vb.net barcode generator Protocols Related to VoIP in Software

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Protocols Related to VoIP
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The softswitch revolution was made possible by the emergence of voice over data, more specifically, VoIP. It should be noted here that softswitch solutions use TDM and ATM. However, the consensus in the industry is that the future is an IP network ultimately dictating a VoIP solution. Before outlining softswitch solutions, it will first be necessary to understand VoIP. VoIP
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Report to Congress on Universal Service, CC Docket No. 96-45, White Paper on IP Voice Services, March 18,1998 (www.von.org/docs/whitepap.pdf).
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Downloaded from Digital Engineering Library @ McGraw-Hill (www.digitalengineeringlibrary.com) Copyright 2004 The McGraw-Hill Companies. All rights reserved. Any use is subject to the Terms of Use as given at the website.
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Voice over Internet Protocol
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Voice over Internet Protocol
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is best understood as a collection of the protocols that make up its mechanics. Those protocols are loosely analogous to the PSTN that is broken down into three categories: access, switching, and transport. Simply put, three categories of protocols are relevant to VoIP: signaling, routing, and transport. Signaling (roughly analogous to the switching function described in the last two chapters) protocols (H.323 and Session Initiation Protocol [SIP]) set up the route for the media stream or conversation. Gateway control protocols such as the Media Gateway Control Protocol (MGCP) and MEGACO (also signaling protocols) establish control and status in media and signaling gateways. Routing (using the User Datagram Protocol [UDP] and Transmission Control Protocol [TCP]) and transporting (Real-Time Transport Protocol [RTP]) the media stream (conversation) once the route of the media stream has been established are the functions of routing and transport protocols. Routing protocols such as UDP and TCP could be compared to the switching function described in 2, The Public Switched Telephone Network (PSTN), and 3, Softswitch Architecture or It s the Architecture, Stupid! RTP would be analogous to the transport function outlined in earlier chapters describing the PSTN and softswitch architectures. The signaling and routing functions establish what route the media stream will take. The routing protocols deliver the bits, that is, the conversation.
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Signaling Protocols
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The process of setting up a VoIP call is roughly similar to that of a circuitswitched call made on the PSTN. A media gateway must be loaded with the parameters to allow proper media encoding and the use of telephony features. Inside the media gateway is an intelligent entity known as an endpoint. When the calling and called parties agree on how to communicate and the signaling criteria is established, the media stream over which the packetized voice conversation will flow is established. Signaling establishes the virtual circuit over the network for that media stream. Signaling is independent of the media flow. It determines the type of media to be used in a call. Signaling is concurrent throughout the call. Two types of signaling are currently popular in VoIP: H.323 and SIP.3
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Douskalis, Bill. IP Telephony The Integration of Robust VoIP Services. New York: Prentice Hall, 2000.
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Voice over Internet Protocol
Figure 4-1 Signaling and transport protocols used in VoIP
4
Signaling (H.323, SIP)
Transport (RTP)
Telephone
IP Cloud Telephone
Figure 4-1 details the relationship between signaling and media flow. This relationship between transport and signaling is very similar to the PSTN in that Signaling System 7 (SS7) is out-of-channel signaling, such as that used in VoIP. H.323 H.323 is the International Telecommunication Union Telecommunications Standardization Sector (ITU-T) recommendation for packetbased multimedia communication. H.323 was developed before the emergence of VoIP. As it was not specifically designed for VoIP, it has faced a good deal of competition from a competing protocol, SIP, which was designed specifically for VoIP. However, it has enjoyed a first-mover advantage and a considerably installed base of H.323 VoIP networks now exists. H.323 is comprised of a number of subprotocols. It uses protocol H.225.0 for registration, admission, status, call signaling, and control. It also uses protocol H.245 for media description and control, terminal capability exchange, and general control of the logical channel carrying the media stream(s). Other protocols make up the complete H.323 specification, which presents a protocol stack for H.323 signaling and media transport. H.323 also defines a set of call control, channel setup and codec specifications for transmitting real-time video and voice over networks that don t offer guaranteed service or quality of service (QoS). As a transport, H.323 uses RTP, an Internet Engineering Task Force (IETF) standard designed to handle the requirements of streaming real-time audio and video via the Internet.4 H.323 was the first VoIP protocol for interoperability among the early VoIP gateway/gatekeeper vendors. Unfortunately, the promise of interoperability between diverse vendors platforms did not materialize with the adoption of H.323. Given the gravity of this protocol, it will be covered in a separate following chapter.
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