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SIP: Alternative Softswitch Architecture
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SIP: Alternative Softswitch Architecture
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Figure 5-1 SIP is a signaling protocol and RTP transports the conversation protocol.
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specifically to ensure transport (RTP), provide signaling with the PSTN, guarantee voice quality (Resource Reservation Setup Protocol [RSVP]), provide directories (Lightweight Directory Access Protocol [LDAP]), authenticate users (Remote Access Dial-In User Service [RADIUS]), and scale to meet anticipated growth curves.
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SIP is focused on two classes of network entities: clients (also called user agents [UAs]) and servers. VoIP calls on SIP to originate at a client and terminate at a server. Types of clients in the technology currently available for SIP telephony include a PC loaded with a telephony agent or an SIP telephone. Clients can also reside on the same platform as a server. For example, a PC on a corporate wide area network (WAN) might be the server for the SIP telephony application, but it may also be used as a user s telephone (client).
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SIP is a client-server architecture. The client in this architecture is the UA, which interacts with the user. It usually has an interface towards the user in the form of a PC or an IP phone (an SIP phone in this case). Four types of SIP servers exist. The type of SIP server used determines the architecture of the network. Those servers are the user agent server (UAS), the redirect server, the proxy server, and a registrar. SIP Calls via a UA Server A UA server accepts SIP requests and contacts the user. A response from the user to the UA server results in an SIP
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SIP: Alternative Softswitch Architecture
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response representing the user. An SIP device will function as both a UA client (UAC) and as a UA server. As a UAC, the SIP device can initiate SIP requests. As a UA server, the device can receive and respond to SIP requests. As a standalone device, the UA can initiate and receive calls that empowers SIP to be used for peer-to-peer communications. Figure 5-2 describes the UA server. The function of SIP is best understood via the HTTP model upon which it is based. SIP is a request/response protocol. A client is an SIP entity that generates a request. A server is an SIP entity that receives requests and returns responses. When a web site is desired, the client generates a request by typing in the URL, such as www.mcgrawhill.com. The server upon which the web site is hosted responds with McGraw-Hill s web page. SIP uses the same procedure. The UA sending the request is known as a UAC, and the UA returning the response is the UAS. This exchange is known as an SIP transaction. Per Figure 5-2, a call initiates with the UA of the calling party sending an INVITE command caller@righthere.org to the called party, callee@theotherside.com. The UA for the caller has translated the name for the called party into an IP address via a Domain Name System (DNS) query accessible via their own domain. The INVITE command is sent to an SIP User Datagram Protocol (UDP) port and contains information such as media format and the From, To, and Via information. The TRYING informational response (180) from the calling party s call agent is analogous to the Q.931 CALL PROCEEDING message used in the PSTN, indicating the call is being routed. In the direct call model, a TRYING response is unlikely, but for proxy and redirect models it is used to monitor call progress. SIP
Figure 5-2 SIP UA server to UA server call
Calller (1) Invite (2) 180 Ringing (3) 200 OK (4) ACK
Calllee
Conversation (5) BYE (6) 200 OK
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