What is VoIP (Voice over IP – Internet Protocol)?

VoIP sends voice signals over the Internet using IP protocols. This has revolutionized the way we communicate. In this article, I will clarify the basics of VoIP technology. I will also examine its network architecture, considering its advantages and functionality.

In short, I will analyze the protocols used in VoIP and their impact on international calls. In conclusion, we will reveal the intricacies of this transformative technology. Let’s explore the world of voice communication and its role in modern communications!

What is VoIP Protocol in Voice Communication?

The VoIP protocol is a protocol that transmits voice signals over the Internet using the IP protocol. This technology transmits voice data via IP protocols. As a result, voice signals can travel over the Internet. This protocol is a vital resource that makes voice communication possible.

The voice signal is sent in digital packets instead of traditional telephone lines. This is done digitally, not over the circuits used for telephony. Mobile VoIP is a voice-over IP protocol used on mobile phones. In addition, in communication between two mobile terminals, the voice signal is transmitted over the Internet. This occurs thanks to the IP protocol.

With recent developments, mobile terminals are now connected to the Internet via wireless networks. These connections provide operations such as surfing the Internet, sending and receiving e-mail, and downloading data. In addition, it supports applications such as instant messaging and sending voice packets.

The protocols used to send voice signals over the IP network are also known as VoIP. These protocols are commercial implementations of the ARPANET’s “Experimental Voice Protocol Network.”

VoIP allows voice traffic to travel over IP networks, such as local area networks. It is also essential to distinguish between Voice over IP (VoIP) and Telephony over IP. VoIP protocol refers to the technology that enables voice communication via IP. In contrast, telephony over IP is a public telephone service using VoIP technology.

VoIP Advantages

The main advantage of this service is that it avoids high phone charges. It is especially advantageous for long-distance calls. In addition, carrying voice and data on the same network with VoIP reduces costs. VoIP to VoIP calls are usually free. However, VoIP to PSTN calls cost the VoIP user.

Using the IP structure provides excellent savings on international calls. Only a fixed internet fee is paid, and this includes anyone with access to an IP connection.

As this technology progresses, mobile VoIP usage will increase. The main reason is that the call duration does not increase the cost. In addition, you can make global calls for the same price.

The development of codecs for the VoIP protocol allows for smaller data packets. This requires lower bandwidth. With the advancement of ADSL connections, this type of communication has become popular. It is especially preferred for international calls.

There are two types of PSTN-VoIP services: Direct Inward Dialing (DID) and Access Numbers. Access Numbers require the VoIP user to enter their extension number. The Direct Inward Dialing (DID) feature connects the caller directly to the VoIP user. Access Numbers are charged as a local call to the PSTN caller. Such rates are much cheaper than those of regional operators.

VoIP Protocol Functionality

It simplifies tasks that are difficult with traditional telephone networks. For example, local calls are automatically routed to the VoIP phone, regardless of where you are connected to the network.

You can also carry a VoIP phone when you are traveling and make calls anywhere you have an Internet connection.

In addition, toll-free numbers with VoIP are available in countries such as the US and the UK. Call center representatives using these phones work efficiently anywhere with an Internet connection.

However, some packages include services that PSTN typically charges extra for. For example, services such as three-way dialing, callback, automatic redial, and caller ID are not available in some countries.

As a result, the VoIP protocol provides these services to users, providing more flexible and economical communication.

Network Architecture

The standard defines three essential elements in its structure:

• Terminals: They are used to replace existing telephones. They can be used in both software and hardware.
• Gatekeepers: They are the center of the entire VoIP organization and will replace the existing center. Generally, in software, if any, all communication goes through this software.
• Gateways: This is the connection to the traditional telephone network and acts transparently for the user.

With these three elements, the structure of the VoIP network can be a connection between two branches of the same company. The advantage is immediate: all communication between the delegations is entirely accessible. The same scheme can be applied to suppliers, with the resulting savings.

VoIP protocols: It is the language that different devices will use for their connections. This section is essential since the efficiency and complexity of the communication will 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 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 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 standardfrom VozTelecom

VoIP has been found to have a number of advantages for both businesses and casual users.

What are VoIP Parameters?

This is the main problem faced by both VoIP and all IP applications today. Providing quality service over the Internet is difficult because of its “best effort” structure.

In addition, bandwidth limitations on the route also affect this situation. As a result, various problems arise in ensuring quality of service. Therefore, users are affected by the instability of the service.

Codec

Encryption is required for voice transmission over an IP network. Codecs encode and compress audio or video. This process is necessary for subsequent decoding and decompression. Also, different amounts of bandwidth are used depending on the codec used. Usually, the quality of the transmitted data is directly proportional to the bandwidth.

In order to send audio over IP, it needs to be digitized. The following codecs are used for this process: G.711, 722, 723, and G.728. These codecs reduce bandwidth by compressing the audio data. As a result, they allow more connections on the same channel.

These Codecs have the following dimensions on their signboard:

• 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 does not mean that bandwidth is used; for example, G729 Codec uses 31.5 Kbps bandwidth in its transmission.

Service Quality

The quality of this service is achieved according to the following criteria:

• Silence suppression provides greater efficiency when performing voice transmissions because bandwidth is better utilized by transmitting less information.
• Header compression, which implements RTP/RTCP standards.
• Ipv6 implementation provides a larger address space and tunneling capabilities.
• Prioritization of packets that require less delay. Current trends:
  • CQ (Custom Queuing): Allocates a percentage of available bandwidth.
  • PQ (Priority Queuing): Sets priority in queues.
  • WFQ (Weight Fair Queuing): Priority is assigned to the least loaded traffic.
  • DiffServ: Avoids intermediate routing tables and makes route decisions on a per-packet basis.

