Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission 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 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 extended to handle IP telephony in conjunction with video and instant messaging. Future enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Cleveland.
Makers of SIP hardware are rapidly crafting new products and software to take advantage of this new Internet communications scheme. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Before setting up a SIP plan in your organization, you should consider the appropriate configuration of your corporate firewall to support SIP. Many of the typical firewalls already installed in business offices are not designed to support the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are often closed on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses because these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line provider or Ethernet fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to appropriately support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is feasible to deal with complicated SIP scenarios for secure and confidential communications.
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