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Copyright 2008 by The McGraw-Hill Companies. Click here for terms of use.
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The software on the computer included a program that allowed you to look up telephone numbers and dial them from the computer. A handset connected to the back of the computer then allowed you to use the computer as your phone (perhaps the first address book ). But the voice itself was still analog and did not go through the data network. In fact, the data connection was made through the telephone network. Then as years moved on, digital switching became more prevalent and we began seeing systems designed to support both voice and data. Of course, voice was always the primary function of the switches, with the added capability of packet switching. I remember the first true digital packet switch I worked with was a thing called the Lexar, built by Lexar Corporation (later to become United Technologies, then Telex, than Memorex Telex, and then I lost track). The voice was actually packetized and sent to digital phones, which meant that data could also be packetized and sent to and from the phones. Computers connected via a serial connector (remember those ) on the back of the phone at a dizzying speed of 19.2 Kbps. We were getting closer to seeing computers and data combined, but still, there lacked a network that could bridge all of these systems together. There were, of course, mainframe computers and dedicated networks running SNA. I did work on some of these for a while, but they were so inflexible and the SNA protocol was very limited compared to today s protocols. The network elements were pretty basic, operating at the transport layer to route data packets from mainframe to terminal. These networks were really designed to provide dumb terminals access to the mainframes, rather than to share data between users. Then came the Internet, and everything changed. Suddenly we all had access to a giant public network that allowed us to connect and exchange all types of data with anyone who had a connection. New Internet service providers (ISPs) began popping up offering electronic mail and message boards, and later news services and newsgroup access. These ISPs continue to grow and offer new services as the Internet matured. When the World Wide Web (WWW) was introduced, communication was changed forever. Suddenly we could shop online, establish our own presence on the Internet, and socialize with people from all over the world. It was only natural that the next big service would be Voice over IP (VoIP). It is very fulfilling for me to see these networks finally maturing to where they have become mainstream after so many decades of experimentation and failed attempts. Today we can finally enjoy the many features and capabilities that packet networks enable with telephone service combined with our e-mail and data service. Yet for many years it was like the Wild Wild West, with many different implementations and many different configurations to support VoIP networking. There were the legacy telephony guys trying to make the packet network emulate a switched network, and the data guys trying to make the packet network support voice. In the end, we meet in the middle. New protocols have been developed to help make VoIP more robust and reliable while supporting quality as good as or superior to legacy networks. We see networks beginning to mature and standardize on their technologies so that they can interoperate with other networks, and so vendor equipment will interoperate end to end.
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And now, the Session Initiation Protocol (SIP) emerges from the dust as the winner for session control, at least as far as the legacy service providers are concerned. The fact that the 3rd Generation Partnership Project (3GPP) has defined SIP as the standard for call session control for their IP Multimedia Subsystem (IMS) architecture is not minor. There is significant work underway today to add to the SIP protocol to enable it to provide services like we have never seen before. This book attempts to capture most of the SIP protocol as it has been defined today. This includes the baseline RFC 3261, and many of the extensions that have followed to support new services in the network using the SIP protocol. As SIP continues to mature and grow, I will capture those changes in subsequent editions of this book, so this has become an ongoing work in progress. I have attempted to simplify SIP as much as possible, and focus on the baseline functions rather than go into great detail about the protocol and its procedures. There is quite a bit of detail here, without going into the nitty-gritty developer stuff. Hopefully this will help everyone to better understand how SIP works, and how to make SIP work in your own network.
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