Consider SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transmission protocol for person-to-person voice traffic over the Internet. 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 handle IP telephony in conjunction 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 Holyoke.
Manufacturers of SIP equipment are rapidly developing new products and software to get the upper hand in this new Internet communications method. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Prior to installing a SIP solution in your organization, you should think about the proper settings of your corporate firewall to accept SIP. Many of the typical firewalls currently deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transferred over dynamically assigned UDP ports that are typically closed on firewalls. Second, SIP clients inside a firewall can not be reached using IP addresses because these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line provider or Metro fiber provider is able to support the SIP protocol from your internal network to the outside world. Your IT administrator will need to evaluate how to properly support SIP to get around these technical problems. By including a SIP proxy and registrar for managing the firewall, it is possible to deal with complicated SIP situations for reliable and private communications.
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