Session Initiation Protocol (News - Alert) (SIP) technology was quickly adopted by many because of its benefits, such as shorter integration time and lower implementation costs, compared to traditional CTI-based (computer telephony integration) deployments.  But, as SIP technology matures, and is being used in higher volume applications, more questions are being asked about the protocol’s extended use.  

 

A hot topic in developer’s circles is that of Early Media (EM), and its effect on high volume dial applications.  BandTel addresses EM in high-volume applications and highlights how SIP further leverages the power of internet telephony.

 

Prior to SS7-ISUP and ISDN, circuit-switched telephone calls used In-Band Call-Progress Tones. These tones let one side know what was happening on the other side of the call. When a number was dialed, audible tones could be heard as the call cut through to the endpoint. The tones would be an audible ring, a busy tone, or a fast busy tone to indicate the state of the downstream station at the far end.

 

Transporting of this legacy call-progress information in SIP space is called “early media” because it describes the audio call-progress media being transported to the calling party. In so doing, legacy signaling techniques are preserved, but at a price. Bandwidth is being consumed. In fact, as much bandwidth is being consumed going back upstream to the calling party as is when the call is answered, stable, and the two parties are talking. Applications that handle high-volume outbound dialing in SIP space have noticed the glaring inefficiencies of the wasted bandwidth.

 

If the SIP User Agent has the ability to synthesize all the audible tones locally for the caller, then why are we wasting bandwidth for unneeded audible signaling during the ring state?

 

That is the question that BandTel’s (News - Alert) team of engineers set out to answer. When SS7-ISUP and ISDN entered the telecommunications world, they allowed those signaling tones to be replaced with a data packet message. To indicate ringing in SS7, it was called an ACM (Address Complete Message), and in ISDN it was called an ALERT.

 

If something besides ringing was occurring at the far end, there was a “cause code” introduced in the Release Message of ISUP and ISDN. This code relayed in great detail what was happening (i.e. the endpoint was busy, the telephone number was bad, no circuit was available, etc.). This was a huge step forward over the old audio technique.

 

But, because the PSTN network was, and still is, made up of thousands of telephone companies all interconnected, old audio signaling mechanisms had to be preserved. Those legacy signaling mechanisms, without the intelligence of ISUP and ISDN signaling, demanded the use of audible tones - even over ubiquitous SS7 PSTN networks with ISDN stations.

 

Here’s where VoIP telephony, and specifically SIP, solve the legacy problem. SIP is similar to SS7 and ISDN in that it also allows the call progress information to be sent upstream to the originating end via data packets, as opposed to audible tones. Furthermore, with intelligence now being moved to the edge of the SIP station device, we have the ability to synthesize all the old audible tones right at the SIP end-point user agent (UA).

 

But legacy persists. Many PSTN networks still cannot send proper call-progress information out-of-band (i.e. via SS7). At the TDM/packet crossover point, the old audio tones live on, and are for the most part, transported back upstream to a SIP UA using the media portion of the SIP call. Hence, “early media” is wasting bandwidth.

 

The designers of the SIP protocol realized the waste early on and endowed the protocol with the ability for SIP endpoints to negotiate between each other to “turn off early media.” Bandwidth will no longer be consumed for audible signaling tones during the ring state. Instead, the upstream UA simply relies on the SIP call-progress messages. The audible tones are then synthesized for the calling party to listen to… if they want to hear it.

 

For small (one and two port) UA devices this is not all that significant as the single call on the IP connection is either ringing or answered and stable. But, for large call centers that dial at high rates, the ability to turn off early media is profound.

 

Outbound call centers generally dial out at a rate that exceeds the number of physical agents that are sitting in the call center. Only a fraction of the calls made get answered at the far end. In order to keep the agent pool busy and talking at all times, a ratio of dialed calls to agents is maintained. Many times that ratio can be as high as 4, 5, or even 6 calls dialing for every agent present.

           

In such heavy dialing scenarios, the effects of early media are dramatic. It is possible to have as much bandwidth consumed inbound to a call center just transporting ring-state call progress information (which nobody wants or needs to listen to) as is being consumed for actual voice communications.

           

With “early media enabled,” the call center may need to provision as much as two or three times the amount of bandwidth to get the desired agent/call-dialed ratio needed to optimize the call center agent utilization. Ordering extra “wasted bandwidth” is expensive. It’s bad enough in countries where broadband is available, but in countries were bandwidth is limited and the cost is high, this becomes a very significant problem. The cost can actually triple or quadruple a call center’s bandwidth bill depending on the dialing patterns.

           

BandTel realized the significance of this problem for high-dial application vertical markets and responded with a solution. For high intensity dial applications, the company engineered a dial-plan product that offers A-Z routes with “early media disabled.” This “Silent Running” treatment will result in significant bandwidth savings—as much as three or four times. These savings in bandwidth go straight to the customer’s bottom line.

           

Looking at the larger picture, because TDM networks have the early media problem built into their architecture, there is no solution. They are shackled to having to use the bearer portion of their networks to transport in-band call-progress information. That expense cannot be billed to customers to recoup costs. As such, we see one more reason why the world is evolving to packet solutions for telephony.  

 

BandTel’s Silent Running solution is available today, allowing any enterprise to take advantage of this feature now, to any destination worldwide. 

 

As co-founder and Chief Technology Officer (CTO) of BandTel, Peter Sandstrom is responsible for setting the technical direction of the company, and promoting the company’s continuous innovation. He can be reached at [email protected].