Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission technology 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 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 combined with video and instant messaging. Forthcoming enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Ware Shoals.
Producers of SIP equipment are rapidly creating innovative hardware and software to get the upper hand in this new Internet communications method. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to installing a SIP plan in your organization, you should consider the proper settings of your corporate firewall to support SIP. Many of the common firewalls currently 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 shut on firewalls. Second, SIP clients within a firewall can not be reached 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 Metro fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to appropriately support SIP to get around these technical issues. By including a SIP proxy and registrar for controlling the firewall, it is possible to handle complicated SIP scenarios for reliable and confidential communications.
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