Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission protocol 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 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 deal with IP telephony combined with video and instant messaging. Future improvements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Montgomery.
Suppliers of SIP hardware are rapidly crafting innovative products and software to take advantage of this new Internet communications technique. SIP telephones, PC client software, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Prior to installing a SIP solution in your organization, you should think about the appropriate configuration of your corporate firewall to support SIP. Many of the typical firewalls already deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are usually shut 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 Ethernet fiber carrier is able to handle 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 difficulties. By including a SIP proxy and registrar for managing the firewall, it is feasible to handle complex SIP scenarios for secure and confidential communications.
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