Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport protocol for person-to-person real time 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 extended to handle IP telephony in conjunction with video and instant messaging. Future improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Groton.
Suppliers of SIP equipment are rapidly developing innovative hardware 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.
Before installing a SIP solution in your company, 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 designed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are often shut 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 broadband carrier or Ethernet fiber service is able to handle the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to correctly support SIP to overcome these technical issues. By including a SIP proxy and registrar for managing the firewall, it is possible to handle complex SIP scenarios for reliable and private communications.
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