One of the beauties of voice over IP (VoIP) is cheap or even free long-distance phone calls. Companies are ableto call other branches at a very low cost because most of the calls happen over the Internet. Enter SIP!
A SIP trunk is a direct connection between your organization and an Internet telephony service provider (ITSP). It enables you to extend voice over IP telephony beyond your organization’s firewall without the need for an IP-PSTN gateway
SIP (session initiation protocol) is a signaling protocol that is widely used for setting up, connecting, and disconnecting communication sessions, typically voice or video calls over the Internet. SIP is a standardized protocol with its basis coming from the IP community; in most cases, it uses UDP or TCP. The protocol can be used for setting up, modifying, and terminating two-party (unicast), or multi-party (multicast) sessions consisting of one or more media streams.
Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call.
Advantages of SIP Trunk
The major advantage of using SIP trunking over PSTN is cost. Companies are able to save a lot on hardware purchases. There are, however, some other advantages.
- * Cost
SIP trunking eliminates the need to purchase ISDN or local PSTN gateways, lowering telephony costs. Also, since most of the configuration is done virtually (over the WAN), there is less hardware required, thereby saving money on purchasing extra hardware or modules.
* Easier Maintenance and Scaling
Because SIP trunks are virtual rather than physical, scaling and maintenance are much easier. While old trunks require expensive installation of circuits and termination points, SIP trunks can be easily adjusted with a change to the software configuration.
- * Integration
Integration of a SIP trunk into an existing infrastructure is not tedious and cumbersome.
- * Converged Network for Voice and Video Communication
A SIP trunk eliminates the need for separate voice and data connections and expands the potential for communications convergence using both voice and data together. This ensures that, as an organization grows, the necessary infrastructure required to handle additional voice/data traffic is already in place. A single corporate SIP trunking account can serve an entire enterprise, no matter the size.
* Eliminate IP-PSTN Gateways (or Even Your Entire PBX)
Because SIP trunks connect directly to your ITSP without traversing the publicly switched telephone network, you can get rid of IP-PSTN gateways and their attendant cost and complexity
SIP is a peer-to-peer protocol. The peers in a session are called user agents (UAs). A user agent can function in one of the following roles:
* User agent client (UAC)—A client application that initiates the SIP request.
* User agent server (UAS)—A server application that contacts the user when a SIP request is received and that returns a response on behalf of the user.
* Proxy server—Receives SIP requests from a client and forwards them on the client’s behalf. Basically, proxy servers receive SIP messages and forward them to the next SIP server in the network. Proxy servers can provide functions such as authentication, authorization, network access control, routing, reliable request retransmission, and security.
SIP Trunk Configuration
Below is a sample configuration of a SIP-enabled network on a Cisco 2951 router.
Please note that, for some ITSP, you may be given a virtual SIP domain, e.g., firstname.lastname@example.org or an IP address. The configuration is not too difficult because just a few modifications will be required.