Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications technology for person-to-person real time traffic over the Web. 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 deal with IP telephony combined with video and instant messaging. Future enhancements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Mckeesport.
Producers of SIP components are rapidly designing new hardware and software to take advantage of this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP solution in your organization, you should consider the proper configuration of your corporate firewall to accept SIP. Many of the common firewalls already deployed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are typically blocked on firewalls. Second, SIP clients inside a firewall can not be reached using IP addresses since these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line carrier or Ethernet fiber provider is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to appropriately support SIP to get around these technical issues. By adding a SIP proxy and registrar for controlling the firewall, it is possible to handle complex SIP scenarios for reliable and private communications.
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