Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission protocol for person-to-person voice traffic over the Web. 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 extended to handle IP telephony in conjunction with video and instant messaging. Forthcoming improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Buffalo.
Manufacturers of SIP equipment are rapidly designing 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 on the market from companies such as Ingate Systems and Cisco.
Prior to setting up a SIP plan in your organization, you should consider the appropriate settings of your corporate firewall to accept SIP. Many of the common firewalls currently deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are often blocked 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 provider 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 correctly support SIP to get around these technical issues. By including a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP situations for secure and private communications.
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