Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications technology for person-to-person real time traffic over the Internet. 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 extended to handle IP telephony in conjunction 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 Two Harbors.
Manufacturers of SIP components are quickly developing innovative hardware 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.
Before implementing a SIP plan in your organization, you should consider the appropriate settings of your corporate firewall to support SIP. Many of the common firewalls already deployed in business offices are not designed to support 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 reached 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 handle the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to appropriately support SIP to overcome these technical problems. By adding a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP scenarios for reliable and confidential communications.
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