Session Initiation Protocol (SIP) Trunking

Session Initiation Protocol or SIP is a packet-switched communications protocol which provides IP-protocol based connectivity between an organization’s internal PBX (Private Branch Exchange) and the PSTN network (Public Switched Telephone Network). The type of calls which can be handled by SIP varies from a typical voice call to a collaborative multimedia conference session i.e. the entire range of communication media.
The connection established over internet is virtual in nature i.e. the voice and/or data does not travel over a dedicated communication line marked for each individual call. Instead the call is routed in the form of packets over an internet line using either TCP (Transmission Control Protocol), or UDP (User Datagram Protocol) or SCTP (Stream Control Transmission Protocol).
The number of voice connections which can be simultaneously opened is only limited by the bandwidth of the internet network over which the packets travel. For secure communications, the transmissions can be secured using TLS or Transport Layer Security.
SIP is the underlying switching technology used in VOIP-based telephone systems.

What is Primary Rate Interface (PRI) Trunking

Primary Rate Interface or PRI is a circuit-switched communications protocol used to connect an organizations internal PBX to a PSTN. A PRI-based trunk line, i.e. the communications line connecting the PBX to the PSTN, has dedicated communication lines for each voice call connection.
In the US, the PRI Trunk lines consist of a T-1 line which has 24 circuit-switched channels which consist of 23 voice channels and 1 control channel which carries control information. The total line rate of a T-1 line is 1.544 Megabytes per second.
In Europe, the PRI Trunk lines consist of an E-1 line which has 32 circuit-switched channels, consisting of 30 voice channels and 2 channels for carrying control information. The total line rate of a E-1 line is 2.048 Megabytes per second.
Initially PRI trunking was based on analog transmission, but with ISDNs (Integrated Services Digital Networks) being used for PRI Trunking, digital transmissions became possible.

Why chose SIP trunking over PRI trunking

SIP trunking has certain inherent advantages over PRI trunking due to its very packet-switched nature versus circuit-switched nature of PRI trunking. Let us look at the various advantages SIP offers over PRI which makes it the de-facto choice for organizations setting up new IP PBX systems, or modernizing their existing PBX system’s connectivity with the PSTN, all over the world –

Advantage 1. Total Cost of Ownership is significantly lower for SIP trunking

SIP is based on virtual technology. From the packets travelling over the connections, the connections themselves, to the switching of calls – everything is virtual or software-based.
PRI on the other hand needs physical terminations for each line and timeslot channel. Moreover, a physical mechanism is required for connection end-points for both data and voice transmissions.
The virtual nature of connection and transmission for SIP makes it much cheaper than PRI-based trunking. SIP turns out to be cheaper in installation (or capital expenditure), maintenance (or operational expenditure) and even has the advantage of lower call costs with least-cost routing feature being provided in-built in SIP systems. This drastically reduces the total cost of ownership for SIP based trunking vis-à-vis PRI based trunking.

Advantage 2. Scalability is cheaper for SIP trunking than with PRI trunking

While SIP runs on a packet-switched network, the number of calls which can take place in parallel are limited only by the backbone internet connection’s bandwidth. Having more channels for communications requires only increasing the network bandwidth which can be done on a temporary need-basis as well.
PRI however is limited by the number of channels (23 for T-1 or 30 for E-1). If more parallel connections are needed, then new line(s) need to be laid down which means extra fixed capital investment.

Advantage 3. Failover and redundancy is possible with SIP trunking

In PRI trunking, any outage in the physical line (T-1 or E-1) between the PBX and the PSTN leads to failure of services.
With SIP such PSTN-specific outages can be easily handled with failover systems built on top of redundant connections with multiple PSTNs. Using IP-based networks, an organization can have connections with multiple PSTNs. In the event of an outage of a particular PSTN’s network, the connections of other PSTN networks can handle extra load and there is no outage and consequent disruption in telecommunication services of the enterprise using SIP trunking.

How Enterprise Systems provides SIP trunking for businesses

Enterprise Systems has developed set of custom solutions for providing SIP trunking for businesses which are either setting up new IP-based software PBX or have already setup traditional hardware PBX and now want to utilize the benefits that SIP trunking provides.

Solution SIP-A: For a traditional PBX System (with SIP-unaware devices)

In this scenario Enterprise Systems recommends a solution which uses a VoIP gateway to convert the PBX interfaces to Ethernet while packetizing the voice traffic for transport over the IP-based trunk to the SIP service provider network. In this case a multiport gateway is used to connect the legacy system with the service provider network.

Solution SIP-B: For new IP-PBX System (for IP devices that are already SIP enabled)

In this scenario, Enterprise System recommends a solution in which the IP devices would typically interact with the trunk directly. In addition, this solution also has provisions for setting up advanced features which include Quality of Service (QoS), call admission control, billing and security. However, for enabling the advanced features intermediate devices may be needed. The intermediate devices which are typically utilized for such purposed include session border controller and an IP PBX.
Also packaged with this solution are features such as voice enabled IVR (Interactive Voice Response), call parking and inter-system integration between subsystems.