Wireless Networks

A mobile terminal needs a powerful processor to use the VoIP protocol. In addition, this processor supports wireless connection to the Internet. It also uses connection options such as Wi-Fi, HSDPA, WiMAX, or EV-DO rev A.

However, EV-DO rev A is more suitable for mobile phones. In short, this type of connection allows faster packet transmission. It will enable fast packet generation for both uplink and downlink.

Protocols

SIP signaling protocol is also used to establish a connection between two users. Also, this protocol determines the location of the users and increases mobility.

In addition, the SIP protocol is complemented by RTP. RTP carries voice data packets. Then, it digitizes and packages these packets with IP protocol.

Mobility

When two users are moving and establishing a connection, the coverage area changes.

In this case, it is necessary to connect to a new base station. However, this takes some time as the connection needs to be reestablished. Also, this process is similar to the handover process in traditional mobile phones.

Time Delay

In addition to the delay introduced by changing coverage from a base station,

• Delays occur when the processor encodes, compresses, and packages data to be sent over the IP network.
• Delays are related to using the IP protocol, such as packet loss and sending incorrect packets.
• Issues may arise due to full node queues or network congestion.

Applications

The processor needs to perform many operations to establish a connection via IP protocol. Therefore, mobile terminals are necessary.

Smartphones are equipped with powerful processors. Thanks to this, they can perform the necessary operations. In addition, we can also make VoIP transfers through applications such as Skype.

After all, such mobile terminals provide VoIP services effectively.

What is VoIPv6?

Voice over Internet Protocol Version 6, also known as VoIPv6. Traditional VoIP uses version 4 of the IPv4 protocol.

IPv4 solved many network-level challenges. However, new concepts emerged: ubiquity, portability, and wireless mobility.

This necessitated the introduction and deployment of IPv6. In short, IPv6 adapts to the rapid growth of the Internet.

Evolution

VoIPv6 benefits from the convergence of TCP/IP architecture and supports traffic in the form of data packets. It also manages all types of traffic by utilizing IP.

Operators carry different traffic and services over public traffic networks. However, with fiber optics and other technologies offering large bandwidth, the focus on a single type of transport has increased.

The adoption of new technologies has provided significant support for industry developments. However, it is managed in packets by various data transmission networks and is affected by QoS.

IPv6 is becoming a technology that supports TDM worldwide. This offers features such as scalability, accessibility, and high robustness.

Third Generation VoIP Networks

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

• These are end-to-end IPv6-based networks.
• It provides full access to any user in the interconnected world.
• It is advanced SIP-based signaling.
• It is integrated with enterprise business networks in terms of protocols and security.
• It is QoS-enabled in wireless LAN environments.
• It is integrated with 3G cellular services.
• It provides high commercial quality service levels, reliability, and security.
• It is end-to-end QoS enabled.
• It provides low-bit rate support for video and video conferencing.
• It adapts to the high independence of the network type that supports it.
• It provides full integration with other environments to support an actual unified messaging environment.

How Does VoIPv6 Work?

A network that provides global communication services requires a robust signaling system. In VoIPv6 deployment, the SIP signaling protocol comes into play. This protocol establishes, modifies, and terminates sessions. Sessions include Internet telephony, multimedia distribution, and conferences.

SIP carries media types and session parameters using IP invitations. These parameters determine the resources required for a particular communication.

In addition, SIP requests a route to the user’s location. In addition, SIP authenticates and authorizes different types of services. As a result, the provider uses Proxy Servers to implement call policies.

Real-Time Communication Requirements

VoIP protocol is the natural successor to traditional telephony and is one of the most demanded applications. Voice services have achieved high global penetration in many business sectors. In addition, the introduction of wireless mobility and other advanced technologies has accelerated. This has made the consumer market safe for voice traffic.

However, real-time communication must meet specific parameters for service quality. In summary, user satisfaction is seriously affected. Some of these requirements are listed below.

1. We must have low jitter.
2. We need low latency.
3. Dynamic adaptability to changing traffic and network conditions is required.
4. Good performance is necessary for large networks and large numbers of connections.
5. Modest requirements for buffers within a network are required.
6. Capacity utilization must be very efficient.
7. Low header header bits per packet are required.
8. Low packet processing redundancy is required within the network and at the end of the system.

Protocols Aiming at QoS Guarantee

Various protocols aim to guarantee QoS. Some of them are at the network level.

For example, RSVP works at the network level and is compatible with applications. Other protocols operate at the link level.

MPLS is an example of such a protocol. Additionally, some protocols provide application-specific functionality. For example, real-time protocol (RTP) is an example.

1. RSVP (Resource ReSerVation Protocol)

We use RSVP in applications such as Unicast and multicast. It also supports receiver-initiated reservations with Simplex or one-way.

It also handles non-persistent states in routers. RSVP provides different reservation styles and supports IPv4 and IPv6.

2. MPLS (Multiprotocol Label Switching)

MPLS provides connection-oriented QoS support. It offers traffic contracts with reliable QoS.

It also defines dynamic routes and performs network optimization. Traffic engineering plans traffic deliveries based on demand.

3. RTP (Real-Time Transport Protocol)

RTP is suitable for real-time applications such as voice and video. It provides transmission over multicast or unicast services. However, RTP does not allocate resources and does not guarantee QoS by itself.

Conclusion

In summary, the VoIP protocol has transformed communication to a great extent. This technology has provided cost-effective solutions and advanced functionality. It has also offered unmatched flexibility.

As technology evolves, I expect VoIP to play a more critical role in the future of telecommunications. It has changed the way we connect with its ability to overcome geographical barriers and facilitate communication.

In the future, VoIP will continue to provide efficiency and cost savings as it has proven the power of innovation. I also believe it will continue to be the cornerstone of modern communications.