Session Initiation Protocol SIP-based VoIP system (Figure)-SIP, VoIP protocol st

Be VoIP One of the signaling standard, it is scalable, flexible, simple and easy to implement. In addition, SIP can provide good QoS support for VoIP features to meet the full requirements. This paper introduces the SIP-based VoIP system works, key technologies, system architecture and business environment, and in a business case for the operations of the system is analyzed.

1, VoIP basic principles

VoIP is an IP network in each analog signal transmission technology, the basic principle is through the voice compression algorithm to compress voice and data coding, and then the voice data by TCP / IP standard package, after IP network packet sent to the destination, then the voice packets are linked up through the decompression processing, revert to the original speech signal for those receiving calls, send voice so as to achieve the purpose by the Internet.

Based on Session Initiation Protocol (SIP) for VoIP protocol stack structure shown in Figure 1.

Figure 1 SIP-based VoIP protocol stack structure

2, VoIP key technology

As VoIP entirely on the basis of packet switching, and packet switching inherent delay, packet loss and other vulnerabilities to VoIP call quality can not be guaranteed. VoIP systems must therefore take special measures to guarantee a certain quality of service. VoIP's key technologies are as follows:

1) signaling

Signaling the smooth realization of the telephone calls and voice quality assurance, the key long-range signaling system includes the International Telecommunication Union Communicate Standardization (ITU-T) in the H.323 series and the Internet Engineering Task Force (IETF) of the SIP. H.323 has developed no quality of service packet networks (PBN) on the multimedia communication standard, is relatively mature and has been widely used in the VoIP field. SIP is the IETF web based IP phone developed a problem in the new agreement, so there is more flexibility. Compared with H.323, SIP is a relatively simple protocol. It did not provide all of the H.323 protocol, SIP is only used to initialize the call, not the media data transmission, which is not caused by additional transport costs. SIP Uniform Resource Locator (URL) can even be embedded in Web pages or other hypertext links, the user can use the mouse to point to issue a call, SIP is built for quick call to support the transmission characteristics of telephone numbers.

2) voice processing technology

Voice compression technology is the core of VoIP technology, at present, mainly defined by ITU-T G.729, G.723/G.723.1 so. Because packet switching networks without quality of service, requiring voice coding has some flexibility, that is, coding rate, coding scale adaptation. G.729 can only 8kbit / s of bandwidth to transmit voice, the algorithm used in the structure of the Health and Algebraic Code Excited Linear Predictive coding (CS-ACELP), this algorithm form the basis of the standard G.729. G.723.1 dual rate by 5.3/6.3kbit/s voice coding, voice quality, but larger processing delay, is now the minimum standard rate of voice coding algorithm.

3) computer telephony integration (CTI) technology

Computer telephony integration is through a number of hardware and software computer and telephone are integrated, and it is based on IP telephony technology. Currently, CTI composition can be divided into two ways: a) PC and phone integrated, PC-based exchange network and computer networks are not integrated together. b) PC and the phone is no direct link between using the client / server system (Client / Server) structure, excellent performance CTI server connected to the user-level exchange (PBX), a large computer database or server distributed architecture. The second approach uses the software more complicated, but when the number of users is large, can reduce the cost of each user.

4) QoS support technology

VoIP network QoS guarantees are there: over-building, priority, queue, to avoid congestion and transmission shaping. The main use of VoIP Resource Reservation Protocol (RSVP) to ensure the IP precedence and random get up early detection and weighting techniques to avoid network congestion, protect the call quality.

5) network management technology

Network management technology is the protection of IP phones to operate. IP telephone network management system includes call management system (CMS), traffic analysis system (TAS), network management system (NMS), network monitoring system. Demanding a real-time communication systems, direct impact on the quality of its network communication quality. Network management technology that can quickly troubleshoot network problems, ensure that each node in the network and a stable, efficient operation.

3, SIP of VoIP system and its business environment SIP-based VoIP

3.1 system

SIP is the IETF standard part of the process, built on the Simple Mail Transfer Protocol (SMTP and Hypertext Transfer Protocol (HTTP)) basis. SIP sessions can be realized using the connection, the establishment and release, and to support unicast, multicast, and mobility. In order to provide telephone services, it needs with the combination of other standards and protocols, in particular, to ensure real-time Transport Protocol (RTP) with the current public switched telephone network (PSTN) signaling interconnection, to ensure voice quality (resource reservation protocol ( RSVP)), to provide directory (Lightweight Directory Access Protocol (LDAP)), to authenticate the user (Remote Authentication Dial In User System (RADIUS)) and so on. In addition, SIP and Session Description Protocol if (SDP) with the use, can dynamically adjust and modify the session properties, such as the call bandwidth, the transmission of media types and codecs. SIP-based VoIP system

main features are as follows:

1) Users Proxy (UA)

VoIP system, it is the terminal, including the use SIP protocol software or hardware, such as a IP telephone Or with PC-client software. Each user agent UA, it also includes client systems (UserAgentClient) and user agent Service System (UserAgentServer). User agent client system is used to send SIP user agent service system with a connection request. User Agent service system receives the user agent client system's request, and give responses, including receiving, redirect or reject the call request.

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