Voice over Internet Protocol

Typical VoIP-based solutions.

A typical analog phone adapter or ATA, to connect an analog phone to a VoIP network.

Avaya 1140E VoIP Phone.

Voice over Internet Protocol or Voice over Internet Protocol, also called Voice over IP, IP Voice, vozIP or VoIP (acronym for Voice orver IP: 'voice over IP'), is a set of resources that make it possible for the voice signal to travel over the Internet using the IP protocol (Internet Protocol). This means that the voice signal is sent digitally, in data packets, instead of being sent in analog form through circuits usable only by conventional telephony, such as PSTN networks (abbreviations of Public Switched Telephone Network, public switched telephone network).

The internet protocols used to send voice signals over the IP network are known as voice over IP protocols or IP protocols. These can be seen as commercial applications of the "experimental voice protocol network" (1973), invented by the ARPANET.

Voice over IP traffic can circulate over any IP network, including those connected to the Internet, such as LANs (local area networks: local area networks).

It is very important to differentiate between voice over IP (VoIP) and telephony over IP.

• VoIP is the set of rules, devices, protocols—in short, the technology—that allows voice to be transmitted over the IP protocol.
• The phone on IP is the telephone service available to the public, therefore with number E.164, made with VoIP technology.

Elements

Customer

The client establishes the voice calls, this information is received through the user's microphone (information input) it is encoded, packed and, in the same way, this information is decoded and reproduced through the speakers or headphones (information output).

A customer can be a Skype user or a user of a company that sells its IP telephony services through equipment such as ATAs (Analog Telephone Adapters) or IP phones or Softphones, which is software that allows you to make calls through a computer connected to the Internet.

Servers

Servers are responsible for managing database operations, performed in real time as well as offline. These operations include accounting, collection, routing, service administration and control, and user registration.

Usually software called Switches or IP-PBX (IP switches) is installed on servers, examples of switches can be "Voipswitch", "Mera", "Nextone&# 3. 4; among others, and IP-PBX can be those of Alcatel-Lucent, Cisco or Avaya in commercial brands and open source Asterisk.

Gateways (gateway or gateway)

The gateways (gateways or gateways) provide a communication bridge between all users, their main function is to provide interfaces with the appropriate traditional telephony, which will function as a platform for virtual users (clients).

The gateways are used to terminate the call, that is: the client originates the call and the gateway ends the call, that is when a client calls a landline or cell phone, there must be the party that makes it possible for that call who comes through the internet manages to connect with a client of a landline or cell phone company.

Network architecture

The standard itself defines three fundamental elements in its structure:

• terminals: are the substitutes for the current phones. They can be implemented in both software and hardware.
• gatekeepers: they are the center of the entire VoIP organization, and are the substitute for the current centers.

Normally they implement by software, if it exists, all the communications that go through it.

• gateways: it is the link with the traditional telephone network, acting transparently for the user.

With these three elements, the structure of the VoIP network could be the connection of two branches of the same company. The advantage is immediate: all communications between delegations are completely free. This same scheme could be applied to suppliers, with the consequent savings that this entails.

A voice, when digitized, can be transmitted on any packet network. Below is an image, interpreted with the Wireshark tool, of a flow of voice over IP protocol.

• VoIP protocols: are the languages that the different VoIP devices will use for connection. This part is important as it will depend on the effectiveness and complexity of communication.
  • By antiquity order (from older to newer):
    • H.323: protocol defined by ITU-T;
    • IAPA: protocol defined by IETF;
    • Megaco (also known as H.248) and MGCP: control protocols;
    • UNIStim: protocol owned by Nortel(Avaya);
    • Skinny Client Control Protocol: protocol owned by Cisco;
    • MiNet: protocol owned by Mitel;
    • CorNet-IP: protocol owned by Siemens;
    • IAX: Original protocol for communication between PBXs Asterisk (It is a standard for other data communications systems,^{[chuckles]required]} is currently in version 2, IAX2);
    • Skype: peer-to-peer proprietary protocol used in Skype application;
    • IAX2: protocol for communication between PBXs Asterisk in replacement of IAX;
    • Jingle: open protocol used in XMPP technology;
    • SCCP: proprietary protocol Cisco;
    • weSIP: protocol free voice license Telecom.

