SIP
What is SIP?
The Session Initiation Protocol (SIP) is an Internet Engineering Task Force (IETF) standard protocol for initiating an interactive user session that involves multimedia elements such as video, voice, chat, gaming, and virtual reality.
Like HTTP or SMTP, SIP works in the Application layer of the Open Systems Interconnection (OSI) communications model. The Application layer is the level responsible for ensuring that communication is possible. SIP can establish multimedia sessions or Internet telephony calls, and modify or terminate them. The protocol can also invite participants to unicast or multicast sessions that do not necessarily involve the initiator. Because the SIP supports name mapping and redirection services, it makes it possible for users to initiate and receive communications and services from any location, and for networks to identify the users wherever they might be.
How can SIP Trunking Save Me Money?
A SIP Trunk is a logical connection between an IP PBX and a Service Provider’s application servers that allows voice over IP traffic to be exchanged between the two. When a call is placed from an internal phone to an external number, the PBX sends the necessary information to the SIP Trunk provider who establishes the call to the dialed number and acts as an intermediary for the call. All signaling and voice traffic between the PBX and the provider is exchanged using SIP and RTP protocol packets over the IP network.
If the number being called is a traditional PSTN telephone, the trunk provider routes the IP packets to a PSTN Gateway that is closest to the number being called, to minimize possible long distance charges. The provider can also terminate PSTN numbers, and route incoming calls for those numbers back to the IP PBX over the SIP which can also Trunk. This allows businesses to offer local phone numbers in several geographical areas, but service them all from a single location.
If the number being called can be reached over a SIP Trunk, the call does not need to be routed over the PSTN, but can be carried on the IP network end to end. Providers can deliver these calls to their customers for very little cost, and many offer them at no charge. Some service providers have agreements and exchange calls for each other’s customers directly over their IP networks. Where no such agreements are in place, calls are routed over the PSTN, even though both endpoints may ultimately be reachable over a SIP Trunk.
The SIP Trunk can be provided by the Internet Service Provider, or by an independent Internet Telephony Service Provider. In fact, there can be several parties involved, each one providing a different part of the service needed to deliver the end-to-end communication: Internet access, SIP termination, PSTN gateway, etc.
Because a SIP Trunk is not a physical connection, there is no explicit limit on the number of calls that can be carried over a single trunk. Each call consumes a certain amount of network bandwidth, so the number of calls is limited by the amount of bandwidth that can flow between the IP PBX and the provider’s equipment.
For marketing reasons, many providers associate a SIP Trunk with a single phone call. When you purchase such a SIP Trunk, it provides the ability to place one call at a time through the service provider’s network. Organizations purchase as many trunks as they need to support their specific needs, typically one trunk per 3-5 employees. Some providers allow several calls to be carried over a single trunk.
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