IP trunking and SIP Trunking

It refers to the use of an IP network to carry voice traffic between two private branch exchanges (PBX) or voice switches. IP trunking suggests that rather than having two separate facilities for voice and data, a single facility for data should be employed. To do this, the PBXs are IP-enabled with a special card that can take digitized voice, place it in IP packets, address them to the remote PBX, and insert them into the IP network. These packets would travel across the inter-location link, along with conventional data traffic.

Underlying,  IP network needs to be able to detect the difference between voice and data packets, to ensure that voice traffic is assured of a minimum bandwidth threshold, and to ensure that voice traffic does not experience unacceptable delay, jitter, or loss as it traverses the network. In short, the IP network needs to be capable of identifying the desired class of service (CoS) of each packet, and delivering the necessary quality of service (QoS).

Over a voice trunk, two things must occur: voice transport and signaling.

It should be noted that the term IP trunking is often used interchangeably with SIP trunking. Although the terms are often interchanged, the concept is quite different. SIP trunking does not merely carry IP packets between two locations, it also deals with SIP signaling in the network. So where an IP trunk is analogous to a tie-line, a SIP trunk is analogous to a Primary Rate Interface (PRI) or T-1 voice trunk that can be obtained from many common carriers today. This trunk connects a private, PBX-based network with the PSTN. SIP trunking is the packet-oriented replacement for traditional circuit-oriented trunking services, such as ISDN Primary Rate Interface (PRI).

SIP trunking is a service that replaces the traditional circuit-based trunk between a subscriber and a voice service provider. It replaces the PRI or simple T-1 trunk traditionally used. For the customer with a PBX, they would equip their PBX with an IP-capable trunk card, and then connect the SIP trunk to this service. The PBX would use SIP to exchange signaling information with the service provider, and would deliver (and receive) voice in IP packets. There are several benefits to SIP trunking. These include:

• Because there are several more bandwidth friendly voice encryption standards, if the service provider supports these options for voice encoding, a trunk can support more calls than it might otherwise support in the circuit-based model, which can produce a significant cost reduction or avoid the need to deploy additional trunks.
• For a private enterprise that has already deployed IP telephony on premises, a SIP trunk eliminates the need for them to deploy a media gateway on premises. They can simply deliver their voice packets directly to the service provider, and the service provider can provide media gateway functions in their network as needed.
• SIP trunking makes converged access possible. Once the access service is packetized, it becomes possible to combine a SIP trunk, a transport data service, and even an Internet service on the same access facility.