is the same) takes the analog voice from microphone and turning it into digital packets, which are sent over the network using IP. VoIP phones can also convert digital packages in analog flows voice and play sound to your headset.
For this actions including data transfer and analogue to digital and digital to analogue conversion VoIP phone uses a specific set of VoIP protocols and voice codecs.
This part may lead to depression, but take it easy, we will not push on full throttle here.
VoIP Protocol defines how your voice packets will be transported over the IP network. In most cases IP-phone support one VoIP protocol, but some of them can be flashed with different firmware and this can be changed (for example Cisco phones may support SIP, MGCP, SCCP and pretty old H.323, but you must flash your device to change this, so better to know which protocol you need before you will buy an IP-phone).
VoIP signalling protocols you should know about
SIP (Session Initiation Protocol)
SIP is a standards-based Protocol that is used and supported by almost all VoIP phone systems and services nowadays (that’s why our web site has URL voip-sip.org);
SCCP (Skinny Client Control Protocol by Cisco)
SCCP is a proprietary Protocol used by Cisco Call Manager and Cisco phones;
MGCP (Media Gateway Control Protocol)
MGCP is newer version of SGCP (Simple Gateway Control Protocol) and this is an old VoIP Protocol. We must mention it, but it’s rarely in use nowadays.
H.323 was the first “VoIP standard” since 1996. It is widely implemented by voice and videoconferencing equipment manufacturers, is used within various Internet real-time applications and widely deployed worldwide by service providers and enterprises for both voice and video services over IP networks.
At the moment I’d say that H.323 is retiree. Some VoIP providers still support it, but only because they have this old hardware (gatekeepers) and it’s supported in addition to SIP, but not as a main signalling protocol.
Voice codec (this term formed by two words: COde and DECode) is responsible for the conversion of an analogue sounds (voice) to digital form that can be stored and sent/received in data networks. Voice codec affect on two main parameters of your “phone line” – the voice quality and IP bandwidth requirements. Modern VoIP phones supports multiple codecs, like GSM, iLBC, G.711 (ulaw/alaw), G.722, G723, G729. We have more detailed codecs overview
More and more people start small business now, it is very important to keep communication with our customers. In normal, most of us just use a simple telephone or use skype to do that. It can fit our basic requirement, but it has obvious shortage, for example, it makes our business looks like a very small business, a personal business. We hope to make our business looks like a big business and our company looks like a professional company. We want to establish a voice interaction for our customers, such as prompt “welcome to xxx company. Please dial the extension or zero for assistance”, etc. Can we do that?
Of course, we can do that! There is a very simple solution to do that by using VOIP. This document gives a step by step guide on how to establish a VOIP network for our business. The solution will be as simple as possible, almost just need only one PC with Windows system.
2. What’s VOIP?
VOIP is “Voice over Internet Protocol”. It is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks. VOIP can be a benefit for reducing communication and infrastructure costs.
In another way, it is very easy to use VOIP to establish our communication system with full features.
In normal, we need to setup three main components: IP-PBX, phones (or soft-phones) and VOIP carriers’ service which you can use to call other people in PSTN network(Public Switched Telephone Network, it is our traditional telephone network).
There are lots of IP-PBX. Some of them are hardware based devices, some of them are software based servers.
Of course, here we suggest miniSipServer to you. miniSipServer is a professional SIP PBX for Windows and Kubuntu/Linux systems. It has all features we need. Most important, it is so easy that we can setup and run it in several minutes!
We can buy SIP phones from Grandstream, Cisco, Linksys, etc. If we still want to use our traditional analog telephone, we can buy a SIP adapter from them.
To get started, we use a softphone and run it in the same computer or another computers. miniSipPhone is suggested here..
2.3 VOIP carriers
Lots of VOIP carriers can provide SIP services. I suggest following VOIP providers:
Once register to these VOIP providers, they will give you a SIP account information, such as SIP server address, account name and password, etc. These information will be used when we configure our IP-PBX.
Following figure describes a simple environment for small business or home based business.
In our demo scenario, the small company only has two members, Holly and G.T.
Holly’s extension number is 100, and G.T’s extension phone number is 101. Both of them use SIP softphones. The IP address of Holly’s PC is 192.168.1.100. The IP address of G.T’s PC is 192.168.1.101. miniSipServer will be installed on another PC whose IP address is 192.168.1.110.
The company establishs connection with PSTN through a VOIP carrier’s network.
We will follow subsequent steps to establish our VOIP network.
Connect local users to miniSipServer.
Connect miniSipServer to VOIP providers’ network.
Configure some wonderful advance services for our small company, such as auto-attendant, ring group and pick-up, and so on.
4.1 Step1: setup miniSipServer
There are several kinds of miniSipServer, such as 20 client, 50 clients etc. For example, “100 clients” version can support 100 extensions. Since there are only two members in our scenario, “20 clients” version is enough.
Before we install miniSipServer, please make sure that PC and network are working well firstly.
Then, please click .EXE file to setup miniSipServer and run it! It is unnecessary to configure anything! So easy!
If everything is fine, miniSipServer should run as following figure.
