Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport protocol for person-to-person real time data 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 handle IP telephony in conjunction with video and instant messaging. Forthcoming enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Beaver.
Suppliers of SIP equipment are quickly creating innovative hardware and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Prior to installing a SIP solution in your business, you should think about the proper settings of your corporate firewall to accept SIP. Many of the common firewalls currently deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are usually closed on firewalls. Second, SIP clients inside a firewall can not be reached using IP addresses because these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line broadband provider or Ethernet fiber carrier 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 overcome these technical difficulties. By including a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP situations for secure and private communications.
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