Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol for person-to-person voice 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 expanded to handle IP telephony in conjunction with video and instant messaging. Forthcoming improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Kemmerer.
Manufacturers of SIP equipment are quickly designing new products 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 organization, you should think about the proper configuration of your corporate firewall to support SIP. Many of the common firewalls currently deployed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are typically blocked on firewalls. Second, SIP clients inside 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 carrier or Ethernet fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to correctly support SIP to overcome these technical problems. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to handle complicated SIP situations for reliable and private communications.
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