What is VoIP (Voice Over IP)? | Definition and Features

VoIP sends voice calls over the Internet using IP protocols. This has changed the way we talk to each other. In this article, I will discuss the basics of VoIP technology. I will examine how it builds its network, explain how it works, and highlight its benefits.

In brief, I will look at the protocols used for VoIP and how they change global calls. In the end, we will show the details of this new technology. Let’s see the world of voice calls and its part in today’s talking!

What is the VoIP Protocol in Voice Communication?

The VoIP system is a set of rules that sends voice sounds over the web. It uses Internet rules to move voice information. As a result, the Internet can transmit voice sounds. This system is a key tool that lets people talk using voice.

The system sends the voice message as data bundles instead of through old phone wires. Computers perform this job, not the old phone systems.

Mobile VoIP is a set of Internet rules for voice on cell phones. Also, when two mobile devices talk, the voice message is sent using the web. This happens because of the Internet’s rules.

Today, mobile devices can link to the web using Wi-Fi or data. This link lets you do things like browse websites, get and send emails, and get files. It also works with apps for quick texts and sending voice messages.

The rules used to send voice calls over the Internet are also called VoIP. These rules are business versions of the ARPANET’s “Experimental Voice Protocol Network.”

VoIP lets voice calls move over the Internet, like a local office network. You also need to know the difference between Voice over IP and Telephony over IP.

VoIP is the technology that lets people talk using the Internet. Meanwhile, telephony over IP is a phone service for everyone who uses VoIP.

VoIP Advantages

The best advantage of this service is that it stops high phone costs. This is very helpful for long-distance calls.

Also, sending calls and data on one network with VoIP saves money. VoIP-to-VoIP calls are most often free. But, VoIP to regular phone calls costs the VoIP user.

Using the IP system saves a lot on global calls. You only pay a set internet fee, and it covers anyone who can use an IP link.

As this tech improves, more people will use mobile VoIP. The main reason is that longer calls do not cost more. Also, you can make world calls for the same price.

Making codecs for VoIP helps send more miniature data packets. This needs less Internet speed. As ADSL lines improved, this way of talking became common. People like it most for international calls.

There are two kinds of PSTN-VoIP services: Direct Inward Dialing (DID) and Access Numbers. Access Numbers need the VoIP user to type their extension number.

The Direct Inward Dialing (DID) feature links the caller straight to the VoIP user. Access Numbers cost the PSTN caller the same as a local call. These prices are much lower than what area operators charge.

How the VoIP System Works

It makes jobs easy that are hard with old phone networks. For example, local calls go right to your VoIP phone, no matter where you join the network.

You can also take a VoIP phone when you travel and make calls anywhere with Internet access.

You can use toll-free numbers with VoIP in countries like the US and the UK. Call center staff using these phones work well anywhere with Internet access.

But some plans include features that PSTN usually costs extra for. For example, some countries do not offer options like three-way calls, callback, auto redial, and caller ID.

So, the VoIP system gives these tools to people, offering simpler and cheaper talking.

Network Architecture

The standard shows three main parts in its structure:

• Terminals: They are used to replace existing telephones. You can use them in both software and hardware.
• Gatekeepers: They are the core of the whole VoIP system and will replace the old center. Usually, in software, if there is any, all calls go through this software.
• Gateways: This links to the old phone network and works clearly for the user.

With these three parts, the VoIP network can connect two branches of the same company.

The benefit is clear: all calls between the offices are fully open. The same setup can be used with suppliers, saving money.

VoIP protocols: It is the language that different devices use to connect. This part is key because the speed and ease of calls depend on it.

In order of age (from oldest to newest):

• H.323 – Protocol defined by ITU-T
• SIP – Protocol defined by IETF
• Megaco (also known as H.248) and MGCP – Control Protocols
• Skinny Client Control Protocol – Cisco’s proprietary standard
• MiNet – Mitel proprietary protocol
• CorNet-IP – Siemens’ proprietary protocol
• IAX – Original standard for communication between Asterisk PBXs (A standard for other data communication systems, currently in version 2 – IAX2)
• Skype – Proprietary peer-to-peer protocol used in the Skype application
• IAX2 – Communication standard between Asterisk PBXs that will replace IAX
• Jingle – Open standard used in Jabber technology
• MGCP – Cisco Proprietary Standard
• WeSIP – Free license standard from VozTelecom

VoIP is known to have many good points for both companies and everyday people.

What are VoIP Parameters?

This is the main problem for both VoIP and all IP apps today. Giving good service over the Internet is hard because it uses a “best effort” method.

Also, limits on internet speed along the route make this worse. Because of this, many problems happen in keeping good service. So, users feel the service is not steady.

Codec

Encryption is needed to send voice over an IP network. Codecs change and shrink audio or video. You must complete this step to enable later decoding and expansion.

Different codecs use different amounts of bandwidth. Usually, the quality of the sent data matches the bandwidth.

You must convert the audio to digital form to send it over IP. The codecs used are G.711, 722, 723, and G.728. These codecs save bandwidth by compressing the audio. As a result, they let more calls use the same channel.

These codecs show the following sizes on their label:

• G.711: 56 or 64 Kbps bit rate.
• G.722: 48, 56, or 64 Kbps bit rate.
• G.723: 5.3 or 6.4 Kbps bit rate.
• G.728: 16 Kbps bit rate.
• G.729: 8 or 13 Kbps bit rate.

This process still consumes bandwidth; for example, the G729 codec uses 31.5 Kbps in its transmission.

