Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology for person-to-person voice data over the Web. Its technical specifications come from the SIP working group of the Internet Engineering Task Force. SIP provides 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 expanded to handle IP telephony in conjunction 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 Rolla.
Makers of SIP equipment are quickly creating innovative products and software to get the upper hand in this new Internet communications method. SIP telephones, PC client software, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Before installing a SIP solution in your company, you should consider the appropriate settings of your corporate firewall to accept SIP. Many of the typical firewalls already deployed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are typically shut on firewalls. Second, SIP clients within a firewall can not be accessed using IP addresses because these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line broadband provider or Metro fiber provider is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to appropriately support SIP to get around these technical problems. By including a SIP proxy and registrar for managing the firewall, it is feasible to handle complicated SIP scenarios for secure and confidential communications.
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