Why You Need SIP Protocol for Voice Communications
Session Initiation Protocol (SIP) 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 extended to handle IP telephony in conjunction with video and instant messaging. Forthcoming enhancements will allow SIP applications such as video conference calls, application sharing, home monitoring, and interactive gaming for businesses in Dona Ana.
Makers of SIP components are quickly creating new products and software to get the upper hand in 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.
Prior to implementing a SIP plan in your business, you should take into account the appropriate settings of your corporate firewall to accept SIP. Many of the common firewalls already deployed in business offices are not designed to support the SIP protocol. First, SIP media streams are transferred over dynamically allocated UDP ports that are often closed on firewalls. Second, SIP clients within 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 provider or Ethernet fiber provider 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 overcome these technical difficulties. By adding a SIP proxy and registrar for managing the firewall, it is feasible to handle complex SIP scenarios for secure and private communications.
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