Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications technology 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 deal with IP telephony in conjunction with video and instant messaging. Future enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Wilson.
Producers of SIP components are quickly designing new hardware and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to implementing a SIP solution in your organization, you should consider the proper settings of your corporate firewall to support SIP. Many of the common firewalls currently installed in business offices are not developed to support the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are typically shut on firewalls. Second, SIP clients within 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 service is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to correctly support SIP to get around these technical challenges. By adding a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP scenarios for reliable and private communications.
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