Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol for person-to-person real time data over the Web. Its technical specifications come from the SIP working group of the Internet Engineering Task Force. SIP gives access to the public switched telephone network for voice data at 3 kHz bandwidth and common number dialing using Voice over IP (VoIP). It can also be extended to deal with IP telephony combined with video and instant messaging. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Cocoa Beach.
Producers of SIP equipment are rapidly crafting innovative products and software to take advantage of this new Internet communications method. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before implementing a SIP plan in your organization, you should think about the appropriate configuration of your corporate firewall to accept SIP. Many of the typical firewalls currently deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are often blocked on firewalls. Second, SIP clients within a firewall can not be accessed using IP addresses since these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Metro fiber carrier is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to appropriately support SIP to overcome these technical problems. By including a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complex SIP scenarios for secure and private communications.
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