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SIP: Alternative Softswitch Architecture
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SIP: Alternative Softswitch Architecture
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Table 5-1 SIP signaling methods
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SIP method INVITE
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Description The first message sent by a calling party to invite users to participate in a session. It contains information in the SIP header that identifies the calling party, call ID, called party, and call and sequence numbers. It indicates a call is being initiated. When a multiple choice of SDP parameters is offered, the ones chosen are returned with the success (200) code in the response message. This is used to acknowledge the reception of a final response to an INVITE. A client originating an INVITE request issues an ACK request when it receives a final response for the INVITE, providing a three-way handshake. This queries a server about its capabilities, including which methods and which SDPs it supports. This determines which media types a remote user supports before placing the call. This is used to abandon sessions. In two-party sessions, abandonment by one of the parties implies that the session is terminated. A return BYE from the other party is not necessary. This method cancels pending transactions. The CANCEL method identifies the call via the call ID, call sequence, and To and From values in the SIP header. These requests are sent by users to inform a server about their current location. SIP servers are co-located with SIP registrars. This enables the SIP server to find a user.
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uses six methods of signaling, as shown in Table 5-1. Additional methods are under consideration by the IETF at this time. When the call arrives at the remote endpoint, the phone rings and a new response is sent to that endpoint: RINGING (180). This is analogous to the Q.931 ALERTING message used in the PSTN. The time between the user dialing the last digit and the time RINGING is received by the caller is known as the postdial delay (PDD) for SIP call setup. If a telephone number is involved in addressing the call, the numbers must be translated into an IP address. Table 5-2 provides a comparison of SIP and PSTN signals. When the called party answers the phone, a 200 response is sent to the calling party s UA. The UA sends another request, ACK, acknowledging the response to the INVITE request. At that moment, media begins to flow on the transport addresses of the two endpoints. The ACK may carry the final SDP parameters for the media type and format provided by the receiving
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SIP: Alternative Softswitch Architecture
Table 5-2 Comparison of SIP and PSTN signals
SIP TRYING RINGING ACK XINVITE PSTN Q.931 Call Proceeding Q.931 Alerting Q.931 Connect Q.931 Connect
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endpoint. The sequence of the INVITE and following ACK messages is similar to the Q.931 CONNECT message. ACKs do not require a response. Table 5-3 displays the response codes for SIP. At this point in the call sequence, as seen in Figure 5-1, media flows over RTP, with the Real-Time Control Protocol (RTCP) providing the monitoring of the quality of the connection and its associated statistics. Next, as the name would imply, a BYE request from either party ends the call. As all messages are sent via UDP, no further action is required. SIP Calls via a Proxy Server Proxy servers in the SIP sense are similar in function to proxy servers that serve a web site (mail relay via SMTP) for a corporate local area network (LAN). An SIP client in this case would send a request to the proxy server that would either handle it or pass it on to another proxy server that, after some translation, would take the call. The secondary servers would see the call as coming from the client. By virtue of receiving and sending requests, the proxy server is both a server and a client. Proxy servers function well in call forwarding and follow-me services. Proxy servers can be classified by the amount of state information they store in a session. SIP defines three types of proxy servers: call stateful, stateful, and stateless. Call stateful proxies need to be informed of all the SIP transactions that occur during the session and are always in the path taken by SIP messages traveling between end users. These proxies store state information from the moment the session is established until the moment it ends. A stateful proxy is sometimes called a transaction stateful proxy as the transaction is its sole concern. A stateful proxy stores a state related to a given transaction until the transaction concludes. It does not need to be in the path taken by the SIP messages for subsequent transactions. Forking proxies are good examples of stateful proxies. Forking proxies are used when a proxy server tries more than one location for the user; that is, it forks the invitation.
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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