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  Site Home » Research & Science » VoIPs
   
 

SIP Server

   
The design is such that SIP clients use TCP and UDP port 5060 to connect the SIP servers and others SIP endpoints. The sole purpose of SIP uses are setting up and tearing down voice or video calls. Where session initiation is required, however, SIP is used as well. These include Event Subscription and Notification, Terminal mobility and so on. A lot of SIP-related RFCs actually define behaviour for such solutions. All voice/video communications are done over separate session protocols, like RTP.

SIP was tasked to provide a signaling and call setup protocol for IP-based communications that can support a superset of the call processing functions and features present in the Public Switched Telephone Network (PSTN). The SIP Protocol by itself, defines the focus of the call-setup and signaling. However, it has been designed to enable the building of these featured network elements called Proxy Servers and User Agents. As such, these are features that permit familiar telephone-like operations: dialing, ringing, hearing ring tones or busy signals. The implementation might be different but the behaviour is similar.

SIP enabled telephony networks can also implement many of the more advanced call processing features present in SS7 (Signaling System 7), though the two protocols themselves are different. SIP is a peer-to-peer protocol, characterized by highly complex central network architecture and dumb endpoints; therefore it requires only a very simple core network with intelligence distributed to the network edge, embedded. The features of SIP are implemented as compared to traditional SS7 features, which are implemented in the network.

SIP is characterized by its proponents as having roots in the IP community rather than the telecom industry although many other VoIP signaling protocols exist. While the H. 323 VoIP protocol has been traditionally more associated with the ITU, SIP has been standardized and governed primarily by the IETF. The two organizations have endorsed both protocols in some fashion.

SIP is like HTTP and shares some of its design principles as well. Things like readable by humans and request-response structured. Codes like '404 not found' are used as well. SIP is much simpler than H. 323. However, some would counter that while SIP has a goal for simplicity, its current state is as complex as H. 323. It is a stateless control, that some might point out, hence making it possible to fail like other difficult protocols as well. SIP and H. 323 are not limited to voice communication but can mediate any kind of communication session from voice to video, maybe even unrealized applications in future.

Author: San Singhania
 
Author Bio:

Article contributed by San Singhania who is a VoIP consultant at Lantone Communications. Led by a team of experienced IT developers, Lantone Communications is one of the leading VoIP Provider in Singapore. Please visit their official website for the latest information on VoIP. This article may be reprinted in its original form as long as the resource box is left intact and the links live and the article is not to be modified in any way.

This article can be searched using: SIP Server, Research & Science, VoIPs, voip data networking, voip p, internet telephony voip
 
 
 

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