Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport technology for person-to-person real time data over the Web. Its defining 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 extended to deal with IP telephony combined with video and instant messaging. Future enhancements will allow SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for companies in Questa.
Makers of SIP equipment are quickly creating innovative hardware and software to take advantage of this new Internet communications scheme. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Prior to implementing a SIP solution in your organization, you should think about the appropriate configuration of your corporate firewall to accept SIP. Many of the common firewalls currently deployed in business offices are not designed to allow the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are typically blocked on firewalls. Second, SIP clients within a firewall can not be accessed 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 Metro fiber provider is able to handle the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to properly support SIP to overcome these technical difficulties. By adding a SIP proxy and registrar for controlling the firewall, it is possible to handle complicated SIP situations for secure and private communications.
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