Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) is the transmission 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 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 expanded to handle 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 Sellersburg.
Manufacturers of SIP equipment are rapidly crafting 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 obtainable from companies such as Ingate Systems and Cisco.
Prior to implementing a SIP plan in your company, you should consider the appropriate configuration of your corporate firewall to accept SIP. Many of the common firewalls currently deployed in business offices are not developed to support the SIP protocol. First, SIP media streams are transmitted over dynamically assigned UDP ports that are often closed on firewalls. Second, SIP clients inside a firewall can not be accessed using IP addresses because these addresses are local and unique to the LAN. Third, you need to ensure that either your T1 line broadband carrier or Ethernet fiber carrier is able to support the SIP protocol from your internal network to the outside world. Your IT manager will need to evaluate how to properly support SIP to get around these technical issues. By adding a SIP proxy and registrar for managing the firewall, it is possible to deal with complex SIP scenarios for secure and private communications.
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