Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the communications 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 in conjunction with video and instant messaging. Forthcoming improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Stamps.
Suppliers of SIP components are quickly designing innovative products and software to take advantage of this new Internet communications method. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now obtainable from companies such as Ingate Systems and Cisco.
Before implementing a SIP solution in your organization, you should take into account the appropriate settings of your corporate firewall to support SIP. Many of the common firewalls already deployed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically allocated UDP ports that are typically 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 carrier or Metro fiber provider is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to ascertain how to properly support SIP to overcome these technical issues. By including a SIP proxy and registrar for managing the firewall, it is possible to handle complicated SIP situations for secure and confidential communications.
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