Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications technology for person-to-person real time traffic 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 Bellville.
Manufacturers of SIP hardware are rapidly creating new hardware and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to implementing a SIP solution in your organization, you should think about the proper settings of your corporate firewall to support SIP. Many of the common firewalls currently installed 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 since 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 properly support SIP to get around these technical issues. By adding a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP scenarios for secure and private communications.
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