Networking and Application of VoIP technology (1)

1 Introduction

At present, the telecom operation mode is facing a huge change. The traditional telecom business is undergoing profound changes in terms of concepts, technologies, businesses, investment, management and services. The new communication mode will be unified to the IP address. The VoIP (Voice over IP) uses the speech compression encoding technology, the grouping real-time transmission technology and the Control Protocol (RTP/RTCP) the technology that carries voice signals over the IP network to implement voice communication. The generation and use of VolP services breaks the old frame in which traditional telecom networks must transmit voice signals in TDM mode, reflecting the trend of Circuit Switching Network and group switching network convergence, it has become a pioneer in unified network technology and business support.

2 basic components of a VoIP network

2.1 Basic Components

Generally, the basic elements that constitute a VoIP network include Gateway (GW), network guard (GK), authentication and billing, and Integrated Access Management Center (CAMS), network management system, and terminal devices.

The gateway is located at the interface between the Public Telephone Network (PSTN/ISDN/GSM) and the IP network to complete the bridge task between the public telephone network and the IP network. Main functions include: telephone/fax signaling, media stream, management information, and synchronous signal conversion between networks on both sides; call Establishment and release on the PSTN/ISDN/GSM side, call establishment and release on the IP network side, and test on network QoS; support for multiple types of Speech Coding including ITU-TG.711, G.723 and G.729; realize ITU-TH.323, H.225, H.245, RTP, RTCP, ISUP, TUP, DSS1 and SS7 signaling. Provides leased line, Modem, ISDN, IP Phone, IP Fax, and WAP user access functions, and implements a large number of value-added services.

The Network Guard provides functions such as gateway registration management, route management, call control, and service control. It supports call services from gateways to gateways and from PCs to gateways, you can plan, monitor, and manage logged-on network phone terminals so that the entire VoIP network system can operate normally, ensuring the security and quality stability of the network phone.

The authentication billing and Integrated Access Management Center uses a large relational database system to connect to the network guard through the IP network, and uses the RADI [JS protocol to accept the user access (roaming) authentication request initiated by the Network guard for identity authentication, it is also responsible for receiving the user billing information collected by the billing collection point and generating the communication fee bill.

The network management system manages the entire network system, reducing the difficulty of network management and improving the efficiency of network operation management. Provides configuration management, fault management, performance management, accounting management, security management, and user management for VoIP networks. Terminal devices usually include network phones that are directly connected to the IP network, visual phones, and general telephones that are connected to the IP network through small VoIP gateways. These terminals comply with the H.323/SIP protocol system.

2.2 voice signaling for VoIP networks

The VoIP network includes multiple types of voice signals, such as H.323, SIP, MGCP, and MeGaCo/H.248, which have a competitive relationship.

H.323: Developed by the ITU-T, it defines protocols and procedures for multimedia communication without QoS Assurance over the Internet or other group networks. H.323 is a protocol family, including H.225, H.245, H.450, T.120, H.235, H.261, G.711, G.723, and G.729. The architecture of H.323 consists of H.323 terminal, gateway, gatekeeper and multi-point control unit MCU.

SIP: the text-based application layer control protocol proposed by IETF, which establishes, adjusts, and terminates multimedia calls and sessions. The protocol is simple and easy to implement.

MGCP: IETF proposes the protocol between the media gateway and the Media Gateway controller. It improves the disadvantages of the Application of H.323 on VoIP.

MeGaCo/H.248: media gateway control protocol developed by IETF and ITU-T for communication between media gateway controllers and media gateways. MeGaCo/H.248 will gradually replace MGCP.

At present, due to the maturity of the H.323 Protocol, many H.323 Protocols are used in China, but their support for IP services is weak, which has defects in flexibility and scalability. SIP is simple and flexible. It is closely integrated with IP addresses and has strong business capabilities. However, the Protocol is not complete, the interconnectivity between vendors is poor, and there are few practical applications. Most vendors in the industry still recommend using a VoIP solution based on the H.323 protocol system.

3. Networking and Application of VoIP

3.1 VoIP network organization and Application

The organization of the VoIP network can determine the network type based on the network coverage and user group size. In practice, you can build a network using a peer-to-peer regional structure, a second-level or multi-level tree structure. Here, only the secondary tree structure is used to describe the network organization and Application of VoIP.

The level-2 tree structure usually includes the level-1 node, level-2 node, and the user terminal that connects to each node. A level-1 node is located at the top of the entire structure and is responsible for scheduling and management of calls between different level-2 areas. The second-level node is responsible for the communication, scheduling and management of VoIP phone terminals and PSTN contacts in the region, and sends calls that require cross-region or cross-Network Communication to the first-level node. Here, the second-level node is the main body that implements the functions of the local region. It establishes a connection with the PSTN and the user terminal through the gateway. Figure 1 shows the components of the VoiP network and the hierarchical network structure.

In terms of device configuration, the level-1 node is configured with a carrier-level or enterprise-level VoIP gateway connected to an IP network, and the level-2 node and branch unit are configured with small and medium VoIP gateways. The primary node is configured with a network guard, and the network guard is organized in the primary standby mode to ensure its reliability. Select whether to configure the secondary node as needed. If the number of connected devices supported by the node is small, the network guard can use a dedicated hardware platform to ensure its reliability and reduce the maintenance difficulty. Configure the network management server at the level 1 node, and configure the network management terminal at the level 1 and level 2 nodes.