As we have seen, VoIP presents a large number of advantages, both for companies and for common users. The question would be why hasn't this technology been implemented yet? Below we will analyze the apparent reasons why VoIP has not yet prevailed over conventional telephony.

VoIP parameters

This is the main problem presented today by the penetration of both VoIP and all IP applications. Guaranteeing quality of service over the internet, which only supports "best effort" (best effort) and may have bandwidth limitations on the path, is currently not possible; For this reason, various problems arise in terms of guaranteeing the quality of the service.

Codecs

Voice must be encoded in order to be transmitted over the IP network. For this, codecs are used that guarantee the encoding and compression of the audio or video for its subsequent decoding and decompression before being able to generate a usable sound or image. Depending on the codec used in the stream, more or less bandwidth will be used. The amount of bandwidth used is usually directly proportional to the quality of the data transmitted.

Among the most widely used codecs in VoIP are G.711, G.723.1 and G.729 (specified by the 'ITU-T').

These codecs have the following encoding bandwidths:

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

This does not mean that it is the bandwidth used, since the traffic added by the lower layers of the TCP/IP protocol must be added. For example, the G729 codec uses 31.5 kbps of bandwidth in its transmission.

Delay or latency

Once the transit delays and the processing delay have been established, the conversation is considered acceptable below 150 ms (which is 1.5 tenths of a second) and would already produce significant delays.

Loss of frames (frames lost): during their journey through the IP network, frames can be lost as a result of network congestion or data corruption. Also, for real-time traffic like voice, retransmission of lost frames at the transport layer is impractical because it causes additional delays. Consequently, voice terminals have to retransmit with lost speech samples, also called Frame Erasures. The effect of lost frames on voice quality depends on how terminals handle Frame Erasures.

In the simplest case if a voice sample is lost the terminal will leave a gap in the voice stream. If many frames are lost, it will sound crackly with missing syllables or words. One possible recovery strategy is to play the previous voice samples. This works well if only a few samples are missing. To better combat burst errors, interpolation systems are usually used. Based on previous speech samples, the decoder will predict lost frames. This technique is known as packet loss concealment (PLC).

ITU-T G.113 Appendix I provides some provisional planning guidance on the effect of frame loss on voice quality. The impact is measured in terms of Ie, the deterioration factor. This is a number in which 0 means no deterioration. The larger value of Ie means more severe deterioration. The following table is derived from G.113 Appendix I and shows the impact of lost frames on the Ie factor.

Quality of service

In order to improve the level of service, the aim has been to reduce the bandwidth used, to which end work has been carried out under the following initiatives:

• The removal of silences gives more efficiency when it comes to a voice transmission, as the bandwidth is better used by transmitting less information.
• Compression of headers using RTP/RTCP standards.

For the measurement of the quality of service QoS, there are four parameters such as bandwidth, time delay (delay), delay variation (jitter) and packet loss.

To solve this type of problem, a network can implement three basic types of QoS:

• Delivering better effort (best effort): this method simply sends packages as you receive them, without applying any real specific task. That is, you have no priority for any service, just try to send the packages the best way.
• Integrated services: This system has as its main function to pre-agre a path for data that need priority, plus this architecture is not scalable, due to the amount of resources you need to be booking the bandwidths of each application. RSVP (resource reservation protocol) was developed as the mechanism to program and reserve the bandwidth required for each of the applications that are transported by the network.
• Different services: This system allows each network device to have the ability to handle the packages individually, plus each router and switch can configure their own QoS policies, to make their own decisions about the delivery of the packages. The differentiated services use 6 bits in the IP header (DSCP: Differentiated Services Code Point). The services for each DSCP are as follows:

