Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission 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 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 expanded to deal with IP telephony combined with video and instant messaging. Forthcoming enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Leonardtown.
Manufacturers of SIP components are rapidly developing new 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 obtainable from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP solution in your organization, you should think about the appropriate settings of your corporate firewall to support SIP. Many of the common 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 usually blocked 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 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 evaluate how to correctly support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is possible to handle complicated SIP scenarios for reliable and confidential communications.
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