UC Unplugged

Planning for SIP Trunking; Considerations to Take into Account

By TMCnet Special Guest
Mike Sheridan, Executive Vice President of Worldwide Sales, Aspect
  |  November 01, 2011

This article originally appeared in the Nov. 2011 issue of INTERNET TELEPHONY.

No matter where I travel in the world, telecom managers’ eyes gleam when the topic of SIP trunking arises. So it wasn’t much of a surprise to me as I looked at a SIP Trunking Snapshot to see that the viewpoint on adoption is still, for the most part, looked at from a futuristic standpoint, but that more and more people are expressing excitement about it. Alright, so we know that there is the potential for immediate and significant cost savings in the business. But what does it mean for those highly regulated outbound contact centers?

From the two-second rule to call abandonment to mandatory caller ID, there are many regulations that organizations understand they need to plan for in regards to their outbound customer contacts. It doesn’t just stop there, though. For other operations, like answering machine detection, it’s important to realize that this transformation requires even further planning to make sure that the experience is equivalent in a SIP environment to a time division multiplexing environment.

And it’s complicated. AMD (News - Alert) for outbound SIP connections is typically done by requesting early media where the caller asks the receiver to send audio, even before the call is officially established, allowing processing of the received audio as soon as it’s available. To send the audio, the receiving SIP phone collects the speaker’s voice, taking tiny slices of the audio stream and sending it over the IP network as a data packet. In many cases, the audio is compressed so it takes less network bandwidth, but that also may reduce the audio quality depending on the encoding strategy. The latency of transmission with the packets going through a switched network adds to the delay in reaching the caller. There is also a jitter buffer on the receiver used to reassemble the packets into a continuous audio stream before playing them out so the playback isn’t broken up as there is no assurance that the next packet will arrive exactly when needed.

All of these added times could affect the ability to meet certain regulations that may be in place on how fast an agent or IVR must be connected to the person that answers an automated call. These rules often mandate a response within two seconds, making even a 5 percent increase in receiving audio a potential issue. So while AMD over SIP is absolutely viable, it’s something that needs to be carefully planned for to eliminate any extra delays that could potentially put an organization out of compliance.

What are some things that you’ve encountered when moving to a SIP environment? What about while applying AMD in your outbound SIP connections?

 

 Mike Sheridan is  executive vice president of worldwide sales with Aspect (News - Alert) (www.aspect.com).


Edited by Stefania Viscusi

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