• Prioritization of packages that require less latency. Current trends are:
  • PQ (Priority Queueing): This prioritization mechanism is characterized by defining 4 queues with high, medium, normal and low priority, In addition, it is necessary to determine which packages are to be in each of these queues, however, if they are not configured, they will be assigned by default to the normal priority. On the other hand, while there are packages in the upper tail, no medium-priority package will be served until the upper tail is empty, as well as for other types of tail.
  • WFQ (Weighted fair queuing): This method divides traffic into flows, provides a number of bandwidth just to the active flows on the network, the flows that are with little volume of traffic will be sent faster. That is, WFQ prioritizes those lower volume applications, these are associated as more sensitive to delay (delay) Like VoIP. On the other hand, it penalizes those that do not associate as real-time applications like FTP.
  • CQ (Custom Queueing): This mechanism assigns a percentage of bandwidth available for each type of traffic (voice, video and/or data), plus specific the number of packages per tail. The tails are served according to Round Robin (RR). The RR method assigns bandwidth to each of the different types of traffic existing on the network. With this method it is not possible to prioritize traffic since all queues are treated in the same way.

• The implementation of IPv6, which provides greater directional space and the possibility of tunneling.

Standards

H.323

Defined in 1996 by the ITU (International Telecommunications Union) it provides the various manufacturers with a series of standards so that they can evolve together.

Main features

Due to its structure, the standard provides the following advantages:

• It allows to control the traffic of the network, so the possibilities for significant drops in performance are diminished. IP-supported networks have the following additional advantages:
• It is independent of the kind of physical network that supports it. It allows integration with current large IP networks.
• It's independent of hardware used.
• Allows to be implemented both in software as in hardwarewith the particularity of hardware would mean removing the initial impact for the common user.
• Allows the integration of Video and TPV.
• It provides a link to the traditional phone network.
• This phone has evolved so much, that up to 800's that are non-geographic numbers, they can call an IP line.
• What was formerly a very infrastructure telephone center, is now summed up in installable software on a small server with the same features.

Session initiation protocol

The "login protocol" (Session Initiation Protocol, 'SIP') is a recent protocol that is widely used today.

VoIP is not a service, it is a technology

In many countries around the world, IP has generated multiple disagreements, between the territorial and the legal about this technology, it is clear and it should be clear that VoIP technology is not a service as such, but a technology that uses the protocol network (IP) through which data packets or datagrams are highly efficiently compressed and decompressed, to allow communication of two or more clients through a network such as the Internet network. With this technology, Telephony or Videoconference services, among others, can be provided.

We should not confuse VoIP (Voice over IP) with VoLTE (Voice over LTE). Even in fourth-generation (4G) mobile phone networks, Not all of them use VoLTE, since in many cases, 4G is used for data, but when making a call, the voice channel uses the 3.5G technology known as VoLGA. In the image below, you can see this method of operation.

Voice over IP is the technology that allows you to send and receive "information packets" that contain voice whose origin was analog, but that has gone through the process of: Sampling - Quantification and Coding, in such a way that it has been "digitized" to be treated like any other packet, except that "VoIP" can make use of the 6 bits of the IP header called "Differentiated Services", all this without being able to offer a guarantee of delivery and reception when it transits outside the network itself.

In the case of VoLTE, the Telephone Operator is responsible for organizing all the service, devices, platforms and networks so that it does guarantee from end to end the values of Differentiated Services that are established according to 3GPP.

Functionality

VoIP can facilitate tasks that would be more difficult to perform using ordinary telephone networks:

• Local phone calls can be automatically routed to a VoIP phone, no matter where it is connected to the network. One could take a VoIP phone with you on a trip, and anywhere connected to the internet, you could receive calls.
• Free VoIP phone numbers are available in the United States of America, the United Kingdom and other countries with VoIP user organizations.
• Call center agents using VoIP phones can work anywhere with internet connection fast enough.
• Some VoIP packages include extra services for which PSTN (commuted public phone network) normally charges an extra charge, or which are not available in some countries, such as 3 calls at a time, callback, automatic demarcation, or call identification.

