Enterprise VoIP Communication Technology

VoIP Overview:

VoIP (Voice over Internet Protocol) is a communication device that uses an IP network to transmit Voice and save user calls. It is mainly suitable for enterprises and group users with branches, it can save enterprises a lot of international, domestic and suburban long-distance calls.

VoIP can digitize the voice over the Internet, compress the voice, transmit the voice in a packet type, decompress the package, and then restore the voice. In seconds, the call function is enabled, this will greatly reduce the expansion cost of the traditional one-way switching network and make more effective use of the existing data network and transmission backbone network. At present, the most commonly used VoIP technology is Internet telephone. Since VoIP can save a considerable amount of long-distance telephone fees for enterprises or individuals, and multinational enterprises have switched to VoIP due to low Telecom fees, many people are talking about VoIP, many service and equipment companies also want to participate in the development of VoIP.

VoIP benefits:

1. Billing: Saving telephone fees is a significant and immediate effect of VoIP, especially for long-distance or international telephone fees. It is also an important way to save operational costs for multinational enterprises.

2. Simplified: the integration of the call and data Open Network (Networks) allows more quasi-jobs and reduces the demand for hard devices. Families or enterprises do not need to set up two sets of Networks: voice and data transmission, all communications will be integrated in a single network.

3. Advanced: long-term benefits of VoIP include support for multimedia applications, which is not comparable to the existing traditional telephone system.

VoIP Communication Technology:

Currently, two mainstream communication technologies are used in VoIP: H.323 and MGCP. H.323 is developed by the International Telecommunications Union (ITU-T), Microsoft NetMeeting and other video conferencing software is to use H.323 for communication, has easy to manage the network architecture and high value; MGCP (Media Gateway Control Protocol) is developed by IETF (Internet Engineering Task Force), which is easier than H.323 in software and communication Protocol design. In addition, the next-generation network telephone communication protocols and architectures include MEGACO, SIP, and SIGTRAN.

The traditional Telephone Network (PSTN-Public Switch Telephone Network) is used to transmit voice in Telephone exchange mode. The required transmission bandwidth is 64bit/s, while the VOIP is based on the IP data Network as the transmission platform, digitize Analog voice signals, compress, package, and perform other special operations so that they can be transmitted using a connectionless UDP protocol. With the popularity of public Internet and high-end enterprises' internal networks in recent years, it has become imperative for corporate decision makers to use existing public Internet and internal data networks to reduce enterprises' long distance and international telephone fees.

I. Basic concepts of VoIP

VOIP is a segmentation and digital transmission technology based on IP technology. Its basic principle is to compress and encode voice data using a speech compression algorithm, then, package the voice data by IP address and other related protocols, transmit the data packets to the receiving location through the IP network, and then concatenate the voice data packets. After decoding and decompression, the original voice signal is restored to achieve the purpose of transmitting voice over an IP network.

The IP telephone system converts analog signals of ordinary phones into IP data packets sent by computers over the Internet, and also converts received IP data packets into analog electrical signals of sound. After the conversion and compression of the IP Phone System, the transmission rate of each common phone is approximately 8 ~ 11 kbit/s bandwidth. Therefore, when using a bandwidth with a transmission rate of 64 kbit/s, the number of IP phones is 5 ~ 8 times.

Ii. Basic principles of VoIP

VOIP is a segmentation and digital transmission technology based on IP technology. Its basic principle is to compress and encode voice data using a speech compression algorithm, then, package the voice data by IP address and other related protocols, transmit the data packets to the receiving location through the IP network, and then concatenate the voice data packets. After decoding and decompression, the original voice signal is restored to achieve the purpose of transmitting voice over an IP network.

The IP telephone system converts analog signals of ordinary phones into IP data packets sent by computers over the Internet, and also converts received IP data packets into analog electrical signals of sound. After the conversion and compression of the IP Phone System, the transmission rate of each common phone is approximately 8 ~ 11 kbit/s bandwidth. Therefore, when using a bandwidth with a transmission rate of 64 kbit/s, the number of IP phones is 5 ~ 8 times.

III. Basic VoIP transmission process

1. Speech-Data Conversion

2. Convert original data to IP Address

3. Transfer

4. IP package-Data Conversion

5. convert digital speech to analog speech

IV. Implementation of VoIP

In terms of implementation, there are four methods for VOIP: telephone to telephone, telephone to PC, PC to telephone, and PC to PC. Initially, the VOIP method was mainly from PC to PC, and the IP address was used for call. Through voice compression and packaging transmission, real-time voice transmission between PCs on the Internet was realized, voice compression, encoding/decoding, and packaging are all done through hardware resources such as processors, sound cards, and NICs on the PC. This method is very different from public telephone communication and is limited to the Internet, so there are great limitations. A call refers to a call that is connected to an IP Phone gateway through a telephone switch. A call is made through an IP network using a telephone number. The sender gateway identifies the caller and translates the phone number/gateway IP address, initiate an IP phone call, connect to the gateway closest to the called call, and complete the voice encoding and packaging. The Receiving Terminal gateway can unpack, decode, and connect to the called call. The gateway is used to correspond to and translate IP addresses and phone numbers, as well as voice coding/decoding and packaging.

V. Advantages of VOIP

Low communication fees

Low network rental and maintenance costs

Video conferencing, data storage and forwarding, fax, and streaming media

1. Basic concepts, principles, and application of VoIP technology
2. Application of server-bound Softswitch VoIP in Small and Medium-sized Enterprises