Consider SIP Protocol for Voice Over IP
Session Initiation Protocol (SIP) is the transmission protocol for person-to-person voice data over the Web. Its technical 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 handle IP telephony combined with video and instant messaging. Forthcoming improvements will enable SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for companies in Twin Falls.
Producers of SIP components are rapidly designing new hardware and software to take advantage of this new Internet communications scheme. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before setting up a SIP solution in your company, you should take into account the proper settings of your corporate firewall to accept SIP. Many of the common firewalls currently installed in business offices are not developed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are usually closed 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 carrier or Metro fiber service is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to appropriately support SIP to overcome these technical issues. By including a SIP proxy and registrar for controlling the firewall, it is possible to deal with complicated SIP scenarios for reliable and private communications.
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