Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person voice data 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 deal with IP telephony in conjunction with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Kimball.
Makers of SIP components are rapidly crafting innovative hardware and software to get the upper hand in this new Internet communications scheme. 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 plan in your organization, you should think about the appropriate configuration of your corporate firewall to accept SIP. Many of the common firewalls currently deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are usually shut 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 carrier or Ethernet fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to correctly support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for managing the firewall, it is possible to deal with complex SIP situations for reliable and confidential communications.
