Design and Implementation of an IP Telephone System Based on the SIP protocol

In addition to its advantages of low cost and high utilization of network resources, the network IP phone can further integrate multimedia information, including voice, images, and data, to achieve interactive real-time communication, it has great development potential and gradually replaces the traditional PSTN.
Telephone trends have become the main form of voice information transmission in NGNNext General Network.
Currently, the signaling protocols used to construct the IP telephone system structure mainly include the H.323 protocol and the SIP protocol, which are completely parallel and incompatible. The H.323 protocol is proposed for Multimedia Conferencing Systems. The Protocol adopts the tedious signaling concept of traditional telecom networks and is very large. It is complex in terms of technical implementation, usage, and management methods. The SIP protocol is based on the existing Internet Protocol to construct the application layer protocol of the IP telephone service network. It pushes the complexity of network devices to the edge of the network and supports unicast, multicast communication, name ing, and redirection services, it also supports implementation of telecom services such as call forwarding and call rejection, and supports user mobility. Compared with H.323, SIP is more suitable for smart user terminals. It is more flexible to use and easier to master.
In view of this, this paper proposes a client-server model that complies with the SIP protocol specifications of the IP Phone System Design implementation scheme. The system has the following features:
(1) It adopts the IP-IP communication mode;
2) The telephone terminal device is directly connected to the USB interface of the user machine to facilitate reliable transmission of text, voice, and other data with the callee;
3) The system includes a complete user data management system and network call management control system;
4) high QoS is provided.
The system makes full use of the SIP protocol and provides many value-added services. In addition to the functions and services of common telephones, the system also provides various functions, such as setting messages, filtering incoming and outgoing calls, and tracking calls, applicable to large and medium-sized enterprise groups or organizations.

