Implementation and network organization of Enterprise VoIP

1 Introduction

With the development of information technology, especially optical fiber technology and Internet technology, enterprises are able to build IP Broadband Networks for the integrated delivery of voice, data and images. How to Implement the transmission of existing voice services over IP networks has become an important issue for enterprises in network construction or transformation. VoIP provides a solution to this problem. Technically speaking, there are two basic implementation methods of VoIP: IP extension and VoIP gateway. The so-called IP extension method means that each phone in the voice network that can enjoy the IP service is a dedicated and independent IP phone, which itself is a small computer, most VoIP operations, such as voice encoding, compression, and packaging, are implemented by the phone device. The other is the VoIP gateway + Analog Phone mode, where the phone is a common phone, all VoIP operations are implemented through gateway devices. Currently, both methods are implemented Based on H.323, and the quality of IP speech is the same. Although the IP extension method has great advantages in Building Integrated Wiring and as a comprehensive business terminal, however, there are big technical gaps in signaling processing, CTI Technology Integration, and provision of standard open platforms. Few vendors support this technology, as well as the uninterruptible power supply problems that cannot be solved at present, so that their applications under the current conditions are greatly restricted; however, the VoIP gateway + analog phones are developed with mature technologies, saving investment, facilitating operation and maintenance, and taking full advantage of the existing voice system of enterprise users, this makes it the preferred choice for IP voice solutions for many domestic and foreign enterprises. This article mainly discusses the implementation of the internal VoIP network and the formulation of the network organization scheme of the VoIP gateway + analog phone.

2. Implementation of VoIP gateway + Analog voice system

The typical VoIP gateway + analog telephone voice system is generally composed of four parts: telephone terminal, Gateway GW), Multi-Point Access Control Unit MCU, and network guard, based on the protocol model defined by the H.323 protocol family. The Gateway provides signaling, H.323 protocol, voice codec, and routing protocol processing. The Gateway provides the relay interfaces connected to the PSTN and the interfaces connected to the IP network respectively; the main functions of the network guard are user authentication, address resolution, bandwidth management, security management, and regional management. In the case of a small network scale, the network guard is not a required device. This saves a lot of construction funds and makes the construction of an enterprise's IP address telephone network very easy.

After an enterprise user sends a call through a phone, the enterprise user connects to the gateway through a small switch or directly. The Gateway triggers the corresponding business process based on the call access code, and then interacts with the user to obtain the called number information; after obtaining the called number information, the gateway compares it with the route data set on the Gateway: if the number corresponding to the called number already contains the corresponding route data, the gateway directly establishes IP communication with the peer gateway. If no route data exists at the first part of the called number, the call fails. The Gateway prompts that the user cannot connect, and allow the user to call the called number again. After the call between two gateways is established, the called-side gateway will initiate a call establishment request to the PSTN switch or PBX where the called user is located or directly connect to the called extension). If the called number exists and is idle, The called phone will ring; after the called voice is detached, the master called user can call the IP address Telephone Gateway and the IP network. The voice is transmitted from the switch or directly) to the gateway, and the gateway compresses the voice, compress the compressed voice packet into an IP packet and send it to the peer gateway over the Internet/Intranet. Decompress the package, restore it to a compressed voice package, and decompress it to a normal voice package, so that users can call.

3. Enterprise VoIP network solution formulation

3.1 compatibility and transition to existing telephone networks of enterprises

Various networking solutions of VoIP gateway + analog phones can be compatible with existing telephone exchange networks of enterprises, and gradually transition from traditional telephone networks to IP networks. Add a voice gateway device to the existing or proposed data network. After the IP Voice Gateway is added, the original telephone network of the Enterprise will not be affected. Enterprise users' telephone calls can select outbound channels as needed, and IP calls can be made through the IP gateway, you can also access the PSTN through a traditional telephone call route. The same is true for the acceptance process. You can accept calls from the IP network or PSTN. There is no difference for users.

