Addressing 'VoIPmageddon': Vetting SIP Trunking ProvidersAddressing 'VoIPmageddon': Vetting SIP Trunking Providers
Understanding how your VoIP provider handles calls across its network is as important as knowing how your VoIP implementations run in your own network.
March 27, 2015
Understanding how your VoIP provider handles calls across its network is as important as knowing how your VoIP implementations run in your own network.
In a recent No Jitter post, "VoIPmageddon: Is Quality Leading to a Telephony Meltdown?," and corresponding session at last week's Enterprise Connect conference in Orlando, Phil Edholm, founder of PKE Consulting, highlighted an important issue in telephony today: the increasing challenge of delivering quality voice as the number of VoIP endpoints grows. The challenge is indeed quite real, but mitigation is possible.
The following best practices can help providers assure delivery of quality VoIP in today's "fractured" world. Enterprise communications decision-makers would be well advised to make sure their providers of choice embrace them.
Use of SIP trunking: As Phil recommended, VoIP providers need to use SIP trunking as much as possible for delivery of voice calls to the customers they service, and for origination from the carriers with which they peer. In this scenario, the provider would only use TDM in the local access network, and VoIP from a local point of interconnection with the PSTN to the customer. By being as close as possible to the originating carrier, the provider can avoid the multiple TDM/SIP conversions that are so detrimental to quality and latency. Replacement of TDM interconnections with SIP-based interconnections is especially relevant where the core network is already packet-based.
As Phil indicated, voice quality in conferences is more easily perceived because of the statistical probability of having multiple VoIP endpoints in a multiparty conference. So if your company operates a conferencing service, additional requirements apply. For example, your VoIP carrier should provide international and domestic SIP backhaul capabilities to your conference bridge.
To maximize voice quality for conferencing, providers should follow similar steps to those outlined above, aggregating the traffic directly to the conference bridge using SIP, and integrating both the PSTN-originated and WebRTC traffic. This minimizes latency and transcoding. The choice of a dedicated IP-based backhaul network is essential to minimize transmission latency and reduce jitter. As a result of this, conferencing providers can reduce the size of required jitter buffers, reducing the latency for conference calls.
The same goes for contact center environments, where similar approaches as stated above are critical -- a challenge made more difficult given that many remote agents are a far distance from the actual contact center core site. Peer management is particularly important in this market, with the use of local points of presence for both customer and agents advisable. In addition, the provider should optimize the backhaul system for latency and transcoding to ensure delivery of a quality experience.
For customers with multiple locations, having direct SIP trunks at geographically diverse locations versus a single trunk using the enterprise IP infrastructure is also critical for both quality and cost. By tying real-time traffic delivery to the best point of presence for the associated endpoint (the enterprise PBX or UC platform), a provider can actively reduce the loops created in the backhaul system when there's only a single point. For example, if a single SIP trunk point is chosen in California, then an employee in New York talking to a client in Europe would essentially be hairpinned -- adding 70 to 80 milliseconds of delay or more (dependent on jitter buffers) to the conversation. By providing separate SIP trunks in both California and New York, a provider can optimize the quality to the endpoint's location. Easing the mapping of those access points to users, and enabling numbering schemes through per-number call delivery, are very helpful for managing quality in large deployments.
Is My SIP Trunking Provider Following These Best Practices?
All of these best practices are, quite frankly, common sense for a SIP trunking provider looking to deliver a quality voice experience (or in the future, video) in the evolving VoIP world. Organizations buying SIP trunks for their real-time applications should develop a list of these capabilities (for example, location of PBX platforms, IP connectivity at these locations, etc.), and discuss them with their potential providers as part of the acquisition process. In turn, SIP trunking providers should help their customers define elements in their networks that can be optimized to enhance quality.As Phil pointed out, it is no longer enough to think about how your VoIP implementations run in your own network. In the new world of ever-increasing VoIP endpoint adoption, the choices you make in how you extend VoIP communications outside your network have a much bigger impact on the quality of your service (and your company).
Gaetan Brichet is COO of Voxbone
