Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport protocol for person-to-person voice traffic over the Internet. 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 enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Bethel.
Makers of SIP hardware are quickly developing new products and software to get the upper hand in this new Internet communications method. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP solution in your company, you should consider the appropriate settings of your corporate firewall to support SIP. Many of the common firewalls already deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are often blocked on firewalls. Second, SIP clients inside a firewall can not be accessed 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 provider is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to properly support SIP to get around these technical problems. By adding a SIP proxy and registrar for managing the firewall, it is possible to handle complicated SIP scenarios for reliable and private communications.
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