3.2 media Coding Technology Selection

Because the service quality in the group switching network cannot be well guaranteed, encoding requires a certain degree of flexibility, that is, variable and adaptive encoding speed. Widely used linear prediction CELP Based on the Principle G.729 and G.723G. 723.1) speech compression coding technology. G.729 was originally the 8 kbit/s speech coding standard. After further improvement, its work scope has been extended to 6.4 ~ 11.8 kbit/s, the voice quality has also changed in this range, but even 6.4 kbit/s, the voice quality is also very good, very suitable for use in VoIP systems, g.723.1 adopts 5.3/6.3 kbit/s dual-Rate Speech Encoding. The quality of speech is good, but the processing latency is large. It is the currently standardized lowest-Rate Speech Encoding Algorithm. G.723.1 is also widely used in H.323 conferencing systems when the access network speed is generally low.

The delay of the IP voice signal mainly comes from two aspects: the time when the voice is compressed and loaded into the data packet and the time when the voice data packet is transmitted online. According to a survey, the delay time for long-distance Internet communication decreases by 20% every year. Related speech technologies also include the mute detection technology and echo elimination technology. Research results show that about 50% of people call each other to listen to the silence time, and 10% to the silence time of a short pause. The mute detection technology can effectively remove silent signals and further reduce the bandwidth required by voice signals to around 3.5 kbit/s; echo Cancellation Technology uses digital filter technology to eliminate echo interference that has a great impact on the quality of calls and ensure the quality of calls. This is particularly important in a group network environment with relatively large latency.

3.3 determine the access method

Because enterprises have different application environments and conditions, the access methods and the number of connected users vary greatly. Select different access methods and devices based on different levels and application environments, it mainly takes into account the number of users and resizing capabilities.

For a large enterprise's central node, the number of users is large. You can select a dedicated voice gateway to connect to the data network through a high-end router and connect to the switch through E1 relay, provides IP phone access from dozens of channels to thousands of channels, supports the No.1 and PRI signaling methods, and works with other devices to provide the No. 7 signaling access. To facilitate management and intercommunication between organizations, you can use network guard for centralized management.

Branch users of medium-sized enterprises or large enterprises can use vro Voice Gateway. A modular router can directly support E1 voice access and connect to vswitches through E1 relay, you can select vrouters of different models and configure modules to support access from dozens to hundreds of IP phone users. In addition, branches can connect to the Headquarters over the Internet using VPN for VoIP communication.

For remote branches of small enterprises or large enterprises, if the number of users is small, you can use E & M Relay or AT0 relay to achieve IP address voice access, the modular router can support up to 30 E & M and AT0 relay interfaces. In addition, for smaller group users, the modular router can directly support AL0 user interfaces, A telephone can be directly mounted to a vro.

For a network that uses the telephone mode of a Telecom group user without a local telephone switch, the connection method is the same as the above, but the number cannot be set completely at the time of dialing, the number of the relay line provided by China Telecom is used as the access number, which is equivalent to dialing after dialing the switchboard.

3.4 relay connection

Most IP Phone Gateways provide a wide range of voice interfaces. For example, E1 digital relay, AT0 second-line loop relay, and E/M Relay are the most commonly used in PBX. Most of the digital relay support the No.1 and PRI signaling methods. In addition, some gateways can also directly provide POTS interfaces to directly connect to common phones, which makes it very convenient to use. Generally, the Telephone Gateway is connected to the IP backbone network through Ethernet. It can also be built into the router through the voice module to directly access the IP backbone network. For enterprise users who already have their own User Switch PBX), during the construction of the IP telephone network, These PBX will be retained and continue to undertake the user access function, the relay connections between these user switches will be completed through the IP backbone network.

IP Phones use IP packets to transmit voice information on the data network. Each packet contains the IP address of the called Gateway. According to these IP addresses, the network device sends the voice packet directly to the called Gateway. Therefore, there is no issue of Central Bureau transfer. Logically, online voice switches are in the same status. Therefore, the number of relay entries in the central bureau can also be calculated by the local bureau. You can retrieve 1/10 of the number of telephone users.

By adding an IP Phone gateway, you can easily create or upgrade a traditional telephone network to an IP phone network. As long as you accurately calculate the number of IP phone relay during the establishment of an IP phone network and make rational use of various technologies, you can build a high-quality IP phone network at a very small cost. This will bring great economic benefits to enterprise users.