1. SIP protocol Overview
In general, the SIP protocol supports the following features in multimedia communication:
(1) User positioning: determine the location of the terminal in the communication;
(2) User availability: Determine whether the called party is willing to participate in communications;
(3) Performance negotiation: Determine the media and media parameters used in communication;
(4) Session creation: the establishment of session parameters of both parties;
(5) session management: includes session transfer and suspension, session parameter changes, and new business calls.
The SIP protocol is a client server protocol used to initiate and manage user sessions. The SIP terminal system is called a user agent, that is, UAUser.
Agent), including the User Agent client UAC (User Agent C1ient) and User Agent server UAS (User Agent)
Sever. An intermediate unit is called a proxy server. Its messages are divided into two categories: requests from the client to the server and responses from the server to the client ). Both the request Message and Response Message are composed of Start-Line, Message-Header, and optional Message bodies.
Request messages start from the request line) can be divided into: (1) Register: Used to Register contact information; (2) Invite: Used to Invite users to join sessions; (3) ACK: used to confirm successful requests; (4) Cancel: Used to Cancel incomplete requests; (5) Bye: session ended; (6) 0 pions: Used to query server performance.
The status code of the response message in the starting line of the Status line) can be divided into: (1) 1XX: temporary response; (2) 2XX: Successful response;
(3) 3XX: Redirection response; (4) 4XX: client error; (5) 5XX: server error; (6) 6XX: Global fault.
The SIP protocol supports three call Methods: User proxy client UAC) direct call to the opposite user Proxy Server UAS); the proxy server initiates a proxy call to the client server on behalf of the user proxy 1 ); redirection call by the user proxy client with the assistance of the redirection Server 2 ). Call method 2 requires the proxy server to forward the user's call signaling, thus increasing the information processing capacity. In order to effectively push the pressure on network devices to the edge of the network, call signaling 3 only specifies the direction of the destination and does not retain the status of each call, thus laying the foundation for building a large-scale IP network.
2. System Design Scheme
This article provides an IP telephone system design solution that complies with the SIP protocol specifications. It adopts the client-server mode and is mainly composed of IP telephone terminal devices based on USB interfaces and computer network terminals (including SIP user agents), LAN or Internet), and SIP network server. The system structure is shown in figure 3. The entire system layout is divided into several domains. Each subsidiary or department is a domain. Each domain has multiple end users and is managed and controlled by one server.
2.1 basic workflow of the system
1) User Registration
After the user is started online, the user agent client automatically sends registration information to the server in the domain. After the SIP registration service module of the server receives registration information, the user must first perform authentication on the client, after confirming that the user is valid, update the user's status information and IP address information.
2) Session Creation
In this system design scheme, proxy call and redirection call are combined to establish a session. That is, customers in the same domain use proxy calls and customers in different domains use redirection calls.
Assume that user A in domain 1 wants to establish A call with user B, and A first transmits the call request to the Proxy Server 1 in the domain through its user proxy client, server 1 first checks whether B also belongs to the domain managed by the server.
①
If B and A belong to the same 1 domain, the server further searches for information about B for exact location. Server 1 determines whether B is connectable, if possible, the call request of A is directly forwarded to B; otherwise, A is rejected.
②
If B is not in the 1 domain, but in other domains such as the 2 domain), Proxy Server 1 forwards the request information to the redirection Server 2 in the 2 domain, and the redirection Server 2 precisely locates B, determine whether B is connectable. For example, if the general rule can be used to transmit the precise address information of B to proxy server 1, Proxy Server 1 will send the call request of A to B; otherwise, the connection failure information is returned to proxy server 1, and Proxy Server 1 returns the denial information to proxy server.
3) Call Process
If the communication link between A and B is established successfully, the communication between them is directly established until the session ends. When the call ends, the proxy server sends the session end request.
2.2 Server Design
In this system, each domain is managed and controlled by a server, which is equivalent to a smart hub in the domain, that is, a network guard keeper ). Each network server consists of four functional modules: proxy service module, redirection module, registration service module, and value-added service module.
(1) SIP proxy)
After receiving a UAC call request, the SIP proxy service module resolves the call ID to determine the region of the called party, and then forwards the request to the next hop server or UAS.
(2) SIP redirect module)
The SIP redirection module receives the request, completes Address Resolution, and returns the address information of the called party to the caller, allowing the caller to send the request directly in the next hop.
(3) SIP register module)
The SIP registration service module is used to assign an ID number to a new user, update the user's login and registration address, and provide the location service.
(4) SIP value-added Service module (SIP value-added module)
The server is also responsible for managing the database of the IP system in addition to the network guard function in the SIP protocol. It uses open interfaces such as detailed records of all sessions and customer Registration Information provided by the IP system database to provide users with a variety of SIP value-added services, such as billing management, ticket query, user messages, incoming/outgoing electricity filtering, and call tracking.
The system's operation, usage, value-added services, and other functional modules are basically centered on tables such as customer registration and session details. There are also many other important data tables, such as the user IP Address Table, user Expense Table, level permission table, and business table.
2.3 client Design
The client consists of two modules: the user agent module and the voice module.
1) SIP User proxy module UA is divided into user proxy client (UAC) module and user Proxy Server UAS module. UAC initiates a call to another customer or server, and UAS responds to the call from another customer or server.
2) The speech module includes audio data acquisition/playback, A/D conversion, encoding/decoding, receiving/sending, and other sub-modules. The data acquisition/playback and A/D switching modules are implemented by the digital telephone terminal device. The end devices of this system are a specially designed digital telephone based on USB interfaces. It adopts an MCU-centered architecture and has functions such as dialing, Audio Acquisition and playback, and A/D switching. The voice sampling rate is 8 kHz, and the sample precision is 8 bit.
The data encoding/decoding module and receiving/sending module are all implemented by the upper-layer application software of the client. This not only reduces the load on the lower computer, but also reduces device costs. In addition, you can complete configuration changes or system expansion of the entire IP Phone System without changing the hardware of the system.
When selecting the voice encoding method, the G.723 encoding technology recommended by CCITT is adopted based on factors such as bandwidth, encoding delay, and application requirements.
2.4 System Protocol Structure
Because SIP is not a vertical communication system and cannot provide services independently, it must be used together with other protocols to establish a complete multimedia architecture. The protocol structure used in this design scheme is as follows:
At the application layer, the SIP protocol is mainly used for session establishment, management, and performance negotiation. Because the SIP protocol itself provides a reliable validation mechanism, UDP protocol is used at the transport layer to support signaling transmission.
Real-time stream protocol (RTSP) is used to control multimedia data streams from one point to multiple points.
To ensure high quality of service (QoS), the system uses the Resource Reservation Protocol (RSVP)
And real-time transmission control protocol RTCP ). The former specifies the IP network resource protection technology, which can reserve resources for one or more) given sessions, and this session takes precedence over any attempt to participate in other media exchanges between the two parties; the latter is used to detect and potentially solve sending problems, so as to monitor session quality and network problems to monitor QoS.
Real-time transmission protocol (RTP) is used to complete real-time transmission of end-to-end voice data. After the SIP-based IP Phone System receives end-to-end QoS support, the UDP protocol can avoid increasing call establishment latency when the connection times out when the network load is heavy. Therefore, this scheme uses UDP to transmit voice information at the transport layer. Here we can regard RTP as an application service running on UDP protocol, which forms different parts of the transmission function required to support real-time data transmission. The RTP Header contains information such as the Server Load balancer format, serial number, timestamp, and transmission monitoring, as shown in. Since RTP data units are carried by UDP groups, the net voice load is usually very short to minimize latency. IP, UDP, and RTP headers are all calculated based on the minimum length. This method inserts multiple voices into the voice data segment to improve transmission efficiency.
0-1234-789-1516-31
VPXCCMPT serial number
Timestamp
Synchronization source flag, etc.
0-15 entries, such as the source flag)
2.5 System Security Mechanism
This design scheme focuses on the security issues in two aspects: server-side database management and network voice data transmission. It can effectively improve the data security of the server through the proxy update mechanism, identity authentication and authorization mechanism. The system strictly limits the UA's Operation Command permissions on the server, and authenticates the user identity, giving different users different permissions.
As needed, the system can encrypt the voice data transmitted between session participants. Data encryption algorithms are embedded in the voice sending and receiving modules of the customer terminal, which can effectively prevent data leaks even if the data is intercepted.
3. Summary
The proposed SIP-based IP telephone system has the advantages of low investment, low cost, convenient and practical, high reliability, and good security. The system is built on a general computer network and can be used on a LAN or the Internet. It has no special requirements for users; in addition, it can be extended into a network interactive multimedia system integrating text and video conferencing. As many terminals use PC resources, the system has a high level of intelligence. system design specifications, the centralized management module not only reduces costs, but also facilitates system operation and maintenance. Because the system adheres to the simplicity of the TCP/UDP protocol family, the vast majority of functions are implemented through software and simple improvements can meet different applications and needs.

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