What Is a VOIP Dialer?

IP-based voice transmission (English: Voice over Internet Protocol, abbreviated as VoIP ) is a voice call technology that uses the Internet Protocol (IP) to achieve voice calls and multimedia conferences, that is, to communicate via the Internet. Other informal names are IP telephony, Internet telephony, broadband telephony, and broadband phone service.

Many friends like to use Internet chat tools for voice chat. This kind of voice is not transmitted through the traditional telephone network of telecommunications operators, but is transmitted through the Internet. This technology that converts voice into IP data packets, partly or entirely based on IP network transmission, is VoIP (Voice over IP).

The basic principle of VoIP is to compress the voice data by using a voice compression algorithm, then package these voice data according to the TCP / IP standard, send the data packets to the receiving place through the IP network, and string these voice data packets , After decompression processing

Many standards organizations and industrial entities around the world have proposed many speech coding schemes. Which includes

Unstoppable network communication, all kinds of instant messaging

Because of its low cost and broad market, VoIP has become a profiteering investment project. However, because there is no policy support, there are still loopholes in operation and management. The development prospect is still unclear, and investment needs to be careful.

After several years of development, though

Call quality is affected by the quality of the network

Unavailable during power failure

There is a gap between the definition and the traditional fixed telephone (under normal network conditions, there is no obvious difference between the call sound quality and the traditional telephone)

There is a risk of being overheard

Can change the number at will, which is easy to cause crimes (must be approved by the operator, users cannot change the number at will)

Commonly used protocols (Control Protocol) such as H.323, SIP, MEGACO and MGCP.

H.323 is an ITU-T standard. It was originally used for multimedia conferences on local area networks (LANs), and later extended to cover VoIP. The standard includes both

For multimedia applications on existing communication networks,

While enjoying the convenience that VoIP technology brings to our lives, let us also understand the development of VoIP:

1. Early VoIP was implemented through software on personal computers. At that time, VoIP phones could only make calls between PCs and PCs, and the call quality was not good. It was just regarded as an application of the Internet.

2. From 2000 to 2002, VoIP technology began to penetrate the telecommunications field. The emergence of telephone services based on IP networks. At this time, VoIP development was very rapid, and VoIP could already be implemented between PC-phone, PC-PC, and phone-phone. At this time, many VoIP phone service providers appeared, and the voice quality of VoIP is also constantly improving.

3. Since 2003, VoIP has evolved into broadband telephony. As a commercial service, VoIP has begun to compete with traditional fixed telephones. The voice quality of VoIP is similar to or even higher than that of traditional fixed telephones.

With the development of mobile Internet, VoIP technology also has new development trends:

• There has been a convergence of VoIP technology and wireless networks. The wireless network mainly includes 3G (The 3rd Generation Mobile Communications), Long Term Evolution (LTE), and Wireless Local Area Network (WLAN).

  Among them, the development of VoIP based on WLAN is particularly rapid. In wireless VoIP, the last access of the signal is WLAN, and the other parts still use the wired network for transmission.

• There has been a convergence of VoIP and P2P (Peer-to-Peer, end-to-end). P2P technology comprehensively utilizes scattered network resources, making the call rate and voice quality of voice calls even surpass the traditional telephone network.

The calculation method of VOIP's common coding bandwidth is related to the coding method selected, but has nothing to do with which manufacturer's, the formula is as follows:

Bandwidth = packet length x packets per second

= Packet length × (1 / packing cycle)

= (Ethernet header + IP header + UDP header + RTP header + payload) × (1 / packing cycle)

= (208bit + 160bit + 64bit + 96bit + payload) × (1 / packing cycle)

= (528bit + (packing cycle (seconds) × bits per second)) × (1 / packing cycle)

= (528 / packing period) + bits per second

According to various encoding methods, we get:

G711: 20ms packet, bandwidth is (528/20 + 64) Kbit / s = 90.4 Kbit / s

G729: 20ms packet, bandwidth is (528/20 + 8) Kbit / s = 34.4 Kbit / s

G723: 5.3k, 30ms packet, bandwidth is (528/30 + 5.3) Kbit / s = 22.9 Kbit / s

The industry generally designs the network bandwidth according to the IP network bandwidth coefficient and Ethernet bandwidth coefficient provided in the following table:

Codec compression rate (Kbps) Packing period (ms) IP network bandwidth coefficient Ethernet bandwidth coefficient

G.711 a / u 64 20 1.25 1.41

G.729 a / b 8 20 0.38 0.54

G.723.1 (5.3kbit / s) 5.3 30 0.27 0.37

G.723.1 (6.3Kbit / s) 6.3 30 0.25 0.36

H.263 (384Kbit / s) 384 10 6 6.2

Note: When using a certain encoding method, the actual bandwidth occupied can be obtained by multiplying 64K by the corresponding bandwidth coefficient. Of course, if it is a relay interface, you also need to consider that signaling occupies a certain bandwidth, which is generally calculated at 2.5%.

On March 17, 2012, the Ministry of Industry and Information Technology issued the "Notice on Follow-up Work on the Issuance of 3G Licenses".

1. No computer required. It will not be like a computer because of a virus or an operating system failure that prevents it from being used properly.

2. Suitable for guaranteeing long-term online. The key to fighting each other on the intranet is to stay online.

3. Encrypted communication.

4. The sound quality is clear. The voice processing adopts the international advanced voice processing chip, which can effectively solve the problems caused by echo cancellation, data packet loss and network jitter. Sound quality is generally higher than softphones.

5, the cost is low, you can change the number at will, display the number.

6, can monitor, record, support three-way communication, secret language, call transfer and other functions. ^[1]

There are few examples of VoIP attacks. Most VoIP phone systems are installed in enterprises, and the encryption technology of VoIP phones also uses general technology.

The security of V0IP is realized through Internet phone encryption, that is, voice data encryption. The current secret methods are TLS and IPSec.

The first step to take is to implement an authentication mechanism in your local area network, which includes devices and users. If you buy some 802.1x-like products, they are not sufficient, because most of the phones, printers, medical equipment, robots, and other equipment do not support 802.1x requirements. ^[1]