Basic SIP application solution

Basic SIP Application

As one of the main VOIP communication protocols, the SIP protocol is simple, flexible, and open, and is gradually dominant in the VOIP communication field. The main methods used for SIP Communication include SIP terminals, proxy/targeted servers, location servers, and PSTN gateways. Currently, the latest standard of the SIP protocol is RFC3261. Major network equipment providers can currently provide SIP Communication devices, such as CISCO. In WINDOWS, UNIX, and other operating systems, there are also many SIP software phones available, because we can see that SIP phones will become increasingly popular in the future.

The SIP protocol is a signaling control protocol. to form a complete communication system, media control protocols such as SDP and RTP and media communication protocols must be attached. SIP is responsible for establishing, maintaining, and releasing calls. SDP is responsible for media negotiation and control, and RTP is responsible for transmitting communication media.

Enterprise Network Conditions

The main application of VOIP should be enterprise networks. Therefore, we must make the current VOIP system well adapted to the enterprise network conditions in order to give full play to its role. After a simple analysis of the enterprise network, we will find that the enterprise network is generally an internal network. That is to say, on the enterprise network, the IP addresses used are all private addresses and communicate with the outside world, all are implemented through NAT. This is because of the lack of IP addresses, it is difficult for the enterprise to apply for a Global IP address with the ISP, and considering the internal security of the enterprise network, enterprises do not want external users to directly access all devices on the enterprise network. Therefore, internal network addresses can enable enterprises to maintain their own networks and communicate with the INTERNET through NAT. In the enterprise network, you can also map internal special servers to the INTERNET through NAT, so that users on the public network can directly access these servers. Such a network structure has a great impact on the application of VOIP in enterprise networks. For example, if a VOIP Terminal in an Enterprise Network wants to be accessed by the outside world, it is difficult to maintain the network by ing all the VOIP terminals to the outside world on the NAT gateway. At the same time, this ing has poor scalability. In NAT gateway, H323 and SIP application-level gateways must be implemented. Currently, most devices do not support this function, this means that enterprises are about to replace network devices. We think most enterprises are unacceptable for such applications.

SIP Enterprise Network Solution

Considering the application of SIP in the enterprise network, we should provide a means to make all the SIP terminals accessible to the outside, so that the application of VOIP in the enterprise network can be meaningful. That is to say, the SIP terminal that uses a private address in the enterprise network can be called by the external SIP terminal as the called address, and the IP address of the SIP terminal does not need to be mapped to the external address by NAT, their IP addresses can be obtained through DHCP or static configuration.

In this system, the enterprise network has NAT applications. If the enterprise network does not have NAT and the global network address is used, the system can also use it, the application mode is the same as that of NAT.

The system consists of a SIP terminal and a SIP server. Like a common SIP terminal, a SIP terminal can initiate or receive a call. It can communicate both inside and outside the enterprise network. It can be an IP Phone gateway or an IP Phone, IP Phone software. The SIP server is responsible for System Call proxy, registration service, location service, media service, and other functions. The premise of system operation is that the SIP server requires a global IP address ing, that is, the NAT gateway of the enterprise network can provide a global IP address ing to the server.

The implementation of the SIP terminal is consistent with that of General devices, but the SIP terminal must have an out-of-band server at the top of the Configuration. That is to say, all calls of the SIP must be completed through the SIP server, when initiating a call, you must first call the server. The end of the call is to notify the server that the call has ended. These operations fully comply with the IETF RFC3261 specifications and comply with standards in terms of protocol consistency. At the same time, in the subsequent SDP and RTP processes, the SIP terminal can perform relevant processing in full accordance with the protocol standards without any special processing.

The main function of the system is to use the SIP server. The work done by the SIP Server includes the RFC standard proxy server, registration server, and location server. The media exchange function will also be added to ensure that the system can send the internal network media information of the enterprise to the outside, and ensure the external and internal media exchange. Specific implementation, design to the company's technical secrets, will provide detailed solutions after the cooperation is successful. The SIP server can be implemented on WINDOWS or UNIX as an application.

The system mainly considers the following applications:

Internal Network calls external network

1) An internal SIP terminal initiates a call, which is sent to an internal SIP server.

2) The internal SIP server will complete locating and searching functions.

3) The internal SIP server sends a call to the external SIP terminal through the NAT gateway.

4) The external SIP terminal notifies the internal SIP server to call and establish a media channel with the internal SIP server.

5) The internal server notifies the internal SIP terminal to establish a call and establishes an internal media channel.

6) communication starts. The SIP server acts as a Media Exchange Server.

External network calls internal network

1) An external SIP terminal initiates a call to call an internal SIP server. in the sip uri, the user name can be specified as a SIP terminal in the internal network.

2) The internal SIP Server determines the internal SIP terminal address based on the URI and calls the internal SIP terminal.

3) The internal SIP terminal responds and establishes a call through the internal SIP server and the external SIP terminal.

4) The internal SIP Server establishes a media channel with the internal SIP terminal and the external SIP terminal respectively.

5) communication starts and internal SIP Server Media Exchange

Internal Network A calls internal network B

1) A network calls A network's SIP Server

2) locate the SIP server in Network A and call it to the SIP server in Network B.

3) Find and locate the SIP server in B network and call the SIP terminal in B network.

4) The SIP terminal of Network B responds and sends A reverse notification to the SIP terminal of Network.

5) communication is established. The SIP server of Network A and the SIP server of Network B establish A media channel and establish A media channel with the SIP terminal of their internal networks.

6) communication starts. The SIP servers in the and B networks run as Media Exchange servers.

Description

This system mentions the application of SIP in enterprise networks. It can adapt to enterprise networks in various situations and play a major role in promoting the popularization of VOIP in enterprise networks. At the same time, the system is designed with reference to relevant specifications and can be fully interconnected with standard equipment without any modification to the existing network structure. To all VOIP protocols, for H323 and MEGACO Protocol applications, the company can also provide corresponding solutions to solve the limitations of enterprise network applications.

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