Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission technology for person-to-person voice data over the Internet. Its technical 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 deal with IP telephony combined 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 Fort Payne.
Producers of SIP hardware are quickly creating innovative hardware and software to get the upper hand in this new Internet communications technique. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to installing a SIP solution in your organization, you should take into account the appropriate settings of your corporate firewall to support SIP. Many of the typical firewalls currently deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are usually blocked on firewalls. Second, SIP clients within a firewall can not be reached using IP addresses since these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Ethernet fiber carrier is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to correctly support SIP to overcome these technical issues. By including a SIP proxy and registrar for controlling the firewall, it is feasible to handle complicated SIP scenarios for secure and private communications.
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