Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission technology for person-to-person voice data over the Internet. Its defining 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 combined with video and instant messaging. Forthcoming enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Weimer.
Makers of SIP equipment are quickly designing new products and software to get the upper hand in 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 implementing a SIP solution in your organization, you should think about the proper settings of your corporate firewall to support SIP. Many of the typical firewalls already deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are usually blocked on firewalls. Second, SIP clients within a firewall can not be accessed 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 provider is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to correctly support SIP to overcome these technical problems. By adding a SIP proxy and registrar for managing the firewall, it is possible to handle complicated SIP scenarios for secure and private communications.
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