Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the communications protocol 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 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 in conjunction with video and instant messaging. Future enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in East Holden.
Producers of SIP components are quickly creating innovative products and software to take advantage of this new Internet communications scheme. SIP telephones, PC client programs, 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 take into account the proper settings of your corporate firewall to support SIP. Many of the typical firewalls already installed in business offices are not developed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are usually closed on firewalls. Second, SIP clients inside 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 provider or Metro fiber carrier is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to properly support SIP to overcome these technical challenges. By adding a SIP proxy and registrar for controlling the firewall, it is possible to deal with complicated SIP situations for reliable and private communications.
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