Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol for person-to-person voice traffic over the Internet. Its defining 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 extended to deal with IP telephony in conjunction with video and instant messaging. Forthcoming enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Woodstock.
Manufacturers of SIP equipment are rapidly creating innovative products and software to get the upper hand in this new Internet communications scheme. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before installing a SIP solution in your company, you should think about the appropriate settings of your corporate firewall to accept SIP. Many of the typical firewalls currently installed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transferred over dynamically allocated 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 unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Metro fiber service is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to correctly support SIP to get around these technical challenges. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complex SIP scenarios for secure and confidential communications.
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