Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission protocol 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 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 extended to deal with IP telephony combined 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 Mountain Ranch.
Suppliers of SIP components are rapidly designing innovative hardware and software to get the upper hand in this new Internet communications scheme. SIP telephones, PC client applications, SIP servers, routers, and firewalls are now available from companies such as Ingate Systems and Cisco.
Before implementing a SIP solution in your business, you should take into account the proper configuration of your corporate firewall to accept SIP. Many of the typical firewalls currently installed in business offices are not designed 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 accessed using IP addresses since these addresses are local and private to the LAN. Third, you need to ensure that either your T1 line 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 ascertain how to properly support SIP to overcome these technical difficulties. By including a SIP proxy and registrar for managing the firewall, it is feasible to handle complicated SIP scenarios for secure and confidential communications.
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