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Establishing a Session in SIP
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This means that the BYE is not sent outside of a dialog. In other words, if there is no dialog or session established, then the BYE method cannot be used. The CANCEL method is used to cancel a request prior to a dialog/session being established. When using the CANCEL method, the UAC initiates the message and sends it to the UAS. The UAS has not yet replied to the request; therefore, the UAS should have no knowledge of the request, which is why the UAC must always initiate the CANCEL. The UAS will then send the 487 REQUEST TERMINATED response to the CANCEL. Should the UAC send the CANCEL and then receive a 2xx final response, it can send an ACK to establish the session, or it can terminate the session by sending the BYE. The UA terminates a session for one of two reasons: the user wishes to terminate the session, or there was an error. An error could occur, for example, if a 200 OK is sent by the UAS but an ACK is never received. The UAS would then send the BYE message to terminate the dialog. The BYE message must contain tags so that it can be correlated with a dialog ID and session. Otherwise, it will not be clear which session/dialog the BYE is terminating. The only response to the BYE is the 200 OK response used to confirm that the BYE was received and resources can be released.
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Extending SIP to Support New Functions
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In the early days of Voice over IP (VoIP), there were many interoperability issues. There were also many different protocols used to control the various sessions within a network, adding to the often chaotic implementation of an IP network. Many vendors supported only some of the protocols, electing not to develop to all of the standards that were being introduced for these networks. Instead they elected to carve their own paths and hope that they could win the influence of enough customers to make their platforms the new standard. This was the biggest frustration shared by many traditional operators trying to implement VoIP in their stable, traditional telephony networks. These networks consist of tried and true technologies that are based on age-old standards with many years of maturity. Vendors have conformed to these standards, ensuring now that interoperability is a given. It wasn t always this way, though, and so it comes as little surprise that we are seeing the same interoperability issues today that we experienced when SS7 was first introduced. Yet the attraction to VoIP for many is the ingenuity. There are many more unique and killer applications now available based on VoIP technologies that we probably would have never realized using the old tried and true technologies operating now. What is needed is a technology that can be standardized to a level that guarantees interoperability, while allowing for ingenuity and vendor uniqueness. That was accomplished through the introduction of the Session Initiation Protocol (SIP) and its support of extensions. SIP was meant to be backward compatible with all earlier version of SIP to prevent operators networks from becoming isolated due to non-compliance with some software upgrade. SIP was also meant to be interoperable with all operators equipment in an effort to eliminate proprietary solutions in the network. These were some of the issues experienced by many operators in the early days of SS7 and the IN. Some equipment did not interoperate with other equipment, forcing operators to purchase their infrastructure from one vendor (or one vendor s partners).
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Of course this traps the operator and the subscriber as well, because they are both then limited to that vendor s solutions and accompanying features. With SIP, operators can implement any platform that is compliant with RFC 3261 and be assured that it will be able to process SIP traffic. While there will still be proprietary SIP implementations, RFC 3261 dictates that all SIP entities must be compatible with one another by adhering to the requirements of the RFC. Other RFCs have been published to define the interactions with other SIP capabilities and functions such as messaging and other similar applications. What this does is allow operators to implement a SIP variation supporting many unique functions and applications through customized SIP headers. These headers allow SIP to provide much more capability, while also ensuring that as SIP travels through other networks, the other networks are still able to recognize the headers needed to route the message to its destination without any degradation in service. There have been many contributions from vendors developing SIP extensions. Some are published, and some remain proprietary. The 3GPP also realized that RFC 3261 did not address all of their immediate needs, and therefore they defined a number of extensions for use within their IMS architecture. Those extensions were documented in subsequent RFCs and published through the IETF. Those extensions are described in this chapter as well.
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