Service Quality

The following points measure the quality of this service:

• Silence suppression makes voice calls more efficient because bandwidth is used better by sending less data.
• Header compression, which uses RTP/RTCP standards.
• IPv6 gives a bigger address space and allows tunneling.
• Giving priority to packets that need less delay. Current trends:
  • CQ (Custom Queuing): Sets aside a part of the available bandwidth.
  • PQ (Priority Queuing): Gives priority in the queues.
  • WFQ (Weight Fair Queuing): Priority goes to the traffic with the least load.
  • DiffServ: Skips the middle routing tables and decides routes for each packet.

Wireless Networks

A mobile device needs a strong processor to run the VoIP system. Also, this processor handles wireless Internet. It uses connection types like Wi-Fi, HSDPA, WiMAX, or EV-DO rev A.

But EV-DO rev A works better for mobile phones. In short, this connection lets packets move faster. It allows quick packet transfer for both uplink and downlink.

Protocols

The system uses the SIP signaling protocol to start a call between two users. Also, it finds the users’ location and makes moving easier.

Also, the SIP protocol works with RTP. RTP carries voice packets. Then, it makes them digital and sends them with the IP protocol.

Mobility

When two users move and start a call, the coverage area changes.

In this case, the device must connect to a new base station. But this takes time because the connection must restart. Also, this is like the handover process in old mobile phones.

Time Delay

Besides the delay caused by switching coverage from a base station.

• Delays happen when the processor codes, compresses, and packs data to send over the IP network.
• Using the IP system can cause delays, such as losing packets or sending incorrect packets.
• Problems can happen because of full node queues or busy networks.

Applications

The processor must do many tasks to make a connection using the IP system. So, mobile devices are needed.

Smartphones have strong processors. Because of this, they can do the needed tasks. Also, we can make VoIP calls using apps like Skype.

After all, these mobile devices provide VoIP services well.

What is VoIPv6?

Voice over Internet Protocol Version 6 is also called VoIPv6. Old VoIP uses version 4 of the IPv4 system.

IPv4 fixed many network problems. But then new ideas came up: being everywhere, moving around, and wireless use.

This needed the start and use of IPv6. In short, IPv6 fits the fast growth of the Internet.

Evolution

VoIPv6 uses the TCP/IP system and carries traffic as data packets. It also handles all traffic types using IP.

Operators move many kinds of data over public networks. But as new tech like fiber offers huge capacity, the focus on one type of moving has grown.

Using new technologies has helped the industry grow a lot. But it is sent in packets by different data networks and is affected by QoS.

IPv6 is now a tech that helps TDM everywhere. This gives it functional parts like growth, reach, and high strength.

Third Generation VoIP Networks

The key features of Third Generation (3G) networks are:

• These are end-to-end IPv6-based networks.
• It gives full access to every user in the connected world.
• It is an advanced signaling system based on SIP.
• It works with business networks in terms of protocols and security.
• It has QoS support in wireless LAN networks.
• It works together with 3G mobile services.
• It gives excellent business service, is reliable, and is safe.
• It has end-to-end QoS support.
• It gives low-bit-rate support for video and video calls.
• It works with many different types of networks that support it.
• It works fully with other systems to support a real unified messaging setup.

How Does VoIPv6 Work?

A worldwide talking system needs a strong signaling network.

The system uses the SIP protocol when running VoIP on IPv6. It sets up, changes, and ends calls. Calls include phone, video, and meetings.

SIP uses IP invites to send call types and details. These details decide what a call needs to work.

Also, SIP asks for a way to find the user; also, SIP checks who is allowed to use services. So, the provider uses Proxy Servers to control calls.

Real-Time Communication Requirements

VoIP is the next step after old phone systems and is a very popular tool. Voice services are now used worldwide in many types of business.

Also, adding wireless movement and other new tech has sped up. This has made the user market safe for voice calls.

But live communication must hit certain marks for quality. In short, this makes users much less happy. A few of these needs are listed next.

1. We must have low jitter.
2. We need low latency.
3. The system must quickly adjust to shifting traffic and network states.
4. Big networks with many links require high speed.
5. There is a small need for memory space inside a network.
6. The use of capacity must be very effective.
7. Low header bits per packet are required.
8. The network and its endpoints must keep the data packet retransmission workload low.

Protocols Aiming at QoS Guarantee

Many rules try to ensure service quality. Some of these work inside the network.

For instance, RSVP works inside the network and fits with apps. Other rules work at the link stage.

MPLS is one example of these rules. Also, some rules add features for specific apps. For instance, the real-time protocol (RTP) is one.

1. RSVP (Resource ReSerVation Protocol)

We use RSVP in programs like Unicast and multicast. It also helps with one-way bookings started by the receiver.

It also manages temporary data in routers. RSVP offers various booking types and works with both IPv4 and IPv6.

2. MPLS (Multiprotocol Label Switching)

MPLS gives a set of services with good quality. It provides traffic with steady quality.

It also sets dynamic routes and does network tuning. Traffic planning orders traffic drops according to the need.

3. RTP (Real-Time Transport Protocol)

RTP works for live uses like calls and movies. It sends data over group or single links. But RTP does not save space and cannot promise good speed alone.

Frequently Asked Questions (FAQ) About VoIP

Conclusion

In short, the VoIP call system has changed a considerable amount. This tool has given cheap answers and better features. It has also given great freedom.

As tech grows, I see VoIP having a bigger key part in future calls. It changed how we link with its power to beat place limits and ease talking.

In the future, VoIP will still give speed and lower costs, as it has shown the strength of new ideas. I also feel it will stay the base part of today’s talking.