SIP Trunking is the way forward

SIP Trunking, an IP technology based on the Session Initiation Protocol (SIP), is the newest method to connect an IP-based Private Branch Exchange (IP-PBX) or Unified Communication and Collaboration (UCC) instance to the Public Switched Telephony Network (PSTN). Because of the benefits it brings, SIP Trunking is replacing traditional analogue (POTS) and digital access (ISDN) technologies at a steady pace. In addition, telecom operators – being PSTN access providers – have gradually transformed their networks from TDM to IP over the last 20 years. So, it is a natural evolution that PSTN access technology changes as well. Telecom operators are urging their customers (or are even “pushing” them a bit) to move to SIP Trunking. Nowadays, there are still cases where an organization has invested in an IP-based PBX or full-fledged UCC instance (which can be Cloud-based), but still connects to the PSTN by means of traditional ISDN lines. This creates needless conversions from IP (inside the organization’s network) to TDM (PSTN access by means of ISDN lines) and back to IP (in the telecom operator’s network providing access to the PSTN) and vice versa. It also means that the organization must invest in media gateways to do the conversion from IP to TDM, which could have been prevented by using SIP Trunking right away.

Advantages of SIP Trunking

SIP Trunking offers the following advantages compared to a traditional PSTN access solution:

1. Economies of scale

Using a single network technology supporting both voice and data leads to economies of scale. Thanks to Voice over IP technology, voice traffic is carried over an organization’s LAN or Virtual Private WAN Network together with data traffic. SIP Trunking enables organizations with a Virtual Private WAN Network to consolidate all their external voice traffic on a central – though redundant – IP link to access the PSTN. The individual locations no longer need their own PSTN access. This cuts down on connectivity costs and organizations also save on infrastructure and rack space (i.e., no need for media gateways). We can draw a comparison with Cloud services: the PSTN is a Cloud service and SIP Trunking provides centralized access to that Cloud service.

2. Optimized bandwidth usage

With a traditional PSTN access solution based on ISDN, bandwidth is scaled as a multiple of discrete 128Kbps (2 simultaneous calls via BRI/BRA/ISDN2) or 2Mb/s connections (30 simultaneous calls via PRI/PRA/ISDN30). In most cases there is overcapacity. SIP Trunking allows you to scale bandwidth in accordance with actual needs. The number of voice channels supported by a SIP Trunk is based on the expected number of simultaneous calls (SimCall) and can be adjusted on demand. The required bandwidth per SimCall depends on the voice codec used: a SimCall consumes +/- 100Kbit/s with G.711 a-law and +/- 40Kbit/s with G.729 annex A. G.711 is recommended when high-quality audio is required. G.729 provides better audio quality than mobile cellular audio quality (GSM R2).

3. Access resiliency

In combination with redundant WAN connectivity to the locations where the organization’s Session Border Controllers (SBC) are hosted (providing access to the organizations IP-PBX’s or UCC instances), SIP Trunking brings additional resiliency compared to a traditional ISDN solution. Each SBC is “dual-homed”: it is logically connected to two different SIP session control entities of the telecom operator providing access to the PSTN. There is automatic failover between both SIP session control entities in case one would fail.

4. Access performance monitoring

SIP Trunking allows for online performance monitoring of connectivity dedicated to voice communication. Whenever needed, additional bandwidth can be added.

5. Multi-country support

All external voice traffic is consolidated on a central IP link to access the PSTN. This central IP link, which is supporting multiple countries, is implemented in a geographically redundant setup.

6. Optimized contract and supplier management

SIP Trunking allows a customer to have a single agreement with one telecom operator for several countries covered by the service. Organizations work with a single contract and single point of contact (Service Desk, Service Delivery Management, reporting, invoicing, …) for networking and voice services. This is helpful if the telecom operator delivering SIP trunking services is also providing WAN services.

Deutsche Telekom’s SIP Trunking offering

Deutsche Telekom‘s SIP Trunking offering is marketed under the name Corporate SIP (CSIP). The service is currently available in the following 20 European countries: Germany, Austria, Belgium, Italy, Portugal, Sweden, Switzerland, Romania, Spain, Denmark, Ireland, Netherlands, Czech Republic, France, United Kingdom, Norway, Slovakia, Finland, Luxembourg and Poland. Additional countries and continents will be added in succession. CSIP is more than just SIP Trunking technology. All country-specific requirements are fulfilled, in full compliance with regulatory requirements, phone number porting, emergency call handling, data retention, the facilitation of local contracting where mandatory, etc. Phone Number Porting from the current provider(s) to Deutsche Telekom is fully managed by Deutsche Telekom. The customer does not need to change its telephone numbers to transfer its voice business. Fax communication is being supported by means of G.711 pass-through or the T.38 standard. It must also be stressed that Deutsche Telekom‘s SIP Trunking offering is not based on the public internet. CSIP is provided via Deutsche Telekom’s own secure IP network. In the comparison with access to Cloud services, once could say that CSIP provides a centralized and dedicated (or virtual private) access to the PSTN.

What about organizations that still have a traditional (TDM-based) PBX environment ?

An organization does not necessarily need to have implemented an IP-PBX or UCC solution in order to benefit from the advantages of SIP Trunking. Organizations that are still using TDM-based PBX’s can benefit from a SIP Trunking solution without the need to change their current PBX landscape overnight. Such organizations may be willing to gradually migrate to IP Telephony and UCC technology and therefore spread investment over time. A phased approach can be taken. As a first step, a centralized SIP Trunking solution is implemented while keeping the existing TDM-based PBXs “untouched.” By installing a media gateway adjacent to each PBX, the existing PBXs can be integrated with the SIP Trunking solution. These media gateways convert TDM (ISDN signalling and voice payload) into IP (SIP signalling and RTP). The media gateway can be a function integrated into a router (Integrated Access Device – IAD) or can be a separate device connected to a router. As a second step, the existing TDM PBXs can be gradually replaced by IP-PBXs or UCC systems. The organization may opt to invest in a distributed IP-PBX/UCC infrastructure (meaning that a system is implemented in every location) or decide to implement a centralized IP-PBX/UCC infrastructure. In any case, each IP-PBX/UCC system will interface with the centralised SIP Trunking solution to allow bundling of voice traffic to the PSTN. The media gateways in the different locations may remain in place as a media gateway (e.g., in a country where SIP Trunking is not available yet), or may be upgraded – if possible – to work as an SBC or as an IP-PBX/UCC Local Survivability Function (LSF).

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