Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission protocol for person-to-person voice traffic over the Internet. Its defining specifications come from the SIP working group of the Internet Engineering Task Force. SIP provides 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 deal with IP telephony in conjunction with video and instant messaging. Forthcoming improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Rockmart.
Manufacturers of SIP components are rapidly designing innovative products and software to get the upper hand in 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 plan in your organization, you should take into account the appropriate configuration of your corporate firewall to accept SIP. Many of the typical firewalls currently installed in business offices are not developed to support the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are normally shut on firewalls. Second, SIP clients inside a firewall can not be reached using IP addresses because these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband provider or Metro 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 overcome these technical problems. By including a SIP proxy and registrar for managing the firewall, it is possible to deal with complex SIP scenarios for secure and confidential communications.
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