Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport protocol for person-to-person voice data 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 expanded to handle IP telephony combined with video and instant messaging. Future improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Fort Defiance.
Producers of SIP hardware are quickly developing innovative products and software to take advantage of this new Internet communications technique. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP plan in your organization, you should consider the appropriate settings of your corporate firewall to support SIP. Many of the typical firewalls currently installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are normally shut on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses because these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier 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 appropriately support SIP to overcome these technical issues. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complex SIP scenarios for secure and confidential communications.
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