Voice over Internet Protocol in Software

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Voice over Internet Protocol
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the deployment of IPv6. IPv6 makes possible infinitely more addresses than IPv4. Enhancements offered by IPv6 over IPv4 include
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Expanded address space Each address is allocated 128 bits instead of 32 bits in IPv4. Simplified header format This enables easier processing of IP datagrams. Improved support for headers and extensions This enables greater flexibility for the introduction of new options. Flow-labeling capability This enables the identification of traffic flows for real-time applications. Authentication and privacy Support for authentication, data integrity, and data confidentiality are supported at the IP level rather than through separate protocols or mechanisms above IP.
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This chapter addressed the building blocks of VoIP. It will be necessary in future chapters to understand many of the concepts contained in this chapter. Just as the PSTN and softswitch networks can be broken down into the three elements of access, switching, and transport, VoIP can be summarized as a study of three types of protocols: signaling, routing, and transport. The proper selection of VoIP signaling protocols for a network is an issue of almost religious proportions among network builders. Although protocols will continue to evolve and new protocols will emerge, those addressed in this chapter will constitute the predominant structure of softswitch architecture for the next few years.
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Source: Softswitch Architecture for VoIP
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CHAPTER
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SIP: Alternative Softswitch Architecture
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SIP: Alternative Softswitch Architecture
5
If the worldwide Public Switched Telephone Network (PSTN) could be replaced overnight, the best candidate architecture, at the time of this writing, would be based on Voice over IP (VoIP) and the Session Initiation Protocol (SIP). Much of the VoIP industry has been based on offering solutions that leverage existing circuit-switched infrastructure (such as VoIP gateways that interface a private branch exchange [PBX] and an Internet Protocol [IP] network). At best, these solutions offer a compromise between circuit- and packet-switching architectures with resulting liabilities of limited features, expensive-to-maintain circuit-switched gear, and questionable quality of service (QoS) as a call is routed between networks based on those technologies. SIP is an architecture that potentially offers more features than a circuit-switched network. SIP is a signaling protocol. It uses a text-based syntax similar to the Hypertext Transfer Protocol (HTTP) like that used in web addresses. Programs that are designed for the parsing of HTTP can be adapted easily for use with SIP. SIP addresses, known as SIP uniform resource locators (URLs) take the form of web addresses. A web address can be the equivalent of a telephone number in an SIP network. In addition, PSTN phone numbers can be incorporated into an SIP address for interfacing with the PSTN. An email address is portable. Using the proxy concept, one can check his or her email from any Internet-connected terminal in the world. Telephone numbers, simply put, are not portable. They only ring at one physical location. SIP offers a mobility function that can follow subscribers to whatever phone they are nearest to at a given time. Like H.323, SIP handles the setup, modification, and teardown of multimedia sessions, including voice. Although it works with most transport protocols, its optimal transport protocol is the Real Time Protocol (RTP) (refer to 3, Softswitch Architecture or It s the Architecture, Stupid! for more information on RTP). Figure 5-1 shows how SIP functions as a signaling protocol while RTP is the transport protocol for a voice conversation. SIP was designed as a part of the Internet Engineering Task Force (IETF) multimedia data and control architecture. It is designed to interwork with other IETF protocols such as the Session Description Protocol (SDP), RTP, and the Session Announcement Protocol (SAP). It is described in the IETF s RFC 2543. Many in the VoIP and softswitch industry believe that SIP will replace H.323 as the standard signaling protocol for VoIP. SIP is part of the IETF standards process and is modeled upon other Internet protocols such as the Simple Mail Transfer Protocol (SMTP) and HTTP. It is used to establish, change, and tear down (end) calls between one or more users in an IP-based network. In order to provide telephony services, a number of different standards and protocols must come together
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