Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission technology for person-to-person voice data 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 expanded to handle IP telephony in conjunction with video and instant messaging. Future improvements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Carrollton.
Producers of SIP equipment are quickly creating innovative hardware and software to take advantage of this new Internet communications technique. SIP telephones, PC client programs, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before setting up a SIP plan in your organization, you should take into account the appropriate configuration of your corporate firewall to support SIP. Many of the common firewalls currently deployed in business offices are not designed to support the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are typically blocked on firewalls. Second, SIP clients within 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 Ethernet 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 appropriately support SIP to get around these technical difficulties. By adding a SIP proxy and registrar for managing the firewall, it is feasible to handle complex SIP scenarios for reliable and private communications.
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