Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) 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 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 extended to deal with IP telephony combined with video and instant messaging. Forthcoming enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Dix.
Manufacturers of SIP equipment are rapidly creating new products and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Prior to implementing 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 already 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 often blocked 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 Ethernet fiber service 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 challenges. By including a SIP proxy and registrar for controlling the firewall, it is possible to handle complicated SIP scenarios for secure and confidential communications.
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