4.2 Step2: connect local users to miniSipServer
When miniSipServer is installed, it will create three default extensions automatically. Please click button “local users” in main window to check it. The default password for these extensions are 100, 101 and 102.
We begin to configure Holly’s softphone to connect to miniSipServer.
Holly use miniSipPhone as her softphone. After install miniSipPhone, please click menu “File -> SIP account”.
In the pop-up window, please configure SIP account like following figure.
The key items are described as:
|SIP server address||192.168.1.110|
Please click ‘OK’ button to complete miniSipPhone configuration. It will try to register to miniSipServer. If it successes, miniSipPhone should display telephone number and be ready to make calls.
Now, Holly’s extension has connected to miniSipServer rightly. We can follow the same step to configure G.T’ extension. If you have other kinds of SIP clients/phones, they should be configured same information.
Both Holly and G.T’ extensions have been connected to miniSipServer. We can show miniSipServer’ local user information to check their status. Their icons should be blue.
After we finish this step, the basic VOIP network is established. Holly and G.T can call each other. Holly can dial ‘101’ to call G.T, and G.T can also dial ‘100’ to call Holly.
4.2.1 Add a new extension in miniSipServer
In above configuration, we use default extensions ‘100’ and ‘101’. In future, with the growth of company, more and more people will join with us, we need add more extensions to support them. So we can do it as following:
In the ‘local users information’ window, please click ‘Add’ button to add a new extension.
In normal, we can just assign extension number and password to a new extension. The new employee can use this new account information to configure his/her SIP phones.
4.3 Step 3: connect miniSipServer to VOIP provider
It is so easy to establish internal VOIP network and Holly and G.T enjoy it. It is time to establish connection with customers now.
In normal, if we want to make call to outside or receive a call from outside, we need a VOIP gateway connect our miniSipServer and traditional telephone or we need VOIP provider to do it for us. We decide to connect our miniSipServer to VOIP providers’ network and we select BroadVoice as our VOIP provider.
After we request a SIP account from BroadVoice, for example, the account is ‘723123456’, we will use use this account to configure miniSipServer to connect BroadVoice.
In the miniSipServer main window, please click button ‘External lines’ to add an external line information.
In the pop up window, please click button ‘Add’ to add an external line with BroadVoice account information.
The key items are described in below table. Of course, you can update it according to your own configuration.
|Peer server address||sip.broadvoice.com|
|Peer server port||5060|
|External line (Account)||7321234567|
|All local users can use this external line to make outgoing calls||Yes|
Some VoIP providers require different authorization numbers with their accounts. In this scenario, we must configure “authorization ID” with such numbers. By default, authorization number is same with voip account, then it is unnecessary to configure “authorization ID” item or configure it as same as voip account.
Because we hope both Holly and G.T can make outgoing call, we select ‘All local users can use this external line to make outgoing calls’.
Here we configure ‘Automatic attendant’ to support receiving incoming call from outside.
If the external line success to connect to peer server ( VOIP provider’s network or VOIP gateway), the icon of the external line should be gray and without cross flag.
Then, we describe some details about making outgroup calls and receiving incoming calls.
4.3.1 Make outgoing call
As we have confirmed in above sections, Holly and G.T can call each other by dialing their extensions number directly. If we want to make outgroup call to our customers, how can we do it?
Since the external line is connected to VOIP provider’s network ( or VOIP gateway), it is no problem to call outside customers, but we have to mention that we need add prefix ‘9’ before we dial our customers numbers. In miniSipServer, prefix ‘9’ is the default outgroup prefix which is used to distinguish call type. For example, if the customer’s number is ‘7321234568’, we need dial ‘97321234568′.
4.3.2 Receive incoming call
When we configured external line, we has indicated ‘automatic attendant’. So when customers calls in, miniSipServer will prompt to enter extension number. For example, once customers calls “7321234567” (the external line number provided by VOIP provider and configured in miniSipServer), they will hear “Welcome, please enter extension number” and customers can enter ‘100’ to call Holly or enter ‘101’ to call G.T.
In another way, the default announcement can be modified and replaced. We can change it according to our real requirement, for example, we can change it to “welcome to xxx company,…”. Please refer to the “Automatic attendant” online document.
Because miniSipServer uses “RFC2833” DTMF mode and SIP-INFO to collect the customer’s input digits, we must confirm with VOIP providers that they can support these DTMF modes.
5. Advance configuration
When we finish above configurations, we success to establish our basic VOIP system. Is it enough? Of course, no! We want more useful services to support our communication, such as Voice mail, ring group, etc.
6.1 How to change SIP port?
The default SIP port is 5060 (UDP) which is defined in SIP standard. But in some special scenarios, we need change this port to another, for example 5090.
To do that, we need modify MSS and SIP phones configurations together.
In MSS, please click menu “data -> system information -> SIP” and change the port. Please refer to following figure.
After that, please restart MSS to enable this modification.
In miniSipPhone, we need indicate it to work with the new port. In SIP account configuration, please set “port” to “5090”:
Some SIP devices, such as Xlite, don’t have ‘port’ configuration, then we need configure server port with server address together. For exmaple, the server address might be ‘192.168.1.110:5090’.