Why You Need SIP Protocol for Voice Communications
SIP (Session Initiation Protocol) is the transport protocol for person-to-person real time traffic over the Internet. Its technical 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 combined with video and instant messaging. Future improvements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Mosca.
Makers of SIP equipment are rapidly designing new hardware and software to take advantage of this new Internet communications technique. SIP telephones, PC client software, 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 think about the appropriate settings of your corporate firewall to support SIP. Many of the common firewalls already installed in business offices are not developed to allow the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are usually blocked on firewalls. Second, SIP clients inside a firewall can not be reached using IP addresses since these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line broadband carrier 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 correctly support SIP to get around these technical difficulties. By including 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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