Mobile

VoIP users can travel anywhere in the world and still make and receive calls by:

• Telephoneline service subscribers can make and receive local calls outside their locality. For example, if a user has a phone number in New York City and is traveling through Europe and someone calls their phone number, it will be received in Europe. Also, if a call is made from Europe to New York, it will be charged as a local call, of course the travel user in Europe must have an internet connection available.
• VoIP-based instant messaging users can also travel anywhere in the world and make and receive phone calls.
• VoIP phones can be integrated with other services available on the internet, including videoconferencing, data exchange and messages with other services in parallel with conversation, audio conferences, address book management and information exchange with others (friends, colleagues, etc.).

Advantages

The main advantage of this type of service is that it avoids the high telephone charges (mainly long distance) that are usual for the public switched telephone network (PSTN) companies.

The development of codecs for VoIP (aLaw, G.729, G.723, etc.) has allowed voice to be encoded into smaller and smaller data packets. This leads to the fact that voice over IP communications require very low bandwidths. Along with the permanent advance of ADSL connections in the residential market, this type of communication is becoming very popular for international calls.

There are two types of PSTN to VoIP service:

1. Direct incoming marking' (Direct Inward Dialling: DID'): connects the person who makes the call directly with the VoIP user, while the access numbers require that it enter the extension number of the VoIP user.
2. Access numbers: they are usually charged as a local call for who made the call from the PSTN and free for the VoIP user.

Disadvantages

• Call quality: it's a bit lower than the phone, since the data travel in the form of packages, that's why you can have some information losses and delay in transmission. The problem in itself is not the protocol but the IP network, as this was not intended to give such guarantees. Another disadvantage is the latency, since when the user is talking and another user is listening, it is not appropriate to have 200ms (milisegundos) of pause in the transmission. When you are going to use VoIP, you must control the use of the network to ensure a quality transmission.
• Data stealing: a cracker may have access to the VoIP server and the stored voice data and the phone service itself to listen to conversations or make free calls by users.
• Virus in the system: in case a virus infects any equipment from a VoIP server, the phone service may be interrupted. Other equipment connected to the system may also be affected. Specialized ID and deception suplantations. If one is not well protected, they may suffer fraud by supplanting identity.

Impact on trade

Voice over IP is making international communications cheaper and therefore improving communication between providers and customers, or between branches of the same group.

Likewise, voice over IP is being integrated, through specific applications, in web portals. As well as applications compatible with Android and iPhone. In this way, users can establish that a specific company calls them at a specific time, which will normally be done through an IP voice operator.

The growing bandwidth worldwide, and the optimization of layer 2 and 3 equipment to guarantee the QoS (Quality of Service) of voice services in real time makes the The future of voice over IP is very promising.

In the United States, voice-over-IP providers such as Vonage gained significant market share. In Spain, thanks to flat voice rates, conventional operators managed to avoid the massive landing of these operators. However, the expansion of this technology is coming from system developers such as Cisco and Avaya that integrate data and voice networks into their platforms. Other exchange manufacturers such as innovaphone, ShoreTel, Panasonic, Alcatel-Lucent, Nortel Networks, Matra, Samsung and LG also develop corporate voice over IP solutions in their private telecommunications equipment.

For international corporations that can count on state-of-the-art systems and optimum bandwidth, VoIP-handling exchanges (IPPBXs) have become a highly desirable piece of equipment. But small and medium-sized companies must evaluate certain issues: This technology operates with operating systems (Windows/Linux) that present certain stability problems. Furthermore, the IP network was not designed to provide guarantees. In addition, some suppliers to lower costs offer central units assembled in a computer or a PC, which face other types of problems, such as failures in their components (Hard Drives, Fans and Power Supplies), it is also necessary to anticipate the change of the traditional telephone sets, since this technology works with special telephones (IP or SIP) unless special equipment is incorporated.

The good news is that all the extra functions that IP PBXs can provide you can be obtained with your traditional PBXs, you only have to connect certain modules that incorporate VoIP technology to your needs. We all know that the transmission quality of traditional exchanges is still superior. In reality, we are already used to the reliability and easy configuration of traditional equipment, which handles very simple programming languages.