Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol for person-to-person voice traffic over the Web. 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 handle IP telephony combined with video and instant messaging. Future improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Bridgewater.
Manufacturers of SIP hardware are rapidly developing new hardware and software to get the upper hand in this new Internet communications method. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before implementing a SIP solution in your business, you should consider the appropriate settings of your corporate firewall to support SIP. Many of the typical firewalls currently installed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are often closed 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 provider or Metro fiber provider is able to support the SIP protocol from your internal network to the outside world. Your IT administrator 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 complicated SIP situations for secure and private communications.
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