Consider SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transport technology for person-to-person voice traffic over the Web. 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 handle IP telephony combined with video and instant messaging. Future enhancements will enable SIP applications like video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Claypool.
Manufacturers of SIP hardware are quickly crafting new products and software to take advantage of this new Internet communications method. SIP telephones, PC client software, SIP servers, routers, and firewalls are now on the market from companies such as Ingate Systems and Cisco.
Prior to implementing a SIP plan in your organization, you should consider the proper 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 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 carrier or Ethernet fiber service is able to handle the SIP protocol from your internal network to the outside world. Your IT administrator will need to ascertain how to correctly support SIP to get around these technical issues. By including a SIP proxy and registrar for managing the firewall, it is feasible to deal with complex SIP situations for reliable and